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-rw-r--r--Documentation/devicetree/bindings/sound/ak4613.txt17
-rw-r--r--Documentation/devicetree/bindings/sound/ak4642.txt22
-rw-r--r--Documentation/devicetree/bindings/sound/renesas,rsnd.txt7
-rw-r--r--Documentation/devicetree/bindings/sound/sun4i-codec.txt33
-rw-r--r--Documentation/devicetree/bindings/sound/tdm-slot.txt11
-rw-r--r--drivers/gpu/drm/i915/i915_dma.c1
-rw-r--r--drivers/gpu/drm/i915/i915_drv.h5
-rw-r--r--drivers/gpu/drm/i915/intel_audio.c179
-rw-r--r--drivers/spi/spi-atmel.c21
-rw-r--r--include/drm/i915_component.h17
-rw-r--r--include/sound/hda_regmap.h4
-rw-r--r--include/sound/hdaudio.h6
-rw-r--r--include/sound/pcm.h6
-rw-r--r--include/sound/rcar_snd.h1
-rw-r--r--include/sound/rt5645.h2
-rw-r--r--include/sound/simple_card.h2
-rw-r--r--include/sound/soc.h14
-rw-r--r--include/uapi/sound/asound.h4
-rw-r--r--include/uapi/sound/firewire.h9
-rw-r--r--sound/core/pcm.c3
-rw-r--r--sound/core/pcm_native.c6
-rw-r--r--sound/core/seq/oss/seq_oss_readq.c6
-rw-r--r--sound/core/seq/oss/seq_oss_writeq.c4
-rw-r--r--sound/firewire/Kconfig27
-rw-r--r--sound/firewire/Makefile4
-rw-r--r--sound/firewire/amdtp-am824.c465
-rw-r--r--sound/firewire/amdtp-am824.h52
-rw-r--r--sound/firewire/amdtp-stream.c (renamed from sound/firewire/amdtp.c)379
-rw-r--r--sound/firewire/amdtp-stream.h (renamed from sound/firewire/amdtp.h)116
-rw-r--r--sound/firewire/bebob/bebob.c9
-rw-r--r--sound/firewire/bebob/bebob.h34
-rw-r--r--sound/firewire/bebob/bebob_focusrite.c26
-rw-r--r--sound/firewire/bebob/bebob_maudio.c32
-rw-r--r--sound/firewire/bebob/bebob_midi.c16
-rw-r--r--sound/firewire/bebob/bebob_pcm.c16
-rw-r--r--sound/firewire/bebob/bebob_proc.c6
-rw-r--r--sound/firewire/bebob/bebob_stream.c40
-rw-r--r--sound/firewire/bebob/bebob_terratec.c10
-rw-r--r--sound/firewire/bebob/bebob_yamaha.c6
-rw-r--r--sound/firewire/dice/dice-midi.c12
-rw-r--r--sound/firewire/dice/dice-pcm.c12
-rw-r--r--sound/firewire/dice/dice-stream.c22
-rw-r--r--sound/firewire/dice/dice.h2
-rw-r--r--sound/firewire/digi00x/Makefile4
-rw-r--r--sound/firewire/digi00x/amdtp-dot.c442
-rw-r--r--sound/firewire/digi00x/digi00x-hwdep.c200
-rw-r--r--sound/firewire/digi00x/digi00x-midi.c160
-rw-r--r--sound/firewire/digi00x/digi00x-pcm.c373
-rw-r--r--sound/firewire/digi00x/digi00x-proc.c99
-rw-r--r--sound/firewire/digi00x/digi00x-stream.c422
-rw-r--r--sound/firewire/digi00x/digi00x-transaction.c137
-rw-r--r--sound/firewire/digi00x/digi00x.c171
-rw-r--r--sound/firewire/digi00x/digi00x.h157
-rw-r--r--sound/firewire/fcp.c2
-rw-r--r--sound/firewire/fireworks/fireworks.c12
-rw-r--r--sound/firewire/fireworks/fireworks.h2
-rw-r--r--sound/firewire/fireworks/fireworks_midi.c12
-rw-r--r--sound/firewire/fireworks/fireworks_pcm.c12
-rw-r--r--sound/firewire/fireworks/fireworks_stream.c8
-rw-r--r--sound/firewire/lib.c138
-rw-r--r--sound/firewire/lib.h56
-rw-r--r--sound/firewire/oxfw/oxfw-midi.c16
-rw-r--r--sound/firewire/oxfw/oxfw-pcm.c10
-rw-r--r--sound/firewire/oxfw/oxfw-stream.c16
-rw-r--r--sound/firewire/oxfw/oxfw.c29
-rw-r--r--sound/firewire/oxfw/oxfw.h3
-rw-r--r--sound/firewire/tascam/Makefile4
-rw-r--r--sound/firewire/tascam/amdtp-tascam.c243
-rw-r--r--sound/firewire/tascam/tascam-hwdep.c201
-rw-r--r--sound/firewire/tascam/tascam-midi.c135
-rw-r--r--sound/firewire/tascam/tascam-pcm.c312
-rw-r--r--sound/firewire/tascam/tascam-proc.c88
-rw-r--r--sound/firewire/tascam/tascam-stream.c496
-rw-r--r--sound/firewire/tascam/tascam-transaction.c293
-rw-r--r--sound/firewire/tascam/tascam.c210
-rw-r--r--sound/firewire/tascam/tascam.h147
-rw-r--r--sound/hda/ext/hdac_ext_stream.c9
-rw-r--r--sound/hda/hdac_device.c81
-rw-r--r--sound/pci/hda/hda_codec.c44
-rw-r--r--sound/pci/hda/hda_codec.h18
-rw-r--r--sound/pci/hda/hda_local.h7
-rw-r--r--sound/pci/hda/patch_hdmi.c19
-rw-r--r--sound/pci/rme9652/hdsp.c1
-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile1
-rw-r--r--sound/soc/atmel/atmel_wm8904.c1
-rw-r--r--sound/soc/au1x/db1000.c10
-rw-r--r--sound/soc/au1x/db1200.c10
-rw-r--r--sound/soc/blackfin/bf5xx-ad1836.c11
-rw-r--r--sound/soc/blackfin/bfin-eval-adau1373.c12
-rw-r--r--sound/soc/blackfin/bfin-eval-adau1701.c12
-rw-r--r--sound/soc/blackfin/bfin-eval-adav80x.c12
-rw-r--r--sound/soc/codecs/Kconfig9
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/ak4613.c497
-rw-r--r--sound/soc/codecs/ak4642.c153
-rw-r--r--sound/soc/codecs/arizona.c16
-rw-r--r--sound/soc/codecs/arizona.h2
-rw-r--r--sound/soc/codecs/hdmi.c109
-rw-r--r--sound/soc/codecs/rt5645.c35
-rw-r--r--sound/soc/codecs/rt5645.h6
-rw-r--r--sound/soc/codecs/tlv320aic3x.c30
-rw-r--r--sound/soc/codecs/wm5110.c187
-rw-r--r--sound/soc/davinci/davinci-mcasp.c305
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c1
-rw-r--r--sound/soc/fsl/fsl_sai.c1
-rw-r--r--sound/soc/generic/simple-card.c8
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c17
-rw-r--r--sound/soc/intel/boards/broadwell.c9
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c33
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c1
-rw-r--r--sound/soc/kirkwood/armada-370-db.c1
-rw-r--r--sound/soc/mediatek/mt8173-max98090.c11
-rw-r--r--sound/soc/mediatek/mt8173-rt5650-rt5676.c11
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c6
-rw-r--r--sound/soc/pxa/brownstone.c9
-rw-r--r--sound/soc/pxa/corgi.c11
-rw-r--r--sound/soc/pxa/e740_wm9705.c5
-rw-r--r--sound/soc/pxa/e750_wm9705.c5
-rw-r--r--sound/soc/pxa/e800_wm9712.c5
-rw-r--r--sound/soc/pxa/hx4700.c4
-rw-r--r--sound/soc/pxa/imote2.c11
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c11
-rw-r--r--sound/soc/pxa/palm27x.c9
-rw-r--r--sound/soc/pxa/poodle.c11
-rw-r--r--sound/soc/pxa/pxa-ssp.c1
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c1
-rw-r--r--sound/soc/pxa/spitz.c5
-rw-r--r--sound/soc/pxa/tosa.c5
-rw-r--r--sound/soc/pxa/ttc-dkb.c12
-rw-r--r--sound/soc/qcom/lpass-cpu.c3
-rw-r--r--sound/soc/rockchip/Kconfig4
-rw-r--r--sound/soc/sh/Kconfig1
-rw-r--r--sound/soc/sh/rcar/adg.c303
-rw-r--r--sound/soc/sh/rcar/core.c12
-rw-r--r--sound/soc/sh/rcar/ctu.c6
-rw-r--r--sound/soc/sh/rcar/dvc.c6
-rw-r--r--sound/soc/sh/rcar/mix.c6
-rw-r--r--sound/soc/sh/rcar/rsnd.h15
-rw-r--r--sound/soc/sh/rcar/src.c17
-rw-r--r--sound/soc/sh/rcar/ssi.c98
-rw-r--r--sound/soc/sh/siu_dai.c85
-rw-r--r--sound/soc/soc-core.c25
-rw-r--r--sound/soc/soc-pcm.c49
-rw-r--r--sound/soc/sunxi/Kconfig11
-rw-r--r--sound/soc/sunxi/Makefile2
-rw-r--r--sound/soc/sunxi/sun4i-codec.c719
-rw-r--r--sound/soc/ux500/mop500.c1
-rw-r--r--sound/soc/ux500/ux500_msp_dai.c1
-rw-r--r--sound/usb/midi.c11
-rw-r--r--sound/usb/mixer_quirks.c2
151 files changed, 8066 insertions, 1393 deletions
diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt
new file mode 100644
index 000000000000..15a919522b42
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4613.txt
@@ -0,0 +1,17 @@
+AK4613 I2C transmitter
+
+This device supports I2C mode only.
+
+Required properties:
+
+- compatible : "asahi-kasei,ak4613"
+- reg : The chip select number on the I2C bus
+
+Example:
+
+&i2c {
+ ak4613: ak4613@0x10 {
+ compatible = "asahi-kasei,ak4613";
+ reg = <0x10>;
+ };
+};
diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt
index 623d4e70ae11..340784db6808 100644
--- a/Documentation/devicetree/bindings/sound/ak4642.txt
+++ b/Documentation/devicetree/bindings/sound/ak4642.txt
@@ -7,7 +7,14 @@ Required properties:
- compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
- reg : The chip select number on the I2C bus
-Example:
+Optional properties:
+
+ - #clock-cells : common clock binding; shall be set to 0
+ - clocks : common clock binding; MCKI clock
+ - clock-frequency : common clock binding; frequency of MCKO
+ - clock-output-names : common clock binding; MCKO clock name
+
+Example 1:
&i2c {
ak4648: ak4648@0x12 {
@@ -15,3 +22,16 @@ Example:
reg = <0x12>;
};
};
+
+Example 2:
+
+&i2c {
+ ak4643: codec@12 {
+ compatible = "asahi-kasei,ak4643";
+ reg = <0x12>;
+ #clock-cells = <0>;
+ clocks = <&audio_clock>;
+ clock-frequency = <12288000>;
+ clock-output-names = "ak4643_mcko";
+ };
+};
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 1173395b5e5c..c57cbd65736c 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -4,10 +4,12 @@ Required properties:
- compatible : "renesas,rcar_sound-<soctype>", fallbacks
"renesas,rcar_sound-gen1" if generation1, and
"renesas,rcar_sound-gen2" if generation2
+ "renesas,rcar_sound-gen3" if generation3
Examples with soctypes are:
- "renesas,rcar_sound-r8a7778" (R-Car M1A)
- "renesas,rcar_sound-r8a7790" (R-Car H2)
- "renesas,rcar_sound-r8a7791" (R-Car M2-W)
+ - "renesas,rcar_sound-r8a7795" (R-Car H3)
- reg : Should contain the register physical address.
required register is
SRU/ADG/SSI if generation1
@@ -30,6 +32,11 @@ Required properties:
- rcar_sound,dai : DAI contents.
The number of DAI subnode should be same as HW.
see below for detail.
+- #sound-dai-cells : it must be 0 if your system is using single DAI
+ it must be 1 if your system is using multi DAI
+- #clock-cells : it must be 0 if your system has audio_clkout
+ it must be 1 if your system has audio_clkout0/1/2/3
+- clock-frequency : for all audio_clkout0/1/2/3
SSI subnode properties:
- interrupts : Should contain SSI interrupt for PIO transfer
diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
new file mode 100644
index 000000000000..680144b74ae9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
@@ -0,0 +1,33 @@
+* Allwinner A10 Codec
+
+Required properties:
+- compatible: must be either "allwinner,sun4i-a10-codec" or
+ "allwinner,sun7i-a20-codec"
+- reg: must contain the registers location and length
+- interrupts: must contain the codec interrupt
+- dmas: DMA channels for tx and rx dma. See the DMA client binding,
+ Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "tx" and "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry
+ in clock-names.
+- clock-names: should contain followings:
+ - "apb": the parent APB clock for this controller
+ - "codec": the parent module clock
+- routing : A list of the connections between audio components. Each
+ entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source.
+
+
+Example:
+codec: codec@01c22c00 {
+ #sound-dai-cells = <0>;
+ compatible = "allwinner,sun7i-a20-codec";
+ reg = <0x01c22c00 0x40>;
+ interrupts = <0 30 4>;
+ clocks = <&apb0_gates 0>, <&codec_clk>;
+ clock-names = "apb", "codec";
+ dmas = <&dma 0 19>, <&dma 0 19>;
+ dma-names = "rx", "tx";
+ routing = "Headphone Jack", "HP Right",
+ "Headphone Jack", "HP Left";
+};
diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt
index 6a2c84247f91..34cf70e2cbc4 100644
--- a/Documentation/devicetree/bindings/sound/tdm-slot.txt
+++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt
@@ -4,11 +4,15 @@ This specifies audio DAI's TDM slot.
TDM slot properties:
dai-tdm-slot-num : Number of slots in use.
-dai-tdm-slot-width : Width in bits for each slot.
+dai-tdm-slot-width : Width in bits for each slot.
+dai-tdm-slot-tx-mask : Transmit direction slot mask, optional
+dai-tdm-slot-rx-mask : Receive direction slot mask, optional
For instance:
dai-tdm-slot-num = <2>;
dai-tdm-slot-width = <8>;
+ dai-tdm-slot-tx-mask = <0 1>;
+ dai-tdm-slot-rx-mask = <1 0>;
And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
to specify a explicit mapping of the channels and the slots. If it's absent
@@ -18,3 +22,8 @@ tx and rx masks.
For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
for an active slot as default, and the default active bits are at the LSB of
the masks.
+
+The explicit masks are given as array of integers, where the first
+number presents bit-0 (LSB), second presents bit-1, etc. Any non zero
+number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask()
+does not do anything, if either mask is set non zero value.
diff --git a/drivers/gpu/drm/i915/i915_dma.c b/drivers/gpu/drm/i915/i915_dma.c
index ab37d1121be8..990f656e6ab0 100644
--- a/drivers/gpu/drm/i915/i915_dma.c
+++ b/drivers/gpu/drm/i915/i915_dma.c
@@ -832,6 +832,7 @@ int i915_driver_load(struct drm_device *dev, unsigned long flags)
mutex_init(&dev_priv->sb_lock);
mutex_init(&dev_priv->modeset_restore_lock);
mutex_init(&dev_priv->csr_lock);
+ mutex_init(&dev_priv->av_mutex);
intel_pm_setup(dev);
diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h
index e1db8de52851..22dd7043c9ef 100644
--- a/drivers/gpu/drm/i915/i915_drv.h
+++ b/drivers/gpu/drm/i915/i915_drv.h
@@ -1885,6 +1885,11 @@ struct drm_i915_private {
/* hda/i915 audio component */
struct i915_audio_component *audio_component;
bool audio_component_registered;
+ /**
+ * av_mutex - mutex for audio/video sync
+ *
+ */
+ struct mutex av_mutex;
uint32_t hw_context_size;
struct list_head context_list;
diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c
index 2a5c76faf9f8..ae8df0a43de6 100644
--- a/drivers/gpu/drm/i915/intel_audio.c
+++ b/drivers/gpu/drm/i915/intel_audio.c
@@ -68,6 +68,31 @@ static const struct {
{ 148500, AUD_CONFIG_PIXEL_CLOCK_HDMI_148500 },
};
+/* HDMI N/CTS table */
+#define TMDS_297M 297000
+#define TMDS_296M DIV_ROUND_UP(297000 * 1000, 1001)
+static const struct {
+ int sample_rate;
+ int clock;
+ int n;
+ int cts;
+} aud_ncts[] = {
+ { 44100, TMDS_296M, 4459, 234375 },
+ { 44100, TMDS_297M, 4704, 247500 },
+ { 48000, TMDS_296M, 5824, 281250 },
+ { 48000, TMDS_297M, 5120, 247500 },
+ { 32000, TMDS_296M, 5824, 421875 },
+ { 32000, TMDS_297M, 3072, 222750 },
+ { 88200, TMDS_296M, 8918, 234375 },
+ { 88200, TMDS_297M, 9408, 247500 },
+ { 96000, TMDS_296M, 11648, 281250 },
+ { 96000, TMDS_297M, 10240, 247500 },
+ { 176400, TMDS_296M, 17836, 234375 },
+ { 176400, TMDS_297M, 18816, 247500 },
+ { 192000, TMDS_296M, 23296, 281250 },
+ { 192000, TMDS_297M, 20480, 247500 },
+};
+
/* get AUD_CONFIG_PIXEL_CLOCK_HDMI_* value for mode */
static u32 audio_config_hdmi_pixel_clock(struct drm_display_mode *mode)
{
@@ -90,6 +115,45 @@ static u32 audio_config_hdmi_pixel_clock(struct drm_display_mode *mode)
return hdmi_audio_clock[i].config;
}
+static int audio_config_get_n(const struct drm_display_mode *mode, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(aud_ncts); i++) {
+ if ((rate == aud_ncts[i].sample_rate) &&
+ (mode->clock == aud_ncts[i].clock)) {
+ return aud_ncts[i].n;
+ }
+ }
+ return 0;
+}
+
+static uint32_t audio_config_setup_n_reg(int n, uint32_t val)
+{
+ int n_low, n_up;
+ uint32_t tmp = val;
+
+ n_low = n & 0xfff;
+ n_up = (n >> 12) & 0xff;
+ tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK);
+ tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) |
+ (n_low << AUD_CONFIG_LOWER_N_SHIFT) |
+ AUD_CONFIG_N_PROG_ENABLE);
+ return tmp;
+}
+
+/* check whether N/CTS/M need be set manually */
+static bool audio_rate_need_prog(struct intel_crtc *crtc,
+ const struct drm_display_mode *mode)
+{
+ if (((mode->clock == TMDS_297M) ||
+ (mode->clock == TMDS_296M)) &&
+ intel_pipe_has_type(crtc, INTEL_OUTPUT_HDMI))
+ return true;
+ else
+ return false;
+}
+
static bool intel_eld_uptodate(struct drm_connector *connector,
int reg_eldv, uint32_t bits_eldv,
int reg_elda, uint32_t bits_elda,
@@ -184,6 +248,8 @@ static void hsw_audio_codec_disable(struct intel_encoder *encoder)
DRM_DEBUG_KMS("Disable audio codec on pipe %c\n", pipe_name(pipe));
+ mutex_lock(&dev_priv->av_mutex);
+
/* Disable timestamps */
tmp = I915_READ(HSW_AUD_CFG(pipe));
tmp &= ~AUD_CONFIG_N_VALUE_INDEX;
@@ -199,6 +265,8 @@ static void hsw_audio_codec_disable(struct intel_encoder *encoder)
tmp &= ~AUDIO_ELD_VALID(pipe);
tmp &= ~AUDIO_OUTPUT_ENABLE(pipe);
I915_WRITE(HSW_AUD_PIN_ELD_CP_VLD, tmp);
+
+ mutex_unlock(&dev_priv->av_mutex);
}
static void hsw_audio_codec_enable(struct drm_connector *connector,
@@ -208,13 +276,20 @@ static void hsw_audio_codec_enable(struct drm_connector *connector,
struct drm_i915_private *dev_priv = connector->dev->dev_private;
struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc);
enum pipe pipe = intel_crtc->pipe;
+ struct i915_audio_component *acomp = dev_priv->audio_component;
const uint8_t *eld = connector->eld;
+ struct intel_digital_port *intel_dig_port =
+ enc_to_dig_port(&encoder->base);
+ enum port port = intel_dig_port->port;
uint32_t tmp;
int len, i;
+ int n, rate;
DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n",
pipe_name(pipe), drm_eld_size(eld));
+ mutex_lock(&dev_priv->av_mutex);
+
/* Enable audio presence detect, invalidate ELD */
tmp = I915_READ(HSW_AUD_PIN_ELD_CP_VLD);
tmp |= AUDIO_OUTPUT_ENABLE(pipe);
@@ -246,13 +321,32 @@ static void hsw_audio_codec_enable(struct drm_connector *connector,
/* Enable timestamps */
tmp = I915_READ(HSW_AUD_CFG(pipe));
tmp &= ~AUD_CONFIG_N_VALUE_INDEX;
- tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK;
if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT))
tmp |= AUD_CONFIG_N_VALUE_INDEX;
else
tmp |= audio_config_hdmi_pixel_clock(mode);
+
+ tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
+ if (audio_rate_need_prog(intel_crtc, mode)) {
+ if (!acomp)
+ rate = 0;
+ else if (port >= PORT_A && port <= PORT_E)
+ rate = acomp->aud_sample_rate[port];
+ else {
+ DRM_ERROR("invalid port: %d\n", port);
+ rate = 0;
+ }
+ n = audio_config_get_n(mode, rate);
+ if (n != 0)
+ tmp = audio_config_setup_n_reg(n, tmp);
+ else
+ DRM_DEBUG_KMS("no suitable N value is found\n");
+ }
+
I915_WRITE(HSW_AUD_CFG(pipe), tmp);
+
+ mutex_unlock(&dev_priv->av_mutex);
}
static void ilk_audio_codec_disable(struct intel_encoder *encoder)
@@ -527,12 +621,91 @@ static int i915_audio_component_get_cdclk_freq(struct device *dev)
return ret;
}
+static int i915_audio_component_sync_audio_rate(struct device *dev,
+ int port, int rate)
+{
+ struct drm_i915_private *dev_priv = dev_to_i915(dev);
+ struct drm_device *drm_dev = dev_priv->dev;
+ struct intel_encoder *intel_encoder;
+ struct intel_digital_port *intel_dig_port;
+ struct intel_crtc *crtc;
+ struct drm_display_mode *mode;
+ struct i915_audio_component *acomp = dev_priv->audio_component;
+ enum pipe pipe = -1;
+ u32 tmp;
+ int n;
+
+ /* HSW, BDW SKL need this fix */
+ if (!IS_SKYLAKE(dev_priv) &&
+ !IS_BROADWELL(dev_priv) &&
+ !IS_HASWELL(dev_priv))
+ return 0;
+
+ mutex_lock(&dev_priv->av_mutex);
+ /* 1. get the pipe */
+ for_each_intel_encoder(drm_dev, intel_encoder) {
+ if (intel_encoder->type != INTEL_OUTPUT_HDMI)
+ continue;
+ intel_dig_port = enc_to_dig_port(&intel_encoder->base);
+ if (port == intel_dig_port->port) {
+ crtc = to_intel_crtc(intel_encoder->base.crtc);
+ if (!crtc) {
+ DRM_DEBUG_KMS("%s: crtc is NULL\n", __func__);
+ continue;
+ }
+ pipe = crtc->pipe;
+ break;
+ }
+ }
+
+ if (pipe == INVALID_PIPE) {
+ DRM_DEBUG_KMS("no pipe for the port %c\n", port_name(port));
+ mutex_unlock(&dev_priv->av_mutex);
+ return -ENODEV;
+ }
+ DRM_DEBUG_KMS("pipe %c connects port %c\n",
+ pipe_name(pipe), port_name(port));
+ mode = &crtc->config->base.adjusted_mode;
+
+ /* port must be valid now, otherwise the pipe will be invalid */
+ acomp->aud_sample_rate[port] = rate;
+
+ /* 2. check whether to set the N/CTS/M manually or not */
+ if (!audio_rate_need_prog(crtc, mode)) {
+ tmp = I915_READ(HSW_AUD_CFG(pipe));
+ tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
+ I915_WRITE(HSW_AUD_CFG(pipe), tmp);
+ mutex_unlock(&dev_priv->av_mutex);
+ return 0;
+ }
+
+ n = audio_config_get_n(mode, rate);
+ if (n == 0) {
+ DRM_DEBUG_KMS("Using automatic mode for N value on port %c\n",
+ port_name(port));
+ tmp = I915_READ(HSW_AUD_CFG(pipe));
+ tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
+ I915_WRITE(HSW_AUD_CFG(pipe), tmp);
+ mutex_unlock(&dev_priv->av_mutex);
+ return 0;
+ }
+
+ /* 3. set the N/CTS/M */
+ tmp = I915_READ(HSW_AUD_CFG(pipe));
+ tmp = audio_config_setup_n_reg(n, tmp);
+ I915_WRITE(HSW_AUD_CFG(pipe), tmp);
+
+ mutex_unlock(&dev_priv->av_mutex);
+ return 0;
+}
+
static const struct i915_audio_component_ops i915_audio_component_ops = {
.owner = THIS_MODULE,
.get_power = i915_audio_component_get_power,
.put_power = i915_audio_component_put_power,
.codec_wake_override = i915_audio_component_codec_wake_override,
.get_cdclk_freq = i915_audio_component_get_cdclk_freq,
+ .sync_audio_rate = i915_audio_component_sync_audio_rate,
};
static int i915_audio_component_bind(struct device *i915_dev,
@@ -540,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev,
{
struct i915_audio_component *acomp = data;
struct drm_i915_private *dev_priv = dev_to_i915(i915_dev);
+ int i;
if (WARN_ON(acomp->ops || acomp->dev))
return -EEXIST;
@@ -547,6 +721,9 @@ static int i915_audio_component_bind(struct device *i915_dev,
drm_modeset_lock_all(dev_priv->dev);
acomp->ops = &i915_audio_component_ops;
acomp->dev = i915_dev;
+ BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS);
+ for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++)
+ acomp->aud_sample_rate[i] = 0;
dev_priv->audio_component = acomp;
drm_modeset_unlock_all(dev_priv->dev);
diff --git a/drivers/spi/spi-atmel.c b/drivers/spi/spi-atmel.c
index 63318e2afba1..41e37a6a1368 100644
--- a/drivers/spi/spi-atmel.c
+++ b/drivers/spi/spi-atmel.c
@@ -871,14 +871,7 @@ static int atmel_spi_set_xfer_speed(struct atmel_spi *as,
* Calculate the lowest divider that satisfies the
* constraint, assuming div32/fdiv/mbz == 0.
*/
- if (xfer->speed_hz)
- scbr = DIV_ROUND_UP(bus_hz, xfer->speed_hz);
- else
- /*
- * This can happend if max_speed is null.
- * In this case, we set the lowest possible speed
- */
- scbr = 0xff;
+ scbr = DIV_ROUND_UP(bus_hz, xfer->speed_hz);
/*
* If the resulting divider doesn't fit into the
@@ -1300,14 +1293,12 @@ static int atmel_spi_one_transfer(struct spi_master *master,
return -EINVAL;
}
- if (xfer->bits_per_word) {
- asd = spi->controller_state;
- bits = (asd->csr >> 4) & 0xf;
- if (bits != xfer->bits_per_word - 8) {
- dev_dbg(&spi->dev,
+ asd = spi->controller_state;
+ bits = (asd->csr >> 4) & 0xf;
+ if (bits != xfer->bits_per_word - 8) {
+ dev_dbg(&spi->dev,
"you can't yet change bits_per_word in transfers\n");
- return -ENOPROTOOPT;
- }
+ return -ENOPROTOOPT;
}
/*
diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h
index b2d56dd483d9..89dc7d6bc1cc 100644
--- a/include/drm/i915_component.h
+++ b/include/drm/i915_component.h
@@ -24,8 +24,18 @@
#ifndef _I915_COMPONENT_H_
#define _I915_COMPONENT_H_
+/* MAX_PORT is the number of port
+ * It must be sync with I915_MAX_PORTS defined i915_drv.h
+ * 5 should be enough as only HSW, BDW, SKL need such fix.
+ */
+#define MAX_PORTS 5
+
struct i915_audio_component {
struct device *dev;
+ /**
+ * @aud_sample_rate: the array of audio sample rate per port
+ */
+ int aud_sample_rate[MAX_PORTS];
const struct i915_audio_component_ops {
struct module *owner;
@@ -33,6 +43,13 @@ struct i915_audio_component {
void (*put_power)(struct device *);
void (*codec_wake_override)(struct device *, bool enable);
int (*get_cdclk_freq)(struct device *);
+ /**
+ * @sync_audio_rate: set n/cts based on the sample rate
+ *
+ * Called from audio driver. After audio driver sets the
+ * sample rate, it will call this function to set n/cts
+ */
+ int (*sync_audio_rate)(struct device *, int port, int rate);
} *ops;
const struct i915_audio_component_audio_ops {
diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h
index df705908480a..2767c55a641e 100644
--- a/include/sound/hda_regmap.h
+++ b/include/sound/hda_regmap.h
@@ -67,7 +67,7 @@ int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg,
* @reg: verb to write
* @val: value to write
*
- * For writing an amp value, use snd_hda_regmap_amp_update().
+ * For writing an amp value, use snd_hdac_regmap_update_amp().
*/
static inline int
snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid,
@@ -85,7 +85,7 @@ snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid,
* @mask: bit mask to update
* @val: value to update
*
- * For updating an amp value, use snd_hda_regmap_amp_update().
+ * For updating an amp value, use snd_hdac_regmap_update_amp().
*/
static inline int
snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid,
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 49bc836fcd84..26e956f4b7c6 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -147,6 +147,12 @@ int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid,
bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid,
unsigned int format);
+int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm);
+int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm);
+bool snd_hdac_check_power_state(struct hdac_device *hdac,
+ hda_nid_t nid, unsigned int target_state);
/**
* snd_hdac_read_parm - read a codec parameter
* @codec: the codec object
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 691e7ee0a510..a4fcc9456194 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -285,8 +285,6 @@ struct snd_pcm_hw_constraint_ranges {
unsigned int mask;
};
-struct snd_pcm_hwptr_log;
-
/*
* userspace-provided audio timestamp config to kernel,
* structure is for internal use only and filled with dedicated unpack routine
@@ -428,10 +426,6 @@ struct snd_pcm_runtime {
/* -- OSS things -- */
struct snd_pcm_oss_runtime oss;
#endif
-
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
- struct snd_pcm_hwptr_log *hwptr_log;
-#endif
};
struct snd_pcm_group { /* keep linked substreams */
diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h
index bb7b2ebfee7b..d8e33d38da43 100644
--- a/include/sound/rcar_snd.h
+++ b/include/sound/rcar_snd.h
@@ -12,7 +12,6 @@
#ifndef RCAR_SND_H
#define RCAR_SND_H
-#include <linux/sh_clk.h>
#define RSND_GEN1_SRU 0
#define RSND_GEN1_ADG 1
diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h
index 22734bc3ffd4..a5cf6152e778 100644
--- a/include/sound/rt5645.h
+++ b/include/sound/rt5645.h
@@ -21,6 +21,8 @@ struct rt5645_platform_data {
/* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */
unsigned int jd_mode;
+ /* Invert JD when jack insert */
+ bool jd_invert;
};
#endif
diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h
index b9b4f289fe6b..0399352f3a62 100644
--- a/include/sound/simple_card.h
+++ b/include/sound/simple_card.h
@@ -19,6 +19,8 @@ struct asoc_simple_dai {
unsigned int sysclk;
int slots;
int slot_width;
+ unsigned int tx_slot_mask;
+ unsigned int rx_slot_mask;
struct clk *clk;
};
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 884e728b09d9..470f20887b61 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -226,6 +226,18 @@
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) }
+#define SOC_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_range, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xreg, .shift = xshift, \
+ .rshift = xshift, .min = xmin, .max = xmax, \
+ .platform_max = xmax, .invert = xinvert} }
#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@@ -1601,6 +1613,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card,
int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
const char *propname);
int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *tx_mask,
+ unsigned int *rx_mask,
unsigned int *slots,
unsigned int *slot_width);
void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card,
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index a45be6bdcf5b..a82108e5d1c0 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -100,9 +100,11 @@ enum {
SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */
SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */
SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */
+ SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */
+ SNDRV_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */
/* Don't forget to change the following: */
- SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_OXFW
+ SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM
};
struct snd_hwdep_info {
diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h
index 49122df3b56b..db79a12fcc78 100644
--- a/include/uapi/sound/firewire.h
+++ b/include/uapi/sound/firewire.h
@@ -9,6 +9,7 @@
#define SNDRV_FIREWIRE_EVENT_LOCK_STATUS 0x000010cc
#define SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION 0xd1ce004e
#define SNDRV_FIREWIRE_EVENT_EFW_RESPONSE 0x4e617475
+#define SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE 0x746e736c
struct snd_firewire_event_common {
unsigned int type; /* SNDRV_FIREWIRE_EVENT_xxx */
@@ -40,11 +41,17 @@ struct snd_firewire_event_efw_response {
__be32 response[0]; /* some responses */
};
+struct snd_firewire_event_digi00x_message {
+ unsigned int type;
+ __u32 message; /* Digi00x-specific message */
+};
+
union snd_firewire_event {
struct snd_firewire_event_common common;
struct snd_firewire_event_lock_status lock_status;
struct snd_firewire_event_dice_notification dice_notification;
struct snd_firewire_event_efw_response efw_response;
+ struct snd_firewire_event_digi00x_message digi00x_message;
};
@@ -56,6 +63,8 @@ union snd_firewire_event {
#define SNDRV_FIREWIRE_TYPE_FIREWORKS 2
#define SNDRV_FIREWIRE_TYPE_BEBOB 3
#define SNDRV_FIREWIRE_TYPE_OXFW 4
+#define SNDRV_FIREWIRE_TYPE_DIGI00X 5
+#define SNDRV_FIREWIRE_TYPE_TASCAM 6
/* RME, MOTU, ... */
struct snd_firewire_get_info {
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 02bd96954dc4..308c9ecf73db 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -1014,9 +1014,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream)
snd_free_pages((void*)runtime->control,
PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)));
kfree(runtime->hw_constraints.rules);
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
- kfree(runtime->hwptr_log);
-#endif
kfree(runtime);
substream->runtime = NULL;
put_pid(substream->pid);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 75888dd38a7f..139887011ba2 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -650,7 +650,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream,
}
snd_pcm_stream_unlock_irq(substream);
- if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
+ if (params->tstamp_mode < 0 ||
+ params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
return -EINVAL;
if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) &&
params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)
@@ -2226,7 +2227,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream)
snd_pcm_drop(substream);
if (substream->hw_opened) {
- if (substream->ops->hw_free != NULL)
+ if (substream->ops->hw_free &&
+ substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
substream->ops->hw_free(substream);
substream->ops->close(substream);
substream->hw_opened = 0;
diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c
index ccd893566f1d..046cb586fb2f 100644
--- a/sound/core/seq/oss/seq_oss_readq.c
+++ b/sound/core/seq/oss/seq_oss_readq.c
@@ -91,8 +91,7 @@ snd_seq_oss_readq_clear(struct seq_oss_readq *q)
q->head = q->tail = 0;
}
/* if someone sleeping, wake'em up */
- if (waitqueue_active(&q->midi_sleep))
- wake_up(&q->midi_sleep);
+ wake_up(&q->midi_sleep);
q->input_time = (unsigned long)-1;
}
@@ -138,8 +137,7 @@ snd_seq_oss_readq_put_event(struct seq_oss_readq *q, union evrec *ev)
q->qlen++;
/* wake up sleeper */
- if (waitqueue_active(&q->midi_sleep))
- wake_up(&q->midi_sleep);
+ wake_up(&q->midi_sleep);
spin_unlock_irqrestore(&q->lock, flags);
diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c
index d50338bbc21f..1f6788a18444 100644
--- a/sound/core/seq/oss/seq_oss_writeq.c
+++ b/sound/core/seq/oss/seq_oss_writeq.c
@@ -138,9 +138,7 @@ snd_seq_oss_writeq_wakeup(struct seq_oss_writeq *q, abstime_t time)
spin_lock_irqsave(&q->sync_lock, flags);
q->sync_time = time;
q->sync_event_put = 0;
- if (waitqueue_active(&q->sync_sleep)) {
- wake_up(&q->sync_sleep);
- }
+ wake_up(&q->sync_sleep);
spin_unlock_irqrestore(&q->sync_lock, flags);
}
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 8850b7de1d38..bee0e5f1a116 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -120,4 +120,31 @@ config SND_BEBOB
To compile this driver as a module, choose M here: the module
will be called snd-bebob.
+config SND_FIREWIRE_DIGI00X
+ tristate "Digidesign Digi 002/003 family support"
+ select SND_FIREWIRE_LIB
+ select SND_HWDEP
+ help
+ Say Y here to include support for Digidesign Digi 002/003 family.
+ * Digi 002 Console
+ * Digi 002 Rack
+ * Digi 003 Console
+ * Digi 003 Rack
+ * Digi 003 Rack+
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-firewire-digi00x.
+
+config SND_FIREWIRE_TASCAM
+ tristate "TASCAM FireWire series support"
+ select SND_FIREWIRE_LIB
+ select SND_HWDEP
+ help
+ Say Y here to include support for TASCAM.
+ * FW-1884
+ * FW-1082
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-firewire-tascam.
+
endif # SND_FIREWIRE
diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile
index 8b37f084b2ab..6ae50f50db62 100644
--- a/sound/firewire/Makefile
+++ b/sound/firewire/Makefile
@@ -1,5 +1,5 @@
snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \
- fcp.o cmp.o amdtp.o
+ fcp.o cmp.o amdtp-stream.o amdtp-am824.o
snd-oxfw-objs := oxfw.o
snd-isight-objs := isight.o
snd-scs1x-objs := scs1x.o
@@ -11,3 +11,5 @@ obj-$(CONFIG_SND_ISIGHT) += snd-isight.o
obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o
obj-$(CONFIG_SND_FIREWORKS) += fireworks/
obj-$(CONFIG_SND_BEBOB) += bebob/
+obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += digi00x/
+obj-$(CONFIG_SND_FIREWIRE_TASCAM) += tascam/
diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
new file mode 100644
index 000000000000..bebddc60fde8
--- /dev/null
+++ b/sound/firewire/amdtp-am824.c
@@ -0,0 +1,465 @@
+/*
+ * AM824 format in Audio and Music Data Transmission Protocol (IEC 61883-6)
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Copyright (c) 2015 Takashi Sakamoto <o-takashi@sakamocchi.jp>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/slab.h>
+
+#include "amdtp-am824.h"
+
+#define CIP_FMT_AM 0x10
+
+/* "Clock-based rate control mode" is just supported. */
+#define AMDTP_FDF_AM824 0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND 3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS 8
+
+struct amdtp_am824 {
+ struct snd_rawmidi_substream *midi[AM824_MAX_CHANNELS_FOR_MIDI * 8];
+ int midi_fifo_limit;
+ int midi_fifo_used[AM824_MAX_CHANNELS_FOR_MIDI * 8];
+ unsigned int pcm_channels;
+ unsigned int midi_ports;
+
+ u8 pcm_positions[AM824_MAX_CHANNELS_FOR_PCM];
+ u8 midi_position;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+
+ unsigned int frame_multiplier;
+};
+
+/**
+ * amdtp_am824_set_parameters - set stream parameters
+ * @s: the AMDTP stream to configure
+ * @rate: the sample rate
+ * @pcm_channels: the number of PCM samples in each data block, to be encoded
+ * as AM824 multi-bit linear audio
+ * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ * @double_pcm_frames: one data block transfers two PCM frames
+ *
+ * The parameters must be set before the stream is started, and must not be
+ * changed while the stream is running.
+ */
+int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels,
+ unsigned int midi_ports,
+ bool double_pcm_frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int midi_channels;
+ unsigned int i;
+ int err;
+
+ if (amdtp_stream_running(s))
+ return -EINVAL;
+
+ if (pcm_channels > AM824_MAX_CHANNELS_FOR_PCM)
+ return -EINVAL;
+
+ midi_channels = DIV_ROUND_UP(midi_ports, 8);
+ if (midi_channels > AM824_MAX_CHANNELS_FOR_MIDI)
+ return -EINVAL;
+
+ if (WARN_ON(amdtp_stream_running(s)) ||
+ WARN_ON(pcm_channels > AM824_MAX_CHANNELS_FOR_PCM) ||
+ WARN_ON(midi_channels > AM824_MAX_CHANNELS_FOR_MIDI))
+ return -EINVAL;
+
+ err = amdtp_stream_set_parameters(s, rate,
+ pcm_channels + midi_channels);
+ if (err < 0)
+ return err;
+
+ s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+ p->pcm_channels = pcm_channels;
+ p->midi_ports = midi_ports;
+
+ /*
+ * In IEC 61883-6, one data block represents one event. In ALSA, one
+ * event equals to one PCM frame. But Dice has a quirk at higher
+ * sampling rate to transfer two PCM frames in one data block.
+ */
+ if (double_pcm_frames)
+ p->frame_multiplier = 2;
+ else
+ p->frame_multiplier = 1;
+
+ /* init the position map for PCM and MIDI channels */
+ for (i = 0; i < pcm_channels; i++)
+ p->pcm_positions[i] = i;
+ p->midi_position = p->pcm_channels;
+
+ /*
+ * We do not know the actual MIDI FIFO size of most devices. Just
+ * assume two bytes, i.e., one byte can be received over the bus while
+ * the previous one is transmitted over MIDI.
+ * (The value here is adjusted for midi_ratelimit_per_packet().)
+ */
+ p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_parameters);
+
+/**
+ * amdtp_am824_set_pcm_position - set an index of data channel for a channel
+ * of PCM frame
+ * @s: the AMDTP stream
+ * @index: the index of data channel in an data block
+ * @position: the channel of PCM frame
+ */
+void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index,
+ unsigned int position)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (index < p->pcm_channels)
+ p->pcm_positions[index] = position;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_position);
+
+/**
+ * amdtp_am824_set_midi_position - set a index of data channel for MIDI
+ * conformant data channel
+ * @s: the AMDTP stream
+ * @position: the index of data channel in an data block
+ */
+void amdtp_am824_set_midi_position(struct amdtp_stream *s,
+ unsigned int position)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ p->midi_position = position;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_midi_position);
+
+static void write_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[p->pcm_positions[c]] =
+ cpu_to_be32((*src >> 8) | 0x40000000);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[p->pcm_positions[c]] =
+ cpu_to_be32((*src << 8) | 0x42000000);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[p->pcm_positions[c]]) << 8;
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int i, c, channels = p->pcm_channels;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[p->pcm_positions[c]] = cpu_to_be32(0x40000000);
+ buffer += s->data_block_quadlets;
+ }
+}
+
+/**
+ * amdtp_am824_set_pcm_format - set the PCM format
+ * @s: the AMDTP stream to configure
+ * @format: the format of the ALSA PCM device
+ *
+ * The sample format must be set after the other parameters (rate/PCM channels/
+ * MIDI) and before the stream is started, and must not be changed while the
+ * stream is running.
+ */
+void amdtp_am824_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_format);
+
+/**
+ * amdtp_am824_add_pcm_hw_constraints - add hw constraints for PCM substream
+ * @s: the AMDTP stream for AM824 data block, must be initialized.
+ * @runtime: the PCM substream runtime
+ *
+ */
+int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ if (err < 0)
+ return err;
+
+ /* AM824 in IEC 61883-6 can deliver 24bit data. */
+ return snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_add_pcm_hw_constraints);
+
+/**
+ * amdtp_am824_midi_trigger - start/stop playback/capture with a MIDI device
+ * @s: the AMDTP stream
+ * @port: index of MIDI port
+ * @midi: the MIDI device to be started, or %NULL to stop the current device
+ *
+ * Call this function on a running isochronous stream to enable the actual
+ * transmission of MIDI data. This function should be called from the MIDI
+ * device's .trigger callback.
+ */
+void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (port < p->midi_ports)
+ ACCESS_ONCE(p->midi[port]) = midi;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_midi_trigger);
+
+/*
+ * To avoid sending MIDI bytes at too high a rate, assume that the receiving
+ * device has a FIFO, and track how much it is filled. This values increases
+ * by one whenever we send one byte in a packet, but the FIFO empties at
+ * a constant rate independent of our packet rate. One packet has syt_interval
+ * samples, so the number of bytes that empty out of the FIFO, per packet(!),
+ * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing
+ * fractional values, the values in midi_fifo_used[] are measured in bytes
+ * multiplied by the sample rate.
+ */
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_am824 *p = s->protocol;
+ int used;
+
+ used = p->midi_fifo_used[port];
+ if (used == 0) /* common shortcut */
+ return true;
+
+ used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+ used = max(used, 0);
+ p->midi_fifo_used[port] = used;
+
+ return used < p->midi_fifo_limit;
+}
+
+static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ p->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int f, port;
+ u8 *b;
+
+ for (f = 0; f < frames; f++) {
+ b = (u8 *)&buffer[p->midi_position];
+
+ port = (s->data_block_counter + f) % 8;
+ if (f < MAX_MIDI_RX_BLOCKS &&
+ midi_ratelimit_per_packet(s, port) &&
+ p->midi[port] != NULL &&
+ snd_rawmidi_transmit(p->midi[port], &b[1], 1) == 1) {
+ midi_rate_use_one_byte(s, port);
+ b[0] = 0x81;
+ } else {
+ b[0] = 0x80;
+ b[1] = 0;
+ }
+ b[2] = 0;
+ b[3] = 0;
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static void read_midi_messages(struct amdtp_stream *s,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int f, port;
+ int len;
+ u8 *b;
+
+ for (f = 0; f < frames; f++) {
+ port = (s->data_block_counter + f) % 8;
+ b = (u8 *)&buffer[p->midi_position];
+
+ len = b[0] - 0x80;
+ if ((1 <= len) && (len <= 3) && (p->midi[port]))
+ snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks, unsigned int *syt)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+ unsigned int pcm_frames;
+
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks * p->frame_multiplier;
+ } else {
+ write_pcm_silence(s, buffer, data_blocks);
+ pcm_frames = 0;
+ }
+
+ if (p->midi_ports)
+ write_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks, unsigned int *syt)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+ unsigned int pcm_frames;
+
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks * p->frame_multiplier;
+ } else {
+ pcm_frames = 0;
+ }
+
+ if (p->midi_ports)
+ read_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+/**
+ * amdtp_am824_init - initialize an AMDTP stream structure to handle AM824
+ * data block
+ * @s: the AMDTP stream to initialize
+ * @unit: the target of the stream
+ * @dir: the direction of stream
+ * @flags: the packet transmission method to use
+ */
+int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, enum cip_flags flags)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+
+ if (dir == AMDTP_IN_STREAM)
+ process_data_blocks = process_tx_data_blocks;
+ else
+ process_data_blocks = process_rx_data_blocks;
+
+ return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+ process_data_blocks,
+ sizeof(struct amdtp_am824));
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_init);
diff --git a/sound/firewire/amdtp-am824.h b/sound/firewire/amdtp-am824.h
new file mode 100644
index 000000000000..73b07b3109db
--- /dev/null
+++ b/sound/firewire/amdtp-am824.h
@@ -0,0 +1,52 @@
+#ifndef SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED
+#define SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED
+
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+
+#include "amdtp-stream.h"
+
+#define AM824_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32
+
+#define AM824_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \
+ SNDRV_PCM_FMTBIT_S32)
+
+/*
+ * This module supports maximum 64 PCM channels for one PCM stream
+ * This is for our convenience.
+ */
+#define AM824_MAX_CHANNELS_FOR_PCM 64
+
+/*
+ * AMDTP packet can include channels for MIDI conformant data.
+ * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
+ * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
+ *
+ * This module supports maximum 1 MIDI conformant data channels.
+ * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
+ */
+#define AM824_MAX_CHANNELS_FOR_MIDI 1
+
+int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels,
+ unsigned int midi_ports,
+ bool double_pcm_frames);
+
+void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index,
+ unsigned int position);
+
+void amdtp_am824_set_midi_position(struct amdtp_stream *s,
+ unsigned int position);
+
+int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+
+void amdtp_am824_set_pcm_format(struct amdtp_stream *s,
+ snd_pcm_format_t format);
+
+void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi);
+
+int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, enum cip_flags flags);
+#endif
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp-stream.c
index 2a153d260836..ed2902609a4c 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp-stream.c
@@ -11,28 +11,14 @@
#include <linux/firewire.h>
#include <linux/module.h>
#include <linux/slab.h>
-#include <linux/sched.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/rawmidi.h>
-#include "amdtp.h"
+#include "amdtp-stream.h"
#define TICKS_PER_CYCLE 3072
#define CYCLES_PER_SECOND 8000
#define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND)
-/*
- * Nominally 3125 bytes/second, but the MIDI port's clock might be
- * 1% too slow, and the bus clock 100 ppm too fast.
- */
-#define MIDI_BYTES_PER_SECOND 3093
-
-/*
- * Several devices look only at the first eight data blocks.
- * In any case, this is more than enough for the MIDI data rate.
- */
-#define MAX_MIDI_RX_BLOCKS 8
-
#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */
/* isochronous header parameters */
@@ -55,12 +41,8 @@
#define CIP_SYT_MASK 0x0000ffff
#define CIP_SYT_NO_INFO 0xffff
-/*
- * Audio and Music transfer protocol specific parameters
- * only "Clock-based rate control mode" is supported
- */
-#define CIP_FMT_AM (0x10 << CIP_FMT_SHIFT)
-#define AMDTP_FDF_AM824 (0 << (CIP_FDF_SHIFT + 3))
+/* Audio and Music transfer protocol specific parameters */
+#define CIP_FMT_AM 0x10
#define AMDTP_FDF_NO_DATA 0xff
/* TODO: make these configurable */
@@ -78,10 +60,23 @@ static void pcm_period_tasklet(unsigned long data);
* @unit: the target of the stream
* @dir: the direction of stream
* @flags: the packet transmission method to use
+ * @fmt: the value of fmt field in CIP header
+ * @process_data_blocks: callback handler to process data blocks
+ * @protocol_size: the size to allocate newly for protocol
*/
int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
- enum amdtp_stream_direction dir, enum cip_flags flags)
+ enum amdtp_stream_direction dir, enum cip_flags flags,
+ unsigned int fmt,
+ amdtp_stream_process_data_blocks_t process_data_blocks,
+ unsigned int protocol_size)
{
+ if (process_data_blocks == NULL)
+ return -EINVAL;
+
+ s->protocol = kzalloc(protocol_size, GFP_KERNEL);
+ if (!s->protocol)
+ return -ENOMEM;
+
s->unit = unit;
s->direction = dir;
s->flags = flags;
@@ -94,6 +89,9 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
s->callbacked = false;
s->sync_slave = NULL;
+ s->fmt = fmt;
+ s->process_data_blocks = process_data_blocks;
+
return 0;
}
EXPORT_SYMBOL(amdtp_stream_init);
@@ -105,6 +103,7 @@ EXPORT_SYMBOL(amdtp_stream_init);
void amdtp_stream_destroy(struct amdtp_stream *s)
{
WARN_ON(amdtp_stream_running(s));
+ kfree(s->protocol);
mutex_destroy(&s->mutex);
}
EXPORT_SYMBOL(amdtp_stream_destroy);
@@ -141,11 +140,6 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
{
int err;
- /* AM824 in IEC 61883-6 can deliver 24bit data */
- err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
- if (err < 0)
- goto end;
-
/*
* Currently firewire-lib processes 16 packets in one software
* interrupt callback. This equals to 2msec but actually the
@@ -190,39 +184,25 @@ EXPORT_SYMBOL(amdtp_stream_add_pcm_hw_constraints);
* amdtp_stream_set_parameters - set stream parameters
* @s: the AMDTP stream to configure
* @rate: the sample rate
- * @pcm_channels: the number of PCM samples in each data block, to be encoded
- * as AM824 multi-bit linear audio
- * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ * @data_block_quadlets: the size of a data block in quadlet unit
*
* The parameters must be set before the stream is started, and must not be
* changed while the stream is running.
*/
-void amdtp_stream_set_parameters(struct amdtp_stream *s,
- unsigned int rate,
- unsigned int pcm_channels,
- unsigned int midi_ports)
+int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int data_block_quadlets)
{
- unsigned int i, sfc, midi_channels;
+ unsigned int sfc;
- midi_channels = DIV_ROUND_UP(midi_ports, 8);
-
- if (WARN_ON(amdtp_stream_running(s)) |
- WARN_ON(pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM) |
- WARN_ON(midi_channels > AMDTP_MAX_CHANNELS_FOR_MIDI))
- return;
-
- for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc)
+ for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc) {
if (amdtp_rate_table[sfc] == rate)
- goto sfc_found;
- WARN_ON(1);
- return;
+ break;
+ }
+ if (sfc == ARRAY_SIZE(amdtp_rate_table))
+ return -EINVAL;
-sfc_found:
- s->pcm_channels = pcm_channels;
s->sfc = sfc;
- s->data_block_quadlets = s->pcm_channels + midi_channels;
- s->midi_ports = midi_ports;
-
+ s->data_block_quadlets = data_block_quadlets;
s->syt_interval = amdtp_syt_intervals[sfc];
/* default buffering in the device */
@@ -231,18 +211,7 @@ sfc_found:
/* additional buffering needed to adjust for no-data packets */
s->transfer_delay += TICKS_PER_SECOND * s->syt_interval / rate;
- /* init the position map for PCM and MIDI channels */
- for (i = 0; i < pcm_channels; i++)
- s->pcm_positions[i] = i;
- s->midi_position = s->pcm_channels;
-
- /*
- * We do not know the actual MIDI FIFO size of most devices. Just
- * assume two bytes, i.e., one byte can be received over the bus while
- * the previous one is transmitted over MIDI.
- * (The value here is adjusted for midi_ratelimit_per_packet().)
- */
- s->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+ return 0;
}
EXPORT_SYMBOL(amdtp_stream_set_parameters);
@@ -264,52 +233,6 @@ unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s)
}
EXPORT_SYMBOL(amdtp_stream_get_max_payload);
-static void write_pcm_s16(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-static void write_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-static void read_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-
-/**
- * amdtp_stream_set_pcm_format - set the PCM format
- * @s: the AMDTP stream to configure
- * @format: the format of the ALSA PCM device
- *
- * The sample format must be set after the other parameters (rate/PCM channels/
- * MIDI) and before the stream is started, and must not be changed while the
- * stream is running.
- */
-void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
- snd_pcm_format_t format)
-{
- if (WARN_ON(amdtp_stream_pcm_running(s)))
- return;
-
- switch (format) {
- default:
- WARN_ON(1);
- /* fall through */
- case SNDRV_PCM_FORMAT_S16:
- if (s->direction == AMDTP_OUT_STREAM) {
- s->transfer_samples = write_pcm_s16;
- break;
- }
- WARN_ON(1);
- /* fall through */
- case SNDRV_PCM_FORMAT_S32:
- if (s->direction == AMDTP_OUT_STREAM)
- s->transfer_samples = write_pcm_s32;
- else
- s->transfer_samples = read_pcm_s32;
- break;
- }
-}
-EXPORT_SYMBOL(amdtp_stream_set_pcm_format);
-
/**
* amdtp_stream_pcm_prepare - prepare PCM device for running
* @s: the AMDTP stream
@@ -412,182 +335,12 @@ static unsigned int calculate_syt(struct amdtp_stream *s,
}
}
-static void write_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- const u32 *src;
-
- channels = s->pcm_channels;
- src = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- buffer[s->pcm_positions[c]] =
- cpu_to_be32((*src >> 8) | 0x40000000);
- src++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- src = (void *)runtime->dma_area;
- }
-}
-
-static void write_pcm_s16(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- const u16 *src;
-
- channels = s->pcm_channels;
- src = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- buffer[s->pcm_positions[c]] =
- cpu_to_be32((*src << 8) | 0x42000000);
- src++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- src = (void *)runtime->dma_area;
- }
-}
-
-static void read_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- u32 *dst;
-
- channels = s->pcm_channels;
- dst = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- *dst = be32_to_cpu(buffer[s->pcm_positions[c]]) << 8;
- dst++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- dst = (void *)runtime->dma_area;
- }
-}
-
-static void write_pcm_silence(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int i, c;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < s->pcm_channels; ++c)
- buffer[s->pcm_positions[c]] = cpu_to_be32(0x40000000);
- buffer += s->data_block_quadlets;
- }
-}
-
-/*
- * To avoid sending MIDI bytes at too high a rate, assume that the receiving
- * device has a FIFO, and track how much it is filled. This values increases
- * by one whenever we send one byte in a packet, but the FIFO empties at
- * a constant rate independent of our packet rate. One packet has syt_interval
- * samples, so the number of bytes that empty out of the FIFO, per packet(!),
- * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing
- * fractional values, the values in midi_fifo_used[] are measured in bytes
- * multiplied by the sample rate.
- */
-static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
-{
- int used;
-
- used = s->midi_fifo_used[port];
- if (used == 0) /* common shortcut */
- return true;
-
- used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
- used = max(used, 0);
- s->midi_fifo_used[port] = used;
-
- return used < s->midi_fifo_limit;
-}
-
-static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
-{
- s->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
-}
-
-static void write_midi_messages(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int f, port;
- u8 *b;
-
- for (f = 0; f < frames; f++) {
- b = (u8 *)&buffer[s->midi_position];
-
- port = (s->data_block_counter + f) % 8;
- if (f < MAX_MIDI_RX_BLOCKS &&
- midi_ratelimit_per_packet(s, port) &&
- s->midi[port] != NULL &&
- snd_rawmidi_transmit(s->midi[port], &b[1], 1) == 1) {
- midi_rate_use_one_byte(s, port);
- b[0] = 0x81;
- } else {
- b[0] = 0x80;
- b[1] = 0;
- }
- b[2] = 0;
- b[3] = 0;
-
- buffer += s->data_block_quadlets;
- }
-}
-
-static void read_midi_messages(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int f, port;
- int len;
- u8 *b;
-
- for (f = 0; f < frames; f++) {
- port = (s->data_block_counter + f) % 8;
- b = (u8 *)&buffer[s->midi_position];
-
- len = b[0] - 0x80;
- if ((1 <= len) && (len <= 3) && (s->midi[port]))
- snd_rawmidi_receive(s->midi[port], b + 1, len);
-
- buffer += s->data_block_quadlets;
- }
-}
-
static void update_pcm_pointers(struct amdtp_stream *s,
struct snd_pcm_substream *pcm,
unsigned int frames)
{
unsigned int ptr;
- /*
- * In IEC 61883-6, one data block represents one event. In ALSA, one
- * event equals to one PCM frame. But Dice has a quirk to transfer
- * two PCM frames in one data block.
- */
- if (s->double_pcm_frames)
- frames *= 2;
-
ptr = s->pcm_buffer_pointer + frames;
if (ptr >= pcm->runtime->buffer_size)
ptr -= pcm->runtime->buffer_size;
@@ -656,23 +409,19 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
{
__be32 *buffer;
unsigned int payload_length;
+ unsigned int pcm_frames;
struct snd_pcm_substream *pcm;
buffer = s->buffer.packets[s->packet_index].buffer;
+ pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt);
+
buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
(s->data_block_quadlets << CIP_DBS_SHIFT) |
s->data_block_counter);
- buffer[1] = cpu_to_be32(CIP_EOH | CIP_FMT_AM | AMDTP_FDF_AM824 |
- (s->sfc << CIP_FDF_SHIFT) | syt);
- buffer += 2;
-
- pcm = ACCESS_ONCE(s->pcm);
- if (pcm)
- s->transfer_samples(s, pcm, buffer, data_blocks);
- else
- write_pcm_silence(s, buffer, data_blocks);
- if (s->midi_ports)
- write_midi_messages(s, buffer, data_blocks);
+ buffer[1] = cpu_to_be32(CIP_EOH |
+ ((s->fmt << CIP_FMT_SHIFT) & CIP_FMT_MASK) |
+ ((s->fdf << CIP_FDF_SHIFT) & CIP_FDF_MASK) |
+ (syt & CIP_SYT_MASK));
s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff;
@@ -680,8 +429,9 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
if (queue_out_packet(s, payload_length, false) < 0)
return -EIO;
- if (pcm)
- update_pcm_pointers(s, pcm, data_blocks);
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm && pcm_frames > 0)
+ update_pcm_pointers(s, pcm, pcm_frames);
/* No need to return the number of handled data blocks. */
return 0;
@@ -689,11 +439,13 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
static int handle_in_packet(struct amdtp_stream *s,
unsigned int payload_quadlets, __be32 *buffer,
- unsigned int *data_blocks)
+ unsigned int *data_blocks, unsigned int syt)
{
u32 cip_header[2];
+ unsigned int fmt, fdf;
unsigned int data_block_quadlets, data_block_counter, dbc_interval;
- struct snd_pcm_substream *pcm = NULL;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
bool lost;
cip_header[0] = be32_to_cpu(buffer[0]);
@@ -704,19 +456,30 @@ static int handle_in_packet(struct amdtp_stream *s,
* For convenience, also check FMT field is AM824 or not.
*/
if (((cip_header[0] & CIP_EOH_MASK) == CIP_EOH) ||
- ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH) ||
- ((cip_header[1] & CIP_FMT_MASK) != CIP_FMT_AM)) {
+ ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH)) {
dev_info_ratelimited(&s->unit->device,
"Invalid CIP header for AMDTP: %08X:%08X\n",
cip_header[0], cip_header[1]);
*data_blocks = 0;
+ pcm_frames = 0;
+ goto end;
+ }
+
+ /* Check valid protocol or not. */
+ fmt = (cip_header[1] & CIP_FMT_MASK) >> CIP_FMT_SHIFT;
+ if (fmt != s->fmt) {
+ dev_info_ratelimited(&s->unit->device,
+ "Detect unexpected protocol: %08x %08x\n",
+ cip_header[0], cip_header[1]);
+ *data_blocks = 0;
+ pcm_frames = 0;
goto end;
}
/* Calculate data blocks */
+ fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT;
if (payload_quadlets < 3 ||
- ((cip_header[1] & CIP_FDF_MASK) ==
- (AMDTP_FDF_NO_DATA << CIP_FDF_SHIFT))) {
+ (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) {
*data_blocks = 0;
} else {
data_block_quadlets =
@@ -763,16 +526,7 @@ static int handle_in_packet(struct amdtp_stream *s,
return -EIO;
}
- if (*data_blocks > 0) {
- buffer += 2;
-
- pcm = ACCESS_ONCE(s->pcm);
- if (pcm)
- s->transfer_samples(s, pcm, buffer, *data_blocks);
-
- if (s->midi_ports)
- read_midi_messages(s, buffer, *data_blocks);
- }
+ pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt);
if (s->flags & CIP_DBC_IS_END_EVENT)
s->data_block_counter = data_block_counter;
@@ -783,8 +537,9 @@ end:
if (queue_in_packet(s) < 0)
return -EIO;
- if (pcm)
- update_pcm_pointers(s, pcm, *data_blocks);
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm && pcm_frames > 0)
+ update_pcm_pointers(s, pcm, pcm_frames);
return 0;
}
@@ -854,15 +609,15 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle,
break;
}
+ syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
if (handle_in_packet(s, payload_quadlets, buffer,
- &data_blocks) < 0) {
+ &data_blocks, syt) < 0) {
s->packet_index = -1;
break;
}
/* Process sync slave stream */
if (s->sync_slave && s->sync_slave->callbacked) {
- syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
if (handle_out_packet(s->sync_slave,
data_blocks, syt) < 0) {
s->packet_index = -1;
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp-stream.h
index b2cf9e75693b..8775704a3665 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp-stream.h
@@ -4,6 +4,7 @@
#include <linux/err.h>
#include <linux/interrupt.h>
#include <linux/mutex.h>
+#include <linux/sched.h>
#include <sound/asound.h>
#include "packets-buffer.h"
@@ -80,100 +81,78 @@ enum cip_sfc {
CIP_SFC_COUNT
};
-#define AMDTP_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32
-
-#define AMDTP_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \
- SNDRV_PCM_FMTBIT_S32)
-
-
-/*
- * This module supports maximum 64 PCM channels for one PCM stream
- * This is for our convenience.
- */
-#define AMDTP_MAX_CHANNELS_FOR_PCM 64
-
-/*
- * AMDTP packet can include channels for MIDI conformant data.
- * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
- * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
- *
- * This module supports maximum 1 MIDI conformant data channels.
- * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
- */
-#define AMDTP_MAX_CHANNELS_FOR_MIDI 1
-
struct fw_unit;
struct fw_iso_context;
struct snd_pcm_substream;
struct snd_pcm_runtime;
-struct snd_rawmidi_substream;
enum amdtp_stream_direction {
AMDTP_OUT_STREAM = 0,
AMDTP_IN_STREAM
};
+struct amdtp_stream;
+typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
+ struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt);
struct amdtp_stream {
struct fw_unit *unit;
enum cip_flags flags;
enum amdtp_stream_direction direction;
- struct fw_iso_context *context;
struct mutex mutex;
- enum cip_sfc sfc;
- unsigned int data_block_quadlets;
- unsigned int pcm_channels;
- unsigned int midi_ports;
- void (*transfer_samples)(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
- u8 pcm_positions[AMDTP_MAX_CHANNELS_FOR_PCM];
- u8 midi_position;
-
- unsigned int syt_interval;
- unsigned int transfer_delay;
- unsigned int source_node_id_field;
+ /* For packet processing. */
+ struct fw_iso_context *context;
struct iso_packets_buffer buffer;
-
- struct snd_pcm_substream *pcm;
- struct tasklet_struct period_tasklet;
-
int packet_index;
+
+ /* For CIP headers. */
+ unsigned int source_node_id_field;
+ unsigned int data_block_quadlets;
unsigned int data_block_counter;
+ unsigned int fmt;
+ unsigned int fdf;
+ /* quirk: fixed interval of dbc between previos/current packets. */
+ unsigned int tx_dbc_interval;
+ /* quirk: indicate the value of dbc field in a first packet. */
+ unsigned int tx_first_dbc;
+ /* Internal flags. */
+ enum cip_sfc sfc;
+ unsigned int syt_interval;
+ unsigned int transfer_delay;
unsigned int data_block_state;
-
unsigned int last_syt_offset;
unsigned int syt_offset_state;
+ /* For a PCM substream processing. */
+ struct snd_pcm_substream *pcm;
+ struct tasklet_struct period_tasklet;
unsigned int pcm_buffer_pointer;
unsigned int pcm_period_pointer;
bool pointer_flush;
- bool double_pcm_frames;
-
- struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
- int midi_fifo_limit;
- int midi_fifo_used[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
-
- /* quirk: fixed interval of dbc between previos/current packets. */
- unsigned int tx_dbc_interval;
- /* quirk: indicate the value of dbc field in a first packet. */
- unsigned int tx_first_dbc;
+ /* To wait for first packet. */
bool callbacked;
wait_queue_head_t callback_wait;
struct amdtp_stream *sync_slave;
+
+ /* For backends to process data blocks. */
+ void *protocol;
+ amdtp_stream_process_data_blocks_t process_data_blocks;
};
int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
- enum amdtp_stream_direction dir,
- enum cip_flags flags);
+ enum amdtp_stream_direction dir, enum cip_flags flags,
+ unsigned int fmt,
+ amdtp_stream_process_data_blocks_t process_data_blocks,
+ unsigned int protocol_size);
void amdtp_stream_destroy(struct amdtp_stream *s);
-void amdtp_stream_set_parameters(struct amdtp_stream *s,
- unsigned int rate,
- unsigned int pcm_channels,
- unsigned int midi_ports);
+int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int data_block_quadlets);
unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s);
int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed);
@@ -182,8 +161,7 @@ void amdtp_stream_stop(struct amdtp_stream *s);
int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
struct snd_pcm_runtime *runtime);
-void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
- snd_pcm_format_t format);
+
void amdtp_stream_pcm_prepare(struct amdtp_stream *s);
unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s);
void amdtp_stream_pcm_abort(struct amdtp_stream *s);
@@ -240,24 +218,6 @@ static inline void amdtp_stream_pcm_trigger(struct amdtp_stream *s,
ACCESS_ONCE(s->pcm) = pcm;
}
-/**
- * amdtp_stream_midi_trigger - start/stop playback/capture with a MIDI device
- * @s: the AMDTP stream
- * @port: index of MIDI port
- * @midi: the MIDI device to be started, or %NULL to stop the current device
- *
- * Call this function on a running isochronous stream to enable the actual
- * transmission of MIDI data. This function should be called from the MIDI
- * device's .trigger callback.
- */
-static inline void amdtp_stream_midi_trigger(struct amdtp_stream *s,
- unsigned int port,
- struct snd_rawmidi_substream *midi)
-{
- if (port < s->midi_ports)
- ACCESS_ONCE(s->midi[port]) = midi;
-}
-
static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
{
return sfc & 1;
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 27a04ac8ffee..091290d1f3ea 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -41,7 +41,8 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS);
#define VEN_EDIROL 0x000040ab
#define VEN_PRESONUS 0x00000a92
#define VEN_BRIDGECO 0x000007f5
-#define VEN_MACKIE 0x0000000f
+#define VEN_MACKIE1 0x0000000f
+#define VEN_MACKIE2 0x00000ff2
#define VEN_STANTON 0x00001260
#define VEN_TASCAM 0x0000022e
#define VEN_BEHRINGER 0x00001564
@@ -334,7 +335,7 @@ static void bebob_remove(struct fw_unit *unit)
snd_card_free_when_closed(bebob->card);
}
-static struct snd_bebob_rate_spec normal_rate_spec = {
+static const struct snd_bebob_rate_spec normal_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate
};
@@ -360,9 +361,9 @@ static const struct ieee1394_device_id bebob_id_table[] = {
/* BridgeCo, Audio5 */
SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal),
/* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */
- SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010065, &spec_normal),
+ SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal),
/* Mackie, d.2 (Firewire Option) */
- SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010067, &spec_normal),
+ SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal),
/* Stanton, ScratchAmp */
SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal),
/* Tascam, IF-FW DM */
diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h
index d23caca7f369..4d8fcc78e747 100644
--- a/sound/firewire/bebob/bebob.h
+++ b/sound/firewire/bebob/bebob.h
@@ -31,7 +31,7 @@
#include "../fcp.h"
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
/* basic register addresses on DM1000/DM1100/DM1500 */
@@ -70,9 +70,9 @@ struct snd_bebob_meter_spec {
int (*get)(struct snd_bebob *bebob, u32 *target, unsigned int size);
};
struct snd_bebob_spec {
- struct snd_bebob_clock_spec *clock;
- struct snd_bebob_rate_spec *rate;
- struct snd_bebob_meter_spec *meter;
+ const struct snd_bebob_clock_spec *clock;
+ const struct snd_bebob_rate_spec *rate;
+ const struct snd_bebob_meter_spec *meter;
};
struct snd_bebob {
@@ -235,19 +235,19 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob);
int snd_bebob_create_hwdep_device(struct snd_bebob *bebob);
/* model specific operations */
-extern struct snd_bebob_spec phase88_rack_spec;
-extern struct snd_bebob_spec phase24_series_spec;
-extern struct snd_bebob_spec yamaha_go_spec;
-extern struct snd_bebob_spec saffirepro_26_spec;
-extern struct snd_bebob_spec saffirepro_10_spec;
-extern struct snd_bebob_spec saffire_le_spec;
-extern struct snd_bebob_spec saffire_spec;
-extern struct snd_bebob_spec maudio_fw410_spec;
-extern struct snd_bebob_spec maudio_audiophile_spec;
-extern struct snd_bebob_spec maudio_solo_spec;
-extern struct snd_bebob_spec maudio_ozonic_spec;
-extern struct snd_bebob_spec maudio_nrv10_spec;
-extern struct snd_bebob_spec maudio_special_spec;
+extern const struct snd_bebob_spec phase88_rack_spec;
+extern const struct snd_bebob_spec phase24_series_spec;
+extern const struct snd_bebob_spec yamaha_go_spec;
+extern const struct snd_bebob_spec saffirepro_26_spec;
+extern const struct snd_bebob_spec saffirepro_10_spec;
+extern const struct snd_bebob_spec saffire_le_spec;
+extern const struct snd_bebob_spec saffire_spec;
+extern const struct snd_bebob_spec maudio_fw410_spec;
+extern const struct snd_bebob_spec maudio_audiophile_spec;
+extern const struct snd_bebob_spec maudio_solo_spec;
+extern const struct snd_bebob_spec maudio_ozonic_spec;
+extern const struct snd_bebob_spec maudio_nrv10_spec;
+extern const struct snd_bebob_spec maudio_special_spec;
int snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814);
int snd_bebob_maudio_load_firmware(struct fw_unit *unit);
diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c
index a1a39494ea6c..f11090057949 100644
--- a/sound/firewire/bebob/bebob_focusrite.c
+++ b/sound/firewire/bebob/bebob_focusrite.c
@@ -200,7 +200,7 @@ end:
return err;
}
-struct snd_bebob_spec saffire_le_spec;
+const struct snd_bebob_spec saffire_le_spec;
static enum snd_bebob_clock_type saffire_both_clk_src_types[] = {
SND_BEBOB_CLOCK_TYPE_INTERNAL,
SND_BEBOB_CLOCK_TYPE_EXTERNAL,
@@ -229,7 +229,7 @@ static const char *const saffire_meter_labels[] = {
static int
saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
{
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
unsigned int channels;
u64 offset;
int err;
@@ -260,60 +260,60 @@ saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
return err;
}
-static struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
+static const struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
.get = &saffirepro_both_clk_freq_get,
.set = &saffirepro_both_clk_freq_set,
};
/* Saffire Pro 26 I/O */
-static struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
+static const struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
.num = ARRAY_SIZE(saffirepro_26_clk_src_types),
.types = saffirepro_26_clk_src_types,
.get = &saffirepro_both_clk_src_get,
};
-struct snd_bebob_spec saffirepro_26_spec = {
+const struct snd_bebob_spec saffirepro_26_spec = {
.clock = &saffirepro_26_clk_spec,
.rate = &saffirepro_both_rate_spec,
.meter = NULL
};
/* Saffire Pro 10 I/O */
-static struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
+static const struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
.num = ARRAY_SIZE(saffirepro_10_clk_src_types),
.types = saffirepro_10_clk_src_types,
.get = &saffirepro_both_clk_src_get,
};
-struct snd_bebob_spec saffirepro_10_spec = {
+const struct snd_bebob_spec saffirepro_10_spec = {
.clock = &saffirepro_10_clk_spec,
.rate = &saffirepro_both_rate_spec,
.meter = NULL
};
-static struct snd_bebob_rate_spec saffire_both_rate_spec = {
+static const struct snd_bebob_rate_spec saffire_both_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-static struct snd_bebob_clock_spec saffire_both_clk_spec = {
+static const struct snd_bebob_clock_spec saffire_both_clk_spec = {
.num = ARRAY_SIZE(saffire_both_clk_src_types),
.types = saffire_both_clk_src_types,
.get = &saffire_both_clk_src_get,
};
/* Saffire LE */
-static struct snd_bebob_meter_spec saffire_le_meter_spec = {
+static const struct snd_bebob_meter_spec saffire_le_meter_spec = {
.num = ARRAY_SIZE(saffire_le_meter_labels),
.labels = saffire_le_meter_labels,
.get = &saffire_meter_get,
};
-struct snd_bebob_spec saffire_le_spec = {
+const struct snd_bebob_spec saffire_le_spec = {
.clock = &saffire_both_clk_spec,
.rate = &saffire_both_rate_spec,
.meter = &saffire_le_meter_spec
};
/* Saffire */
-static struct snd_bebob_meter_spec saffire_meter_spec = {
+static const struct snd_bebob_meter_spec saffire_meter_spec = {
.num = ARRAY_SIZE(saffire_meter_labels),
.labels = saffire_meter_labels,
.get = &saffire_meter_get,
};
-struct snd_bebob_spec saffire_spec = {
+const struct snd_bebob_spec saffire_spec = {
.clock = &saffire_both_clk_spec,
.rate = &saffire_both_rate_spec,
.meter = &saffire_meter_spec
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index 057495d54ab0..7b86a6b99f07 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -687,7 +687,7 @@ static const char *const nrv10_meter_labels[] = {
static int
normal_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
{
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
unsigned int c, channels;
int err;
@@ -712,85 +712,85 @@ end:
}
/* for special customized devices */
-static struct snd_bebob_rate_spec special_rate_spec = {
+static const struct snd_bebob_rate_spec special_rate_spec = {
.get = &special_get_rate,
.set = &special_set_rate,
};
-static struct snd_bebob_clock_spec special_clk_spec = {
+static const struct snd_bebob_clock_spec special_clk_spec = {
.num = ARRAY_SIZE(special_clk_types),
.types = special_clk_types,
.get = &special_clk_get,
};
-static struct snd_bebob_meter_spec special_meter_spec = {
+static const struct snd_bebob_meter_spec special_meter_spec = {
.num = ARRAY_SIZE(special_meter_labels),
.labels = special_meter_labels,
.get = &special_meter_get
};
-struct snd_bebob_spec maudio_special_spec = {
+const struct snd_bebob_spec maudio_special_spec = {
.clock = &special_clk_spec,
.rate = &special_rate_spec,
.meter = &special_meter_spec
};
/* Firewire 410 specification */
-static struct snd_bebob_rate_spec usual_rate_spec = {
+static const struct snd_bebob_rate_spec usual_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-static struct snd_bebob_meter_spec fw410_meter_spec = {
+static const struct snd_bebob_meter_spec fw410_meter_spec = {
.num = ARRAY_SIZE(fw410_meter_labels),
.labels = fw410_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_fw410_spec = {
+const struct snd_bebob_spec maudio_fw410_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &fw410_meter_spec
};
/* Firewire Audiophile specification */
-static struct snd_bebob_meter_spec audiophile_meter_spec = {
+static const struct snd_bebob_meter_spec audiophile_meter_spec = {
.num = ARRAY_SIZE(audiophile_meter_labels),
.labels = audiophile_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_audiophile_spec = {
+const struct snd_bebob_spec maudio_audiophile_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &audiophile_meter_spec
};
/* Firewire Solo specification */
-static struct snd_bebob_meter_spec solo_meter_spec = {
+static const struct snd_bebob_meter_spec solo_meter_spec = {
.num = ARRAY_SIZE(solo_meter_labels),
.labels = solo_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_solo_spec = {
+const struct snd_bebob_spec maudio_solo_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &solo_meter_spec
};
/* Ozonic specification */
-static struct snd_bebob_meter_spec ozonic_meter_spec = {
+static const struct snd_bebob_meter_spec ozonic_meter_spec = {
.num = ARRAY_SIZE(ozonic_meter_labels),
.labels = ozonic_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_ozonic_spec = {
+const struct snd_bebob_spec maudio_ozonic_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &ozonic_meter_spec
};
/* NRV10 specification */
-static struct snd_bebob_meter_spec nrv10_meter_spec = {
+static const struct snd_bebob_meter_spec nrv10_meter_spec = {
.num = ARRAY_SIZE(nrv10_meter_labels),
.labels = nrv10_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_nrv10_spec = {
+const struct snd_bebob_spec maudio_nrv10_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &nrv10_meter_spec
diff --git a/sound/firewire/bebob/bebob_midi.c b/sound/firewire/bebob/bebob_midi.c
index 5681143925cd..90d95be499b0 100644
--- a/sound/firewire/bebob/bebob_midi.c
+++ b/sound/firewire/bebob/bebob_midi.c
@@ -72,11 +72,11 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&bebob->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&bebob->tx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&bebob->tx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&bebob->tx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&bebob->tx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&bebob->lock, flags);
}
@@ -89,11 +89,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&bebob->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&bebob->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&bebob->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&bebob->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&bebob->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&bebob->lock, flags);
}
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index c0f018a61fdc..ef224d6f5c24 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -122,11 +122,11 @@ pcm_init_hw_params(struct snd_bebob *bebob,
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
s = &bebob->tx_stream;
formations = bebob->tx_stream_formations;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
s = &bebob->rx_stream;
formations = bebob->rx_stream_formations;
}
@@ -146,7 +146,7 @@ pcm_init_hw_params(struct snd_bebob *bebob,
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(s, runtime);
end:
return err;
}
@@ -155,7 +155,7 @@ static int
pcm_open(struct snd_pcm_substream *substream)
{
struct snd_bebob *bebob = substream->private_data;
- struct snd_bebob_rate_spec *spec = bebob->spec->rate;
+ const struct snd_bebob_rate_spec *spec = bebob->spec->rate;
unsigned int sampling_rate;
enum snd_bebob_clock_type src;
int err;
@@ -220,8 +220,8 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream,
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&bebob->substreams_counter);
- amdtp_stream_set_pcm_format(&bebob->tx_stream,
- params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&bebob->tx_stream, params_format(hw_params));
return 0;
}
@@ -239,8 +239,8 @@ pcm_playback_hw_params(struct snd_pcm_substream *substream,
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&bebob->substreams_counter);
- amdtp_stream_set_pcm_format(&bebob->rx_stream,
- params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&bebob->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c
index 301cc6a93945..ec24f96794f5 100644
--- a/sound/firewire/bebob/bebob_proc.c
+++ b/sound/firewire/bebob/bebob_proc.c
@@ -73,7 +73,7 @@ proc_read_meters(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_bebob *bebob = entry->private_data;
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
u32 *buf;
unsigned int i, c, channels, size;
@@ -138,8 +138,8 @@ proc_read_clock(struct snd_info_entry *entry,
"SYT-Match",
};
struct snd_bebob *bebob = entry->private_data;
- struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
enum snd_bebob_clock_type src;
unsigned int rate;
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 5be5242e1ed8..926e5dcbb66a 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -119,7 +119,7 @@ end:
int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob,
enum snd_bebob_clock_type *src)
{
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7];
unsigned int id;
enum avc_bridgeco_plug_type type;
@@ -338,7 +338,7 @@ map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
err = -ENOSYS;
goto end;
}
- s->midi_position = stm_pos;
+ amdtp_am824_set_midi_position(s, stm_pos);
midi = stm_pos;
break;
/* for PCM data channel */
@@ -354,11 +354,12 @@ map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
case 0x09: /* Digital */
default:
location = pcm + sec_loc;
- if (location >= AMDTP_MAX_CHANNELS_FOR_PCM) {
+ if (location >= AM824_MAX_CHANNELS_FOR_PCM) {
err = -ENOSYS;
goto end;
}
- s->pcm_positions[location] = stm_pos;
+ amdtp_am824_set_pcm_position(s, location,
+ stm_pos);
break;
}
}
@@ -427,12 +428,19 @@ make_both_connections(struct snd_bebob *bebob, unsigned int rate)
index = get_formation_index(rate);
pcm_channels = bebob->tx_stream_formations[index].pcm;
midi_channels = bebob->tx_stream_formations[index].midi;
- amdtp_stream_set_parameters(&bebob->tx_stream,
- rate, pcm_channels, midi_channels * 8);
+ err = amdtp_am824_set_parameters(&bebob->tx_stream, rate,
+ pcm_channels, midi_channels * 8,
+ false);
+ if (err < 0)
+ goto end;
+
pcm_channels = bebob->rx_stream_formations[index].pcm;
midi_channels = bebob->rx_stream_formations[index].midi;
- amdtp_stream_set_parameters(&bebob->rx_stream,
- rate, pcm_channels, midi_channels * 8);
+ err = amdtp_am824_set_parameters(&bebob->rx_stream, rate,
+ pcm_channels, midi_channels * 8,
+ false);
+ if (err < 0)
+ goto end;
/* establish connections for both streams */
err = cmp_connection_establish(&bebob->out_conn,
@@ -530,8 +538,8 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
if (err < 0)
goto end;
- err = amdtp_stream_init(&bebob->tx_stream, bebob->unit,
- AMDTP_IN_STREAM, CIP_BLOCKING);
+ err = amdtp_am824_init(&bebob->tx_stream, bebob->unit,
+ AMDTP_IN_STREAM, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(&bebob->tx_stream);
destroy_both_connections(bebob);
@@ -559,8 +567,8 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
if (bebob->maudio_special_quirk)
bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC;
- err = amdtp_stream_init(&bebob->rx_stream, bebob->unit,
- AMDTP_OUT_STREAM, CIP_BLOCKING);
+ err = amdtp_am824_init(&bebob->rx_stream, bebob->unit,
+ AMDTP_OUT_STREAM, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(&bebob->tx_stream);
amdtp_stream_destroy(&bebob->rx_stream);
@@ -572,7 +580,7 @@ end:
int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
{
- struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
+ const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
struct amdtp_stream *master, *slave;
enum cip_flags sync_mode;
unsigned int curr_rate;
@@ -864,8 +872,8 @@ parse_stream_formation(u8 *buf, unsigned int len,
}
}
- if (formation[i].pcm > AMDTP_MAX_CHANNELS_FOR_PCM ||
- formation[i].midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+ if (formation[i].pcm > AM824_MAX_CHANNELS_FOR_PCM ||
+ formation[i].midi > AM824_MAX_CHANNELS_FOR_MIDI)
return -ENOSYS;
return 0;
@@ -959,7 +967,7 @@ end:
int snd_bebob_stream_discover(struct snd_bebob *bebob)
{
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES];
enum avc_bridgeco_plug_type type;
unsigned int i;
diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c
index 9242e33d2cf1..c38358b82ada 100644
--- a/sound/firewire/bebob/bebob_terratec.c
+++ b/sound/firewire/bebob/bebob_terratec.c
@@ -55,30 +55,30 @@ phase24_series_clk_src_get(struct snd_bebob *bebob, unsigned int *id)
return 0;
}
-static struct snd_bebob_rate_spec phase_series_rate_spec = {
+static const struct snd_bebob_rate_spec phase_series_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
/* PHASE 88 Rack FW */
-static struct snd_bebob_clock_spec phase88_rack_clk = {
+static const struct snd_bebob_clock_spec phase88_rack_clk = {
.num = ARRAY_SIZE(phase88_rack_clk_src_types),
.types = phase88_rack_clk_src_types,
.get = &phase88_rack_clk_src_get,
};
-struct snd_bebob_spec phase88_rack_spec = {
+const struct snd_bebob_spec phase88_rack_spec = {
.clock = &phase88_rack_clk,
.rate = &phase_series_rate_spec,
.meter = NULL
};
/* 'PHASE 24 FW' and 'PHASE X24 FW' */
-static struct snd_bebob_clock_spec phase24_series_clk = {
+static const struct snd_bebob_clock_spec phase24_series_clk = {
.num = ARRAY_SIZE(phase24_series_clk_src_types),
.types = phase24_series_clk_src_types,
.get = &phase24_series_clk_src_get,
};
-struct snd_bebob_spec phase24_series_spec = {
+const struct snd_bebob_spec phase24_series_spec = {
.clock = &phase24_series_clk,
.rate = &phase_series_rate_spec,
.meter = NULL
diff --git a/sound/firewire/bebob/bebob_yamaha.c b/sound/firewire/bebob/bebob_yamaha.c
index 58101702410b..90d4404f77ce 100644
--- a/sound/firewire/bebob/bebob_yamaha.c
+++ b/sound/firewire/bebob/bebob_yamaha.c
@@ -46,16 +46,16 @@ clk_src_get(struct snd_bebob *bebob, unsigned int *id)
return 0;
}
-static struct snd_bebob_clock_spec clock_spec = {
+static const struct snd_bebob_clock_spec clock_spec = {
.num = ARRAY_SIZE(clk_src_types),
.types = clk_src_types,
.get = &clk_src_get,
};
-static struct snd_bebob_rate_spec rate_spec = {
+static const struct snd_bebob_rate_spec rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-struct snd_bebob_spec yamaha_go_spec = {
+const struct snd_bebob_spec yamaha_go_spec = {
.clock = &clock_spec,
.rate = &rate_spec,
.meter = NULL
diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c
index fe43ce791f84..151b09f240f2 100644
--- a/sound/firewire/dice/dice-midi.c
+++ b/sound/firewire/dice/dice-midi.c
@@ -52,10 +52,10 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&dice->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&dice->tx_stream,
+ amdtp_am824_midi_trigger(&dice->tx_stream,
substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&dice->tx_stream,
+ amdtp_am824_midi_trigger(&dice->tx_stream,
substrm->number, NULL);
spin_unlock_irqrestore(&dice->lock, flags);
@@ -69,11 +69,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&dice->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&dice->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&dice->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&dice->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&dice->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&dice->lock, flags);
}
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 4e67b1da0fe6..9b3431999fc8 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -133,11 +133,11 @@ static int init_hw_info(struct snd_dice *dice,
SNDRV_PCM_INFO_BLOCK_TRANSFER;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- hw->formats = AMDTP_IN_PCM_FORMAT_BITS;
+ hw->formats = AM824_IN_PCM_FORMAT_BITS;
stream = &dice->tx_stream;
pcm_channels = dice->tx_channels;
} else {
- hw->formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ hw->formats = AM824_OUT_PCM_FORMAT_BITS;
stream = &dice->rx_stream;
pcm_channels = dice->rx_channels;
}
@@ -156,7 +156,7 @@ static int init_hw_info(struct snd_dice *dice,
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(stream, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(stream, runtime);
end:
return err;
}
@@ -243,8 +243,7 @@ static int capture_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&dice->mutex);
}
- amdtp_stream_set_pcm_format(&dice->tx_stream,
- params_format(hw_params));
+ amdtp_am824_set_pcm_format(&dice->tx_stream, params_format(hw_params));
return 0;
}
@@ -265,8 +264,7 @@ static int playback_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&dice->mutex);
}
- amdtp_stream_set_pcm_format(&dice->rx_stream,
- params_format(hw_params));
+ amdtp_am824_set_pcm_format(&dice->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 07dbd01d7a6b..2108f7f1a764 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -100,6 +100,7 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream,
{
struct fw_iso_resources *resources;
unsigned int i, mode, pcm_chs, midi_ports;
+ bool double_pcm_frames;
int err;
err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
@@ -125,21 +126,24 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream,
* For this quirk, blocking mode is required and PCM buffer size should
* be aligned to SYT_INTERVAL.
*/
- if (mode > 1) {
+ double_pcm_frames = mode > 1;
+ if (double_pcm_frames) {
rate /= 2;
pcm_chs *= 2;
- stream->double_pcm_frames = true;
- } else {
- stream->double_pcm_frames = false;
}
- amdtp_stream_set_parameters(stream, rate, pcm_chs, midi_ports);
- if (mode > 1) {
+ err = amdtp_am824_set_parameters(stream, rate, pcm_chs, midi_ports,
+ double_pcm_frames);
+ if (err < 0)
+ goto end;
+
+ if (double_pcm_frames) {
pcm_chs /= 2;
for (i = 0; i < pcm_chs; i++) {
- stream->pcm_positions[i] = i * 2;
- stream->pcm_positions[i + pcm_chs] = i * 2 + 1;
+ amdtp_am824_set_pcm_position(stream, i, i * 2);
+ amdtp_am824_set_pcm_position(stream, i + pcm_chs,
+ i * 2 + 1);
}
}
@@ -302,7 +306,7 @@ static int init_stream(struct snd_dice *dice, struct amdtp_stream *stream)
goto end;
resources->channels_mask = 0x00000000ffffffffuLL;
- err = amdtp_stream_init(stream, dice->unit, dir, CIP_BLOCKING);
+ err = amdtp_am824_init(stream, dice->unit, dir, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
fw_iso_resources_destroy(resources);
diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h
index ecf5dc862235..101550ac1a24 100644
--- a/sound/firewire/dice/dice.h
+++ b/sound/firewire/dice/dice.h
@@ -34,7 +34,7 @@
#include <sound/pcm_params.h>
#include <sound/rawmidi.h>
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../iso-resources.h"
#include "../lib.h"
#include "dice-interface.h"
diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile
new file mode 100644
index 000000000000..1123e68c8b28
--- /dev/null
+++ b/sound/firewire/digi00x/Makefile
@@ -0,0 +1,4 @@
+snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \
+ digi00x-pcm.o digi00x-hwdep.o \
+ digi00x-transaction.o digi00x-midi.o digi00x.o
+obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
new file mode 100644
index 000000000000..b02a5e8cad44
--- /dev/null
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -0,0 +1,442 @@
+/*
+ * amdtp-dot.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ * Copyright (C) 2012 Robin Gareus <robin@gareus.org>
+ * Copyright (C) 2012 Damien Zammit <damien@zamaudio.com>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "digi00x.h"
+
+#define CIP_FMT_AM 0x10
+
+/* 'Clock-based rate control mode' is just supported. */
+#define AMDTP_FDF_AM824 0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND 3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS 8
+
+/*
+ * The double-oh-three algorithm was discovered by Robin Gareus and Damien
+ * Zammit in 2012, with reverse-engineering for Digi 003 Rack.
+ */
+struct dot_state {
+ u8 carry;
+ u8 idx;
+ unsigned int off;
+};
+
+struct amdtp_dot {
+ unsigned int pcm_channels;
+ struct dot_state state;
+
+ unsigned int midi_ports;
+ /* 2 = MAX(DOT_MIDI_IN_PORTS, DOT_MIDI_OUT_PORTS) */
+ struct snd_rawmidi_substream *midi[2];
+ int midi_fifo_used[2];
+ int midi_fifo_limit;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+};
+
+/*
+ * double-oh-three look up table
+ *
+ * @param idx index byte (audio-sample data) 0x00..0xff
+ * @param off channel offset shift
+ * @return salt to XOR with given data
+ */
+#define BYTE_PER_SAMPLE (4)
+#define MAGIC_DOT_BYTE (2)
+#define MAGIC_BYTE_OFF(x) (((x) * BYTE_PER_SAMPLE) + MAGIC_DOT_BYTE)
+static const u8 dot_scrt(const u8 idx, const unsigned int off)
+{
+ /*
+ * the length of the added pattern only depends on the lower nibble
+ * of the last non-zero data
+ */
+ static const u8 len[16] = {0, 1, 3, 5, 7, 9, 11, 13, 14,
+ 12, 10, 8, 6, 4, 2, 0};
+
+ /*
+ * the lower nibble of the salt. Interleaved sequence.
+ * this is walked backwards according to len[]
+ */
+ static const u8 nib[15] = {0x8, 0x7, 0x9, 0x6, 0xa, 0x5, 0xb, 0x4,
+ 0xc, 0x3, 0xd, 0x2, 0xe, 0x1, 0xf};
+
+ /* circular list for the salt's hi nibble. */
+ static const u8 hir[15] = {0x0, 0x6, 0xf, 0x8, 0x7, 0x5, 0x3, 0x4,
+ 0xc, 0xd, 0xe, 0x1, 0x2, 0xb, 0xa};
+
+ /*
+ * start offset for upper nibble mapping.
+ * note: 9 is /special/. In the case where the high nibble == 0x9,
+ * hir[] is not used and - coincidentally - the salt's hi nibble is
+ * 0x09 regardless of the offset.
+ */
+ static const u8 hio[16] = {0, 11, 12, 6, 7, 5, 1, 4,
+ 3, 0x00, 14, 13, 8, 9, 10, 2};
+
+ const u8 ln = idx & 0xf;
+ const u8 hn = (idx >> 4) & 0xf;
+ const u8 hr = (hn == 0x9) ? 0x9 : hir[(hio[hn] + off) % 15];
+
+ if (len[ln] < off)
+ return 0x00;
+
+ return ((nib[14 + off - len[ln]]) | (hr << 4));
+}
+
+static void dot_encode_step(struct dot_state *state, __be32 *const buffer)
+{
+ u8 * const data = (u8 *) buffer;
+
+ if (data[MAGIC_DOT_BYTE] != 0x00) {
+ state->off = 0;
+ state->idx = data[MAGIC_DOT_BYTE] ^ state->carry;
+ }
+ data[MAGIC_DOT_BYTE] ^= state->carry;
+ state->carry = dot_scrt(state->idx, ++(state->off));
+}
+
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels)
+{
+ struct amdtp_dot *p = s->protocol;
+ int err;
+
+ if (amdtp_stream_running(s))
+ return -EBUSY;
+
+ /*
+ * A first data channel is for MIDI conformant data channel, the rest is
+ * Multi Bit Linear Audio data channel.
+ */
+ err = amdtp_stream_set_parameters(s, rate, pcm_channels + 1);
+ if (err < 0)
+ return err;
+
+ s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+ p->pcm_channels = pcm_channels;
+
+ if (s->direction == AMDTP_IN_STREAM)
+ p->midi_ports = DOT_MIDI_IN_PORTS;
+ else
+ p->midi_ports = DOT_MIDI_OUT_PORTS;
+
+ /*
+ * We do not know the actual MIDI FIFO size of most devices. Just
+ * assume two bytes, i.e., one byte can be received over the bus while
+ * the previous one is transmitted over MIDI.
+ * (The value here is adjusted for midi_ratelimit_per_packet().)
+ */
+ p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+ return 0;
+}
+
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32((*src >> 8) | 0x40000000);
+ dot_encode_step(&p->state, &buffer[c]);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32((*src << 8) | 0x40000000);
+ dot_encode_step(&p->state, &buffer[c]);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[c]) << 8;
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int channels, i, c;
+
+ channels = p->pcm_channels;
+
+ buffer++;
+ for (i = 0; i < data_blocks; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[c] = cpu_to_be32(0x40000000);
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_dot *p = s->protocol;
+ int used;
+
+ used = p->midi_fifo_used[port];
+ if (used == 0)
+ return true;
+
+ used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+ used = max(used, 0);
+ p->midi_fifo_used[port] = used;
+
+ return used < p->midi_fifo_limit;
+}
+
+static inline void midi_use_bytes(struct amdtp_stream *s,
+ unsigned int port, unsigned int count)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ p->midi_fifo_used[port] += amdtp_rate_table[s->sfc] * count;
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int f, port;
+ int len;
+ u8 *b;
+
+ for (f = 0; f < data_blocks; f++) {
+ port = (s->data_block_counter + f) % 8;
+ b = (u8 *)&buffer[0];
+
+ len = 0;
+ if (port < p->midi_ports &&
+ midi_ratelimit_per_packet(s, port) &&
+ p->midi[port] != NULL)
+ len = snd_rawmidi_transmit(p->midi[port], b + 1, 2);
+
+ if (len > 0) {
+ b[3] = (0x10 << port) | len;
+ midi_use_bytes(s, port, len);
+ } else {
+ b[1] = 0;
+ b[2] = 0;
+ b[3] = 0;
+ }
+ b[0] = 0x80;
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int f, port, len;
+ u8 *b;
+
+ for (f = 0; f < data_blocks; f++) {
+ b = (u8 *)&buffer[0];
+ port = b[3] >> 4;
+ len = b[3] & 0x0f;
+
+ if (port < p->midi_ports && p->midi[port] && len > 0)
+ snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ /* This protocol delivers 24 bit data in 32bit data channel. */
+ err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ if (err < 0)
+ return err;
+
+ return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ if (port < p->midi_ports)
+ ACCESS_ONCE(p->midi[port]) = midi;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks;
+ } else {
+ pcm_frames = 0;
+ }
+
+ read_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks;
+ } else {
+ write_pcm_silence(s, buffer, data_blocks);
+ pcm_frames = 0;
+ }
+
+ write_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+ enum cip_flags flags;
+
+ /* Use different mode between incoming/outgoing. */
+ if (dir == AMDTP_IN_STREAM) {
+ flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK;
+ process_data_blocks = process_tx_data_blocks;
+ } else {
+ flags = CIP_BLOCKING;
+ process_data_blocks = process_rx_data_blocks;
+ }
+
+ return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+ process_data_blocks, sizeof(struct amdtp_dot));
+}
+
+void amdtp_dot_reset(struct amdtp_stream *s)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ p->state.carry = 0x00;
+ p->state.idx = 0x00;
+ p->state.off = 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c
new file mode 100644
index 000000000000..f188e4758fd2
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-hwdep.c
@@ -0,0 +1,200 @@
+/*
+ * digi00x-hwdep.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node information
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ * 4.get asynchronous messaging
+ */
+
+#include "digi00x.h"
+
+static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count,
+ loff_t *offset)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+ DEFINE_WAIT(wait);
+ union snd_firewire_event event;
+
+ spin_lock_irq(&dg00x->lock);
+
+ while (!dg00x->dev_lock_changed && dg00x->msg == 0) {
+ prepare_to_wait(&dg00x->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&dg00x->lock);
+ schedule();
+ finish_wait(&dg00x->hwdep_wait, &wait);
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ spin_lock_irq(&dg00x->lock);
+ }
+
+ memset(&event, 0, sizeof(event));
+ if (dg00x->dev_lock_changed) {
+ event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+ event.lock_status.status = (dg00x->dev_lock_count > 0);
+ dg00x->dev_lock_changed = false;
+
+ count = min_t(long, count, sizeof(event.lock_status));
+ } else {
+ event.digi00x_message.type =
+ SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE;
+ event.digi00x_message.message = dg00x->msg;
+ dg00x->msg = 0;
+
+ count = min_t(long, count, sizeof(event.digi00x_message));
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ if (copy_to_user(buf, &event, count))
+ return -EFAULT;
+
+ return count;
+}
+
+static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file,
+ poll_table *wait)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+ unsigned int events;
+
+ poll_wait(file, &dg00x->hwdep_wait, wait);
+
+ spin_lock_irq(&dg00x->lock);
+ if (dg00x->dev_lock_changed || dg00x->msg)
+ events = POLLIN | POLLRDNORM;
+ else
+ events = 0;
+ spin_unlock_irq(&dg00x->lock);
+
+ return events;
+}
+
+static int hwdep_get_info(struct snd_dg00x *dg00x, void __user *arg)
+{
+ struct fw_device *dev = fw_parent_device(dg00x->unit);
+ struct snd_firewire_get_info info;
+
+ memset(&info, 0, sizeof(info));
+ info.type = SNDRV_FIREWIRE_TYPE_DIGI00X;
+ info.card = dev->card->index;
+ *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+ *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+ strlcpy(info.device_name, dev_name(&dev->device),
+ sizeof(info.device_name));
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+static int hwdep_lock(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ if (dg00x->dev_lock_count == 0) {
+ dg00x->dev_lock_count = -1;
+ err = 0;
+ } else {
+ err = -EBUSY;
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ return err;
+}
+
+static int hwdep_unlock(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ if (dg00x->dev_lock_count == -1) {
+ dg00x->dev_lock_count = 0;
+ err = 0;
+ } else {
+ err = -EBADFD;
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ return err;
+}
+
+static int hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+
+ spin_lock_irq(&dg00x->lock);
+ if (dg00x->dev_lock_count == -1)
+ dg00x->dev_lock_count = 0;
+ spin_unlock_irq(&dg00x->lock);
+
+ return 0;
+}
+
+static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+
+ switch (cmd) {
+ case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+ return hwdep_get_info(dg00x, (void __user *)arg);
+ case SNDRV_FIREWIRE_IOCTL_LOCK:
+ return hwdep_lock(dg00x);
+ case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+ return hwdep_unlock(dg00x);
+ default:
+ return -ENOIOCTLCMD;
+ }
+}
+
+#ifdef CONFIG_COMPAT
+static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return hwdep_ioctl(hwdep, file, cmd,
+ (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+ .read = hwdep_read,
+ .release = hwdep_release,
+ .poll = hwdep_poll,
+ .ioctl = hwdep_ioctl,
+ .ioctl_compat = hwdep_compat_ioctl,
+};
+
+int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x)
+{
+ struct snd_hwdep *hwdep;
+ int err;
+
+ err = snd_hwdep_new(dg00x->card, "Digi00x", 0, &hwdep);
+ if (err < 0)
+ return err;
+
+ strcpy(hwdep->name, "Digi00x");
+ hwdep->iface = SNDRV_HWDEP_IFACE_FW_DIGI00X;
+ hwdep->ops = hwdep_ops;
+ hwdep->private_data = dg00x;
+ hwdep->exclusive = true;
+
+ return err;
+}
diff --git a/sound/firewire/digi00x/digi00x-midi.c b/sound/firewire/digi00x/digi00x-midi.c
new file mode 100644
index 000000000000..9aa8b4623108
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-midi.c
@@ -0,0 +1,160 @@
+/*
+ * digi00x-midi.h - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int midi_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ int err;
+
+ /* This port is for asynchronous transaction. */
+ if (substream->number == 0)
+ return 0;
+
+ err = snd_dg00x_stream_lock_try(dg00x);
+ if (err < 0)
+ return err;
+
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ err = snd_dg00x_stream_start_duplex(dg00x, 0);
+ mutex_unlock(&dg00x->mutex);
+ if (err < 0)
+ snd_dg00x_stream_lock_release(dg00x);
+
+ return err;
+}
+
+static int midi_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+
+ /* This port is for asynchronous transaction. */
+ if (substream->number == 0)
+ return 0;
+
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter--;
+ snd_dg00x_stream_stop_duplex(dg00x);
+ mutex_unlock(&dg00x->mutex);
+
+ snd_dg00x_stream_lock_release(dg00x);
+ return 0;
+}
+
+static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+ struct snd_dg00x *dg00x = substrm->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ /* This port is for asynchronous transaction. */
+ if (substrm->number == 0) {
+ if (up)
+ dg00x->in_control = substrm;
+ else
+ dg00x->in_control = NULL;
+ } else {
+ if (up)
+ amdtp_dot_midi_trigger(&dg00x->tx_stream,
+ substrm->number - 1, substrm);
+ else
+ amdtp_dot_midi_trigger(&dg00x->tx_stream,
+ substrm->number - 1, NULL);
+ }
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+ struct snd_dg00x *dg00x = substrm->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ /* This port is for asynchronous transaction. */
+ if (substrm->number == 0) {
+ if (up)
+ snd_fw_async_midi_port_run(&dg00x->out_control,
+ substrm);
+ } else {
+ if (up)
+ amdtp_dot_midi_trigger(&dg00x->rx_stream,
+ substrm->number - 1, substrm);
+ else
+ amdtp_dot_midi_trigger(&dg00x->rx_stream,
+ substrm->number - 1, NULL);
+ }
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_capture_ops = {
+ .open = midi_open,
+ .close = midi_close,
+ .trigger = midi_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_playback_ops = {
+ .open = midi_open,
+ .close = midi_close,
+ .trigger = midi_playback_trigger,
+};
+
+static void set_midi_substream_names(struct snd_dg00x *dg00x,
+ struct snd_rawmidi_str *str)
+{
+ struct snd_rawmidi_substream *subs;
+
+ list_for_each_entry(subs, &str->substreams, list) {
+ if (subs->number > 0)
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ dg00x->card->shortname, subs->number);
+ else
+ /* This port is for asynchronous transaction. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s control",
+ dg00x->card->shortname);
+ }
+}
+
+int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x)
+{
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_str *str;
+ int err;
+
+ err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 0,
+ DOT_MIDI_OUT_PORTS + 1, DOT_MIDI_IN_PORTS + 1, &rmidi);
+ if (err < 0)
+ return err;
+
+ snprintf(rmidi->name, sizeof(rmidi->name),
+ "%s MIDI", dg00x->card->shortname);
+ rmidi->private_data = dg00x;
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &midi_capture_ops);
+ str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+ set_midi_substream_names(dg00x, str);
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &midi_playback_ops);
+ str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+ set_midi_substream_names(dg00x, str);
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ return 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
new file mode 100644
index 000000000000..cac28f70aef7
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -0,0 +1,373 @@
+/*
+ * digi00x-pcm.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ const struct snd_interval *c =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1,
+ };
+ unsigned int i;
+
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (!snd_interval_test(c,
+ snd_dg00x_stream_pcm_channels[i]))
+ continue;
+
+ t.min = min(t.min, snd_dg00x_stream_rates[i]);
+ t.max = max(t.max, snd_dg00x_stream_rates[i]);
+ }
+
+ return snd_interval_refine(r, &t);
+}
+
+static int hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ const struct snd_interval *r =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1,
+ };
+ unsigned int i;
+
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (!snd_interval_test(r, snd_dg00x_stream_rates[i]))
+ continue;
+
+ t.min = min(t.min, snd_dg00x_stream_pcm_channels[i]);
+ t.max = max(t.max, snd_dg00x_stream_pcm_channels[i]);
+ }
+
+ return snd_interval_refine(c, &t);
+}
+
+static int pcm_init_hw_params(struct snd_dg00x *dg00x,
+ struct snd_pcm_substream *substream)
+{
+ static const struct snd_pcm_hardware hardware = {
+ .info = SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_JOINT_DUPLEX |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 18,
+ .period_bytes_min = 4 * 18,
+ .period_bytes_max = 4 * 18 * 2048,
+ .buffer_bytes_max = 4 * 18 * 2048 * 2,
+ .periods_min = 2,
+ .periods_max = UINT_MAX,
+ };
+ struct amdtp_stream *s;
+ int err;
+
+ substream->runtime->hw = hardware;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
+ s = &dg00x->tx_stream;
+ } else {
+ substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16 |
+ SNDRV_PCM_FMTBIT_S32;
+ s = &dg00x->rx_stream;
+ }
+
+ err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_channels, NULL,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_rate, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (err < 0)
+ return err;
+
+ return amdtp_dot_add_pcm_hw_constraints(s, substream->runtime);
+}
+
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ enum snd_dg00x_clock clock;
+ bool detect;
+ unsigned int rate;
+ int err;
+
+ err = snd_dg00x_stream_lock_try(dg00x);
+ if (err < 0)
+ goto end;
+
+ err = pcm_init_hw_params(dg00x, substream);
+ if (err < 0)
+ goto err_locked;
+
+ /* Check current clock source. */
+ err = snd_dg00x_stream_get_clock(dg00x, &clock);
+ if (err < 0)
+ goto err_locked;
+ if (clock != SND_DG00X_CLOCK_INTERNAL) {
+ err = snd_dg00x_stream_check_external_clock(dg00x, &detect);
+ if (err < 0)
+ goto err_locked;
+ if (!detect) {
+ err = -EBUSY;
+ goto err_locked;
+ }
+ }
+
+ if ((clock != SND_DG00X_CLOCK_INTERNAL) ||
+ amdtp_stream_pcm_running(&dg00x->rx_stream) ||
+ amdtp_stream_pcm_running(&dg00x->tx_stream)) {
+ err = snd_dg00x_stream_get_external_rate(dg00x, &rate);
+ if (err < 0)
+ goto err_locked;
+ substream->runtime->hw.rate_min = rate;
+ substream->runtime->hw.rate_max = rate;
+ }
+
+ snd_pcm_set_sync(substream);
+end:
+ return err;
+err_locked:
+ snd_dg00x_stream_lock_release(dg00x);
+ return err;
+}
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ snd_dg00x_stream_lock_release(dg00x);
+
+ return 0;
+}
+
+static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ mutex_unlock(&dg00x->mutex);
+ }
+
+ amdtp_dot_set_pcm_format(&dg00x->tx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ mutex_unlock(&dg00x->mutex);
+ }
+
+ amdtp_dot_set_pcm_format(&dg00x->rx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ mutex_lock(&dg00x->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ dg00x->substreams_counter--;
+
+ snd_dg00x_stream_stop_duplex(dg00x);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ mutex_lock(&dg00x->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ dg00x->substreams_counter--;
+
+ snd_dg00x_stream_stop_duplex(dg00x);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&dg00x->mutex);
+
+ err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&dg00x->tx_stream);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return err;
+}
+
+static int pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&dg00x->mutex);
+
+ err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+ if (err >= 0) {
+ amdtp_stream_pcm_prepare(&dg00x->rx_stream);
+ amdtp_dot_reset(&dg00x->rx_stream);
+ }
+
+ mutex_unlock(&dg00x->mutex);
+
+ return err;
+}
+
+static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&dg00x->tx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&dg00x->tx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&dg00x->rx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&dg00x->rx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_dg00x *dg00x = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&dg00x->tx_stream);
+}
+
+static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_dg00x *dg00x = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&dg00x->rx_stream);
+}
+
+static struct snd_pcm_ops pcm_capture_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_capture_hw_params,
+ .hw_free = pcm_capture_hw_free,
+ .prepare = pcm_capture_prepare,
+ .trigger = pcm_capture_trigger,
+ .pointer = pcm_capture_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops pcm_playback_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_playback_hw_params,
+ .hw_free = pcm_playback_hw_free,
+ .prepare = pcm_playback_prepare,
+ .trigger = pcm_playback_trigger,
+ .pointer = pcm_playback_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+};
+
+int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(dg00x->card, dg00x->card->driver, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = dg00x;
+ snprintf(pcm->name, sizeof(pcm->name),
+ "%s PCM", dg00x->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+ return 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-proc.c b/sound/firewire/digi00x/digi00x-proc.c
new file mode 100644
index 000000000000..a1d601f31165
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-proc.c
@@ -0,0 +1,99 @@
+/*
+ * digi00x-proc.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int get_optical_iface_mode(struct snd_dg00x *dg00x,
+ enum snd_dg00x_optical_mode *mode)
+{
+ __be32 data;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_OPT_IFACE_MODE,
+ &data, sizeof(data), 0);
+ if (err >= 0)
+ *mode = be32_to_cpu(data) & 0x01;
+
+ return err;
+}
+
+static void proc_read_clock(struct snd_info_entry *entry,
+ struct snd_info_buffer *buf)
+{
+ static const char *const source_name[] = {
+ [SND_DG00X_CLOCK_INTERNAL] = "internal",
+ [SND_DG00X_CLOCK_SPDIF] = "s/pdif",
+ [SND_DG00X_CLOCK_ADAT] = "adat",
+ [SND_DG00X_CLOCK_WORD] = "word clock",
+ };
+ static const char *const optical_name[] = {
+ [SND_DG00X_OPT_IFACE_MODE_ADAT] = "adat",
+ [SND_DG00X_OPT_IFACE_MODE_SPDIF] = "s/pdif",
+ };
+ struct snd_dg00x *dg00x = entry->private_data;
+ enum snd_dg00x_optical_mode mode;
+ unsigned int rate;
+ enum snd_dg00x_clock clock;
+ bool detect;
+
+ if (get_optical_iface_mode(dg00x, &mode) < 0)
+ return;
+ if (snd_dg00x_stream_get_local_rate(dg00x, &rate) < 0)
+ return;
+ if (snd_dg00x_stream_get_clock(dg00x, &clock) < 0)
+ return;
+
+ snd_iprintf(buf, "Optical mode: %s\n", optical_name[mode]);
+ snd_iprintf(buf, "Sampling Rate: %d\n", rate);
+ snd_iprintf(buf, "Clock Source: %s\n", source_name[clock]);
+
+ if (clock == SND_DG00X_CLOCK_INTERNAL)
+ return;
+
+ if (snd_dg00x_stream_check_external_clock(dg00x, &detect) < 0)
+ return;
+ snd_iprintf(buf, "External source: %s\n", detect ? "detected" : "not");
+ if (!detect)
+ return;
+
+ if (snd_dg00x_stream_get_external_rate(dg00x, &rate) >= 0)
+ snd_iprintf(buf, "External sampling rate: %d\n", rate);
+}
+
+void snd_dg00x_proc_init(struct snd_dg00x *dg00x)
+{
+ struct snd_info_entry *root, *entry;
+
+ /*
+ * All nodes are automatically removed at snd_card_disconnect(),
+ * by following to link list.
+ */
+ root = snd_info_create_card_entry(dg00x->card, "firewire",
+ dg00x->card->proc_root);
+ if (root == NULL)
+ return;
+
+ root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(root) < 0) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ entry = snd_info_create_card_entry(dg00x->card, "clock", root);
+ if (entry == NULL) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ snd_info_set_text_ops(entry, dg00x, proc_read_clock);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ snd_info_free_entry(root);
+ }
+}
diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c
new file mode 100644
index 000000000000..4d3b4ebbdd49
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-stream.c
@@ -0,0 +1,422 @@
+/*
+ * digi00x-stream.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+#define CALLBACK_TIMEOUT 500
+
+const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT] = {
+ [SND_DG00X_RATE_44100] = 44100,
+ [SND_DG00X_RATE_48000] = 48000,
+ [SND_DG00X_RATE_88200] = 88200,
+ [SND_DG00X_RATE_96000] = 96000,
+};
+
+/* Multi Bit Linear Audio data channels for each sampling transfer frequency. */
+const unsigned int
+snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT] = {
+ /* Analog/ADAT/SPDIF */
+ [SND_DG00X_RATE_44100] = (8 + 8 + 2),
+ [SND_DG00X_RATE_48000] = (8 + 8 + 2),
+ /* Analog/SPDIF */
+ [SND_DG00X_RATE_88200] = (8 + 2),
+ [SND_DG00X_RATE_96000] = (8 + 2),
+};
+
+int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x, unsigned int *rate)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = be32_to_cpu(reg) & 0x0f;
+ if (data < ARRAY_SIZE(snd_dg00x_stream_rates))
+ *rate = snd_dg00x_stream_rates[data];
+ else
+ err = -EIO;
+
+ return err;
+}
+
+int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ __be32 reg;
+ unsigned int i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_dg00x_stream_rates); i++) {
+ if (rate == snd_dg00x_stream_rates[i])
+ break;
+ }
+ if (i == ARRAY_SIZE(snd_dg00x_stream_rates))
+ return -EINVAL;
+
+ reg = cpu_to_be32(i);
+ return snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE,
+ &reg, sizeof(reg), 0);
+}
+
+int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
+ enum snd_dg00x_clock *clock)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_CLOCK_SOURCE,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ *clock = be32_to_cpu(reg) & 0x0f;
+ if (*clock >= SND_DG00X_CLOCK_COUNT)
+ err = -EIO;
+
+ return err;
+}
+
+int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x, bool *detect)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_DETECT_EXTERNAL,
+ &reg, sizeof(reg), 0);
+ if (err >= 0)
+ *detect = be32_to_cpu(reg) > 0;
+
+ return err;
+}
+
+int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_EXTERNAL_RATE,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = be32_to_cpu(reg) & 0x0f;
+ if (data < ARRAY_SIZE(snd_dg00x_stream_rates))
+ *rate = snd_dg00x_stream_rates[data];
+ /* This means desync. */
+ else
+ err = -EBUSY;
+
+ return err;
+}
+
+static void finish_session(struct snd_dg00x *dg00x)
+{
+ __be32 data = cpu_to_be32(0x00000003);
+
+ snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_SET,
+ &data, sizeof(data), 0);
+}
+
+static int begin_session(struct snd_dg00x *dg00x)
+{
+ __be32 data;
+ u32 curr;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_STATE,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+ curr = be32_to_cpu(data);
+
+ if (curr == 0)
+ curr = 2;
+
+ curr--;
+ while (curr > 0) {
+ data = cpu_to_be32(curr);
+ err = snd_fw_transaction(dg00x->unit,
+ TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE +
+ DG00X_OFFSET_STREAMING_SET,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+
+ msleep(20);
+ curr--;
+ }
+
+ return 0;
+error:
+ finish_session(dg00x);
+ return err;
+}
+
+static void release_resources(struct snd_dg00x *dg00x)
+{
+ __be32 data = 0;
+
+ /* Unregister isochronous channels for both direction. */
+ snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+ &data, sizeof(data), 0);
+
+ /* Release isochronous resources. */
+ fw_iso_resources_free(&dg00x->tx_resources);
+ fw_iso_resources_free(&dg00x->rx_resources);
+}
+
+static int keep_resources(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ unsigned int i;
+ __be32 data;
+ int err;
+
+ /* Check sampling rate. */
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (snd_dg00x_stream_rates[i] == rate)
+ break;
+ }
+ if (i == SND_DG00X_RATE_COUNT)
+ return -EINVAL;
+
+ /* Keep resources for out-stream. */
+ err = amdtp_dot_set_parameters(&dg00x->rx_stream, rate,
+ snd_dg00x_stream_pcm_channels[i]);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&dg00x->rx_resources,
+ amdtp_stream_get_max_payload(&dg00x->rx_stream),
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ return err;
+
+ /* Keep resources for in-stream. */
+ err = amdtp_dot_set_parameters(&dg00x->tx_stream, rate,
+ snd_dg00x_stream_pcm_channels[i]);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&dg00x->tx_resources,
+ amdtp_stream_get_max_payload(&dg00x->tx_stream),
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ /* Register isochronous channels for both direction. */
+ data = cpu_to_be32((dg00x->tx_resources.channel << 16) |
+ dg00x->rx_resources.channel);
+ err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ release_resources(dg00x);
+ return err;
+}
+
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ /* For out-stream. */
+ err = fw_iso_resources_init(&dg00x->rx_resources, dg00x->unit);
+ if (err < 0)
+ goto error;
+ err = amdtp_dot_init(&dg00x->rx_stream, dg00x->unit, AMDTP_OUT_STREAM);
+ if (err < 0)
+ goto error;
+
+ /* For in-stream. */
+ err = fw_iso_resources_init(&dg00x->tx_resources, dg00x->unit);
+ if (err < 0)
+ goto error;
+ err = amdtp_dot_init(&dg00x->tx_stream, dg00x->unit, AMDTP_IN_STREAM);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ snd_dg00x_stream_destroy_duplex(dg00x);
+ return err;
+}
+
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
+void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x)
+{
+ amdtp_stream_destroy(&dg00x->rx_stream);
+ fw_iso_resources_destroy(&dg00x->rx_resources);
+
+ amdtp_stream_destroy(&dg00x->tx_stream);
+ fw_iso_resources_destroy(&dg00x->tx_resources);
+}
+
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ unsigned int curr_rate;
+ int err = 0;
+
+ if (dg00x->substreams_counter == 0)
+ goto end;
+
+ /* Check current sampling rate. */
+ err = snd_dg00x_stream_get_local_rate(dg00x, &curr_rate);
+ if (err < 0)
+ goto error;
+ if (rate == 0)
+ rate = curr_rate;
+ if (curr_rate != rate ||
+ amdtp_streaming_error(&dg00x->tx_stream) ||
+ amdtp_streaming_error(&dg00x->rx_stream)) {
+ finish_session(dg00x);
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ release_resources(dg00x);
+ }
+
+ /*
+ * No packets are transmitted without receiving packets, reagardless of
+ * which source of clock is used.
+ */
+ if (!amdtp_stream_running(&dg00x->rx_stream)) {
+ err = snd_dg00x_stream_set_local_rate(dg00x, rate);
+ if (err < 0)
+ goto error;
+
+ err = keep_resources(dg00x, rate);
+ if (err < 0)
+ goto error;
+
+ err = begin_session(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = amdtp_stream_start(&dg00x->rx_stream,
+ dg00x->rx_resources.channel,
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&dg00x->rx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ /*
+ * The value of SYT field in transmitted packets is always 0x0000. Thus,
+ * duplex streams with timestamp synchronization cannot be built.
+ */
+ if (!amdtp_stream_running(&dg00x->tx_stream)) {
+ err = amdtp_stream_start(&dg00x->tx_stream,
+ dg00x->tx_resources.channel,
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&dg00x->tx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+end:
+ return err;
+error:
+ finish_session(dg00x);
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ release_resources(dg00x);
+
+ return err;
+}
+
+void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x)
+{
+ if (dg00x->substreams_counter > 0)
+ return;
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ finish_session(dg00x);
+ release_resources(dg00x);
+
+ /*
+ * Just after finishing the session, the device may lost transmitting
+ * functionality for a short time.
+ */
+ msleep(50);
+}
+
+void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x)
+{
+ fw_iso_resources_update(&dg00x->tx_resources);
+ fw_iso_resources_update(&dg00x->rx_resources);
+
+ amdtp_stream_update(&dg00x->tx_stream);
+ amdtp_stream_update(&dg00x->rx_stream);
+}
+
+void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x)
+{
+ dg00x->dev_lock_changed = true;
+ wake_up(&dg00x->hwdep_wait);
+}
+
+int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ /* user land lock this */
+ if (dg00x->dev_lock_count < 0) {
+ err = -EBUSY;
+ goto end;
+ }
+
+ /* this is the first time */
+ if (dg00x->dev_lock_count++ == 0)
+ snd_dg00x_stream_lock_changed(dg00x);
+ err = 0;
+end:
+ spin_unlock_irq(&dg00x->lock);
+ return err;
+}
+
+void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x)
+{
+ spin_lock_irq(&dg00x->lock);
+
+ if (WARN_ON(dg00x->dev_lock_count <= 0))
+ goto end;
+ if (--dg00x->dev_lock_count == 0)
+ snd_dg00x_stream_lock_changed(dg00x);
+end:
+ spin_unlock_irq(&dg00x->lock);
+}
diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c
new file mode 100644
index 000000000000..554324d8c602
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-transaction.c
@@ -0,0 +1,137 @@
+/*
+ * digi00x-transaction.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/asound.h>
+#include "digi00x.h"
+
+static int fill_midi_message(struct snd_rawmidi_substream *substream, u8 *buf)
+{
+ int bytes;
+
+ buf[0] = 0x80;
+ bytes = snd_rawmidi_transmit_peek(substream, buf + 1, 2);
+ if (bytes >= 0)
+ buf[3] = 0xc0 | bytes;
+
+ return bytes;
+}
+
+static void handle_midi_control(struct snd_dg00x *dg00x, __be32 *buf,
+ unsigned int length)
+{
+ struct snd_rawmidi_substream *substream;
+ unsigned int i;
+ unsigned int len;
+ u8 *b;
+
+ substream = ACCESS_ONCE(dg00x->in_control);
+ if (substream == NULL)
+ return;
+
+ length /= 4;
+
+ for (i = 0; i < length; i++) {
+ b = (u8 *)&buf[i];
+ len = b[3] & 0xf;
+ if (len > 0)
+ snd_rawmidi_receive(dg00x->in_control, b + 1, len);
+ }
+}
+
+static void handle_unknown_message(struct snd_dg00x *dg00x,
+ unsigned long long offset, __be32 *buf)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+ dg00x->msg = be32_to_cpu(*buf);
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+
+ wake_up(&dg00x->hwdep_wait);
+}
+
+static void handle_message(struct fw_card *card, struct fw_request *request,
+ int tcode, int destination, int source,
+ int generation, unsigned long long offset,
+ void *data, size_t length, void *callback_data)
+{
+ struct snd_dg00x *dg00x = callback_data;
+ __be32 *buf = (__be32 *)data;
+
+ if (offset == dg00x->async_handler.offset)
+ handle_unknown_message(dg00x, offset, buf);
+ else if (offset == dg00x->async_handler.offset + 4)
+ handle_midi_control(dg00x, buf, length);
+
+ fw_send_response(card, request, RCODE_COMPLETE);
+}
+
+int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x)
+{
+ struct fw_device *device = fw_parent_device(dg00x->unit);
+ __be32 data[2];
+ int err;
+
+ /* Unknown. 4bytes. */
+ data[0] = cpu_to_be32((device->card->node_id << 16) |
+ (dg00x->async_handler.offset >> 32));
+ data[1] = cpu_to_be32(dg00x->async_handler.offset);
+ err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MESSAGE_ADDR,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return err;
+
+ /* Asynchronous transactions for MIDI control message. */
+ data[0] = cpu_to_be32((device->card->node_id << 16) |
+ (dg00x->async_handler.offset >> 32));
+ data[1] = cpu_to_be32(dg00x->async_handler.offset + 4);
+ return snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MIDI_CTL_ADDR,
+ &data, sizeof(data), 0);
+}
+
+int snd_dg00x_transaction_register(struct snd_dg00x *dg00x)
+{
+ static const struct fw_address_region resp_register_region = {
+ .start = 0xffffe0000000ull,
+ .end = 0xffffe000ffffull,
+ };
+ int err;
+
+ dg00x->async_handler.length = 12;
+ dg00x->async_handler.address_callback = handle_message;
+ dg00x->async_handler.callback_data = dg00x;
+
+ err = fw_core_add_address_handler(&dg00x->async_handler,
+ &resp_register_region);
+ if (err < 0)
+ return err;
+
+ err = snd_dg00x_transaction_reregister(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_fw_async_midi_port_init(&dg00x->out_control, dg00x->unit,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MMC,
+ 4, fill_midi_message);
+ if (err < 0)
+ goto error;
+
+ return err;
+error:
+ fw_core_remove_address_handler(&dg00x->async_handler);
+ dg00x->async_handler.address_callback = NULL;
+ return err;
+}
+
+void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x)
+{
+ snd_fw_async_midi_port_destroy(&dg00x->out_control);
+ fw_core_remove_address_handler(&dg00x->async_handler);
+}
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
new file mode 100644
index 000000000000..bbe3be7fea9b
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x.c
@@ -0,0 +1,171 @@
+/*
+ * digi00x.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+MODULE_DESCRIPTION("Digidesign Digi 002/003 family Driver");
+MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>");
+MODULE_LICENSE("GPL v2");
+
+#define VENDOR_DIGIDESIGN 0x00a07e
+#define MODEL_DIGI00X 0x000002
+
+static int name_card(struct snd_dg00x *dg00x)
+{
+ struct fw_device *fw_dev = fw_parent_device(dg00x->unit);
+ char name[32] = {0};
+ char *model;
+ int err;
+
+ err = fw_csr_string(dg00x->unit->directory, CSR_MODEL, name,
+ sizeof(name));
+ if (err < 0)
+ return err;
+
+ model = skip_spaces(name);
+
+ strcpy(dg00x->card->driver, "Digi00x");
+ strcpy(dg00x->card->shortname, model);
+ strcpy(dg00x->card->mixername, model);
+ snprintf(dg00x->card->longname, sizeof(dg00x->card->longname),
+ "Digidesign %s, GUID %08x%08x at %s, S%d", model,
+ cpu_to_be32(fw_dev->config_rom[3]),
+ cpu_to_be32(fw_dev->config_rom[4]),
+ dev_name(&dg00x->unit->device), 100 << fw_dev->max_speed);
+
+ return 0;
+}
+
+static void dg00x_card_free(struct snd_card *card)
+{
+ struct snd_dg00x *dg00x = card->private_data;
+
+ snd_dg00x_stream_destroy_duplex(dg00x);
+ snd_dg00x_transaction_unregister(dg00x);
+
+ fw_unit_put(dg00x->unit);
+
+ mutex_destroy(&dg00x->mutex);
+}
+
+static int snd_dg00x_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_card *card;
+ struct snd_dg00x *dg00x;
+ int err;
+
+ /* create card */
+ err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
+ sizeof(struct snd_dg00x), &card);
+ if (err < 0)
+ return err;
+ card->private_free = dg00x_card_free;
+
+ /* initialize myself */
+ dg00x = card->private_data;
+ dg00x->card = card;
+ dg00x->unit = fw_unit_get(unit);
+
+ mutex_init(&dg00x->mutex);
+ spin_lock_init(&dg00x->lock);
+ init_waitqueue_head(&dg00x->hwdep_wait);
+
+ err = name_card(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_stream_init_duplex(dg00x);
+ if (err < 0)
+ goto error;
+
+ snd_dg00x_proc_init(dg00x);
+
+ err = snd_dg00x_create_pcm_devices(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_create_midi_devices(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_create_hwdep_device(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_transaction_register(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ dev_set_drvdata(&unit->device, dg00x);
+
+ return err;
+error:
+ snd_card_free(card);
+ return err;
+}
+
+static void snd_dg00x_update(struct fw_unit *unit)
+{
+ struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+
+ snd_dg00x_transaction_reregister(dg00x);
+
+ mutex_lock(&dg00x->mutex);
+ snd_dg00x_stream_update_duplex(dg00x);
+ mutex_unlock(&dg00x->mutex);
+}
+
+static void snd_dg00x_remove(struct fw_unit *unit)
+{
+ struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(dg00x->card);
+}
+
+static const struct ieee1394_device_id snd_dg00x_id_table[] = {
+ /* Both of 002/003 use the same ID. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = VENDOR_DIGIDESIGN,
+ .model_id = MODEL_DIGI00X,
+ },
+ {}
+};
+MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table);
+
+static struct fw_driver dg00x_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "snd-firewire-digi00x",
+ .bus = &fw_bus_type,
+ },
+ .probe = snd_dg00x_probe,
+ .update = snd_dg00x_update,
+ .remove = snd_dg00x_remove,
+ .id_table = snd_dg00x_id_table,
+};
+
+static int __init snd_dg00x_init(void)
+{
+ return driver_register(&dg00x_driver.driver);
+}
+
+static void __exit snd_dg00x_exit(void)
+{
+ driver_unregister(&dg00x_driver.driver);
+}
+
+module_init(snd_dg00x_init);
+module_exit(snd_dg00x_exit);
diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h
new file mode 100644
index 000000000000..907e73993677
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x.h
@@ -0,0 +1,157 @@
+/*
+ * digi00x.h - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_DIGI00X_H_INCLUDED
+#define SOUND_DIGI00X_H_INCLUDED
+
+#include <linux/compat.h>
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+
+#include "../lib.h"
+#include "../iso-resources.h"
+#include "../amdtp-stream.h"
+
+struct snd_dg00x {
+ struct snd_card *card;
+ struct fw_unit *unit;
+
+ struct mutex mutex;
+ spinlock_t lock;
+
+ struct amdtp_stream tx_stream;
+ struct fw_iso_resources tx_resources;
+
+ struct amdtp_stream rx_stream;
+ struct fw_iso_resources rx_resources;
+
+ unsigned int substreams_counter;
+
+ /* for uapi */
+ int dev_lock_count;
+ bool dev_lock_changed;
+ wait_queue_head_t hwdep_wait;
+
+ /* For asynchronous messages. */
+ struct fw_address_handler async_handler;
+ u32 msg;
+
+ /* For asynchronous MIDI controls. */
+ struct snd_rawmidi_substream *in_control;
+ struct snd_fw_async_midi_port out_control;
+};
+
+#define DG00X_ADDR_BASE 0xffffe0000000ull
+
+#define DG00X_OFFSET_STREAMING_STATE 0x0000
+#define DG00X_OFFSET_STREAMING_SET 0x0004
+#define DG00X_OFFSET_MIDI_CTL_ADDR 0x0008
+/* For LSB of the address 0x000c */
+/* unknown 0x0010 */
+#define DG00X_OFFSET_MESSAGE_ADDR 0x0014
+/* For LSB of the address 0x0018 */
+/* unknown 0x001c */
+/* unknown 0x0020 */
+/* not used 0x0024--0x00ff */
+#define DG00X_OFFSET_ISOC_CHANNELS 0x0100
+/* unknown 0x0104 */
+/* unknown 0x0108 */
+/* unknown 0x010c */
+#define DG00X_OFFSET_LOCAL_RATE 0x0110
+#define DG00X_OFFSET_EXTERNAL_RATE 0x0114
+#define DG00X_OFFSET_CLOCK_SOURCE 0x0118
+#define DG00X_OFFSET_OPT_IFACE_MODE 0x011c
+/* unknown 0x0120 */
+/* Mixer control on/off 0x0124 */
+/* unknown 0x0128 */
+#define DG00X_OFFSET_DETECT_EXTERNAL 0x012c
+/* unknown 0x0138 */
+#define DG00X_OFFSET_MMC 0x0400
+
+enum snd_dg00x_rate {
+ SND_DG00X_RATE_44100 = 0,
+ SND_DG00X_RATE_48000,
+ SND_DG00X_RATE_88200,
+ SND_DG00X_RATE_96000,
+ SND_DG00X_RATE_COUNT,
+};
+
+enum snd_dg00x_clock {
+ SND_DG00X_CLOCK_INTERNAL = 0,
+ SND_DG00X_CLOCK_SPDIF,
+ SND_DG00X_CLOCK_ADAT,
+ SND_DG00X_CLOCK_WORD,
+ SND_DG00X_CLOCK_COUNT,
+};
+
+enum snd_dg00x_optical_mode {
+ SND_DG00X_OPT_IFACE_MODE_ADAT = 0,
+ SND_DG00X_OPT_IFACE_MODE_SPDIF,
+ SND_DG00X_OPT_IFACE_MODE_COUNT,
+};
+
+#define DOT_MIDI_IN_PORTS 1
+#define DOT_MIDI_OUT_PORTS 2
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir);
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels);
+void amdtp_dot_reset(struct amdtp_stream *s);
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi);
+
+int snd_dg00x_transaction_register(struct snd_dg00x *dg00x);
+int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x);
+void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x);
+
+extern const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT];
+extern const unsigned int snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT];
+int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate);
+int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate);
+int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate);
+int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
+ enum snd_dg00x_clock *clock);
+int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x,
+ bool *detect);
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x);
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate);
+void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x);
+
+void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x);
+int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x);
+
+void snd_dg00x_proc_init(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x);
+#endif
diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c
index 0619597e3a3f..cce19768f43d 100644
--- a/sound/firewire/fcp.c
+++ b/sound/firewire/fcp.c
@@ -17,7 +17,7 @@
#include <linux/delay.h>
#include "fcp.h"
#include "lib.h"
-#include "amdtp.h"
+#include "amdtp-stream.h"
#define CTS_AVC 0x00
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index c94a432f7cc6..d5b19bc11e59 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -138,12 +138,12 @@ get_hardware_info(struct snd_efw *efw)
efw->midi_out_ports = hwinfo->midi_out_ports;
efw->midi_in_ports = hwinfo->midi_in_ports;
- if (hwinfo->amdtp_tx_pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_tx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_tx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM) {
+ if (hwinfo->amdtp_tx_pcm_channels > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_tx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_tx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM) {
err = -ENOSYS;
goto end;
}
diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h
index 084d414b228c..c7cb7deafe48 100644
--- a/sound/firewire/fireworks/fireworks.h
+++ b/sound/firewire/fireworks/fireworks.h
@@ -29,7 +29,7 @@
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
#include "../lib.h"
diff --git a/sound/firewire/fireworks/fireworks_midi.c b/sound/firewire/fireworks/fireworks_midi.c
index cf9c65260439..fba01bbba456 100644
--- a/sound/firewire/fireworks/fireworks_midi.c
+++ b/sound/firewire/fireworks/fireworks_midi.c
@@ -73,10 +73,10 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&efw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&efw->tx_stream,
+ amdtp_am824_midi_trigger(&efw->tx_stream,
substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&efw->tx_stream,
+ amdtp_am824_midi_trigger(&efw->tx_stream,
substrm->number, NULL);
spin_unlock_irqrestore(&efw->lock, flags);
@@ -90,11 +90,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&efw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&efw->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&efw->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&efw->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&efw->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&efw->lock, flags);
}
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index c30b2ffa8dfb..d27135bac513 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -159,11 +159,11 @@ pcm_init_hw_params(struct snd_efw *efw,
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
s = &efw->tx_stream;
pcm_channels = efw->pcm_capture_channels;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
s = &efw->rx_stream;
pcm_channels = efw->pcm_playback_channels;
}
@@ -187,7 +187,7 @@ pcm_init_hw_params(struct snd_efw *efw,
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(s, runtime);
end:
return err;
}
@@ -253,7 +253,8 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&efw->capture_substreams);
- amdtp_stream_set_pcm_format(&efw->tx_stream, params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&efw->tx_stream, params_format(hw_params));
return 0;
}
@@ -270,7 +271,8 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&efw->playback_substreams);
- amdtp_stream_set_pcm_format(&efw->rx_stream, params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&efw->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 7e353f1f7bff..759f6e3ed44a 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -31,7 +31,7 @@ init_stream(struct snd_efw *efw, struct amdtp_stream *stream)
if (err < 0)
goto end;
- err = amdtp_stream_init(stream, efw->unit, s_dir, CIP_BLOCKING);
+ err = amdtp_am824_init(stream, efw->unit, s_dir, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
cmp_connection_destroy(conn);
@@ -73,8 +73,10 @@ start_stream(struct snd_efw *efw, struct amdtp_stream *stream,
midi_ports = efw->midi_in_ports;
}
- amdtp_stream_set_parameters(stream, sampling_rate,
- pcm_channels, midi_ports);
+ err = amdtp_am824_set_parameters(stream, sampling_rate,
+ pcm_channels, midi_ports, false);
+ if (err < 0)
+ goto end;
/* establish connection via CMP */
err = cmp_connection_establish(conn,
diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c
index 7409edba9f06..edf1c8bd25a6 100644
--- a/sound/firewire/lib.c
+++ b/sound/firewire/lib.c
@@ -9,6 +9,7 @@
#include <linux/device.h>
#include <linux/firewire.h>
#include <linux/module.h>
+#include <linux/slab.h>
#include "lib.h"
#define ERROR_RETRY_DELAY_MS 20
@@ -66,6 +67,143 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode,
}
EXPORT_SYMBOL(snd_fw_transaction);
+static void async_midi_port_callback(struct fw_card *card, int rcode,
+ void *data, size_t length,
+ void *callback_data)
+{
+ struct snd_fw_async_midi_port *port = callback_data;
+ struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream);
+
+ if (rcode == RCODE_COMPLETE && substream != NULL)
+ snd_rawmidi_transmit_ack(substream, port->consume_bytes);
+ else if (!rcode_is_permanent_error(rcode))
+ /* To start next transaction immediately for recovery. */
+ port->next_ktime = ktime_set(0, 0);
+ else
+ /* Don't continue processing. */
+ port->error = true;
+
+ port->idling = true;
+
+ if (!snd_rawmidi_transmit_empty(substream))
+ schedule_work(&port->work);
+}
+
+static void midi_port_work(struct work_struct *work)
+{
+ struct snd_fw_async_midi_port *port =
+ container_of(work, struct snd_fw_async_midi_port, work);
+ struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream);
+ int generation;
+ int type;
+
+ /* Under transacting or error state. */
+ if (!port->idling || port->error)
+ return;
+
+ /* Nothing to do. */
+ if (substream == NULL || snd_rawmidi_transmit_empty(substream))
+ return;
+
+ /* Do it in next chance. */
+ if (ktime_after(port->next_ktime, ktime_get())) {
+ schedule_work(&port->work);
+ return;
+ }
+
+ /*
+ * Fill the buffer. The callee must use snd_rawmidi_transmit_peek().
+ * Later, snd_rawmidi_transmit_ack() is called.
+ */
+ memset(port->buf, 0, port->len);
+ port->consume_bytes = port->fill(substream, port->buf);
+ if (port->consume_bytes <= 0) {
+ /* Do it in next chance, immediately. */
+ if (port->consume_bytes == 0) {
+ port->next_ktime = ktime_set(0, 0);
+ schedule_work(&port->work);
+ } else {
+ /* Fatal error. */
+ port->error = true;
+ }
+ return;
+ }
+
+ /* Calculate type of transaction. */
+ if (port->len == 4)
+ type = TCODE_WRITE_QUADLET_REQUEST;
+ else
+ type = TCODE_WRITE_BLOCK_REQUEST;
+
+ /* Set interval to next transaction. */
+ port->next_ktime = ktime_add_ns(ktime_get(),
+ port->consume_bytes * 8 * NSEC_PER_SEC / 31250);
+
+ /* Start this transaction. */
+ port->idling = false;
+
+ /*
+ * In Linux FireWire core, when generation is updated with memory
+ * barrier, node id has already been updated. In this module, After
+ * this smp_rmb(), load/store instructions to memory are completed.
+ * Thus, both of generation and node id are available with recent
+ * values. This is a light-serialization solution to handle bus reset
+ * events on IEEE 1394 bus.
+ */
+ generation = port->parent->generation;
+ smp_rmb();
+
+ fw_send_request(port->parent->card, &port->transaction, type,
+ port->parent->node_id, generation,
+ port->parent->max_speed, port->addr,
+ port->buf, port->len, async_midi_port_callback,
+ port);
+}
+
+/**
+ * snd_fw_async_midi_port_init - initialize asynchronous MIDI port structure
+ * @port: the asynchronous MIDI port to initialize
+ * @unit: the target of the asynchronous transaction
+ * @addr: the address to which transactions are transferred
+ * @len: the length of transaction
+ * @fill: the callback function to fill given buffer, and returns the
+ * number of consumed bytes for MIDI message.
+ *
+ */
+int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port,
+ struct fw_unit *unit, u64 addr, unsigned int len,
+ snd_fw_async_midi_port_fill fill)
+{
+ port->len = DIV_ROUND_UP(len, 4) * 4;
+ port->buf = kzalloc(port->len, GFP_KERNEL);
+ if (port->buf == NULL)
+ return -ENOMEM;
+
+ port->parent = fw_parent_device(unit);
+ port->addr = addr;
+ port->fill = fill;
+ port->idling = true;
+ port->next_ktime = ktime_set(0, 0);
+ port->error = false;
+
+ INIT_WORK(&port->work, midi_port_work);
+
+ return 0;
+}
+EXPORT_SYMBOL(snd_fw_async_midi_port_init);
+
+/**
+ * snd_fw_async_midi_port_destroy - free asynchronous MIDI port structure
+ * @port: the asynchronous MIDI port structure
+ */
+void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port)
+{
+ snd_fw_async_midi_port_finish(port);
+ cancel_work_sync(&port->work);
+ kfree(port->buf);
+}
+EXPORT_SYMBOL(snd_fw_async_midi_port_destroy);
+
MODULE_DESCRIPTION("FireWire audio helper functions");
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
MODULE_LICENSE("GPL v2");
diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h
index 02cfabc9c3c4..f3f6f84c48d6 100644
--- a/sound/firewire/lib.h
+++ b/sound/firewire/lib.h
@@ -3,6 +3,8 @@
#include <linux/firewire-constants.h>
#include <linux/types.h>
+#include <linux/sched.h>
+#include <sound/rawmidi.h>
struct fw_unit;
@@ -20,4 +22,58 @@ static inline bool rcode_is_permanent_error(int rcode)
return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR;
}
+struct snd_fw_async_midi_port;
+typedef int (*snd_fw_async_midi_port_fill)(
+ struct snd_rawmidi_substream *substream,
+ u8 *buf);
+
+struct snd_fw_async_midi_port {
+ struct fw_device *parent;
+ struct work_struct work;
+ bool idling;
+ ktime_t next_ktime;
+ bool error;
+
+ u64 addr;
+ struct fw_transaction transaction;
+
+ u8 *buf;
+ unsigned int len;
+
+ struct snd_rawmidi_substream *substream;
+ snd_fw_async_midi_port_fill fill;
+ unsigned int consume_bytes;
+};
+
+int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port,
+ struct fw_unit *unit, u64 addr, unsigned int len,
+ snd_fw_async_midi_port_fill fill);
+void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port);
+
+/**
+ * snd_fw_async_midi_port_run - run transactions for the async MIDI port
+ * @port: the asynchronous MIDI port
+ * @substream: the MIDI substream
+ */
+static inline void
+snd_fw_async_midi_port_run(struct snd_fw_async_midi_port *port,
+ struct snd_rawmidi_substream *substream)
+{
+ if (!port->error) {
+ port->substream = substream;
+ schedule_work(&port->work);
+ }
+}
+
+/**
+ * snd_fw_async_midi_port_finish - finish the asynchronous MIDI port
+ * @port: the asynchronous MIDI port
+ */
+static inline void
+snd_fw_async_midi_port_finish(struct snd_fw_async_midi_port *port)
+{
+ port->substream = NULL;
+ port->error = false;
+}
+
#endif
diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c
index 540a30338516..37a86cf69cbf 100644
--- a/sound/firewire/oxfw/oxfw-midi.c
+++ b/sound/firewire/oxfw/oxfw-midi.c
@@ -90,11 +90,11 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&oxfw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&oxfw->tx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&oxfw->tx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&oxfw->tx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&oxfw->tx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&oxfw->lock, flags);
}
@@ -107,11 +107,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&oxfw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&oxfw->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&oxfw->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&oxfw->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&oxfw->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&oxfw->lock, flags);
}
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 9c73930d0278..8d233417695d 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -134,11 +134,11 @@ static int init_hw_params(struct snd_oxfw *oxfw,
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
stream = &oxfw->tx_stream;
formats = oxfw->tx_stream_formats;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
stream = &oxfw->rx_stream;
formats = oxfw->rx_stream_formats;
}
@@ -158,7 +158,7 @@ static int init_hw_params(struct snd_oxfw *oxfw,
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(stream, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(stream, runtime);
end:
return err;
}
@@ -244,7 +244,7 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&oxfw->mutex);
}
- amdtp_stream_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
+ amdtp_am824_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
return 0;
}
@@ -265,7 +265,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&oxfw->mutex);
}
- amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
+ amdtp_am824_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 77ad5b98e806..2c63058bd245 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -155,7 +155,10 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream,
err = -EINVAL;
goto end;
}
- amdtp_stream_set_parameters(stream, rate, pcm_channels, midi_ports);
+ err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports,
+ false);
+ if (err < 0)
+ goto end;
err = cmp_connection_establish(conn,
amdtp_stream_get_max_payload(stream));
@@ -225,7 +228,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw,
if (err < 0)
goto end;
- err = amdtp_stream_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING);
+ err = amdtp_am824_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
cmp_connection_destroy(conn);
@@ -238,9 +241,12 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw,
* packets. As a result, next isochronous packet includes more data
* blocks than IEC 61883-6 defines.
*/
- if (stream == &oxfw->tx_stream)
+ if (stream == &oxfw->tx_stream) {
oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK |
CIP_JUMBO_PAYLOAD;
+ if (oxfw->wrong_dbs)
+ oxfw->tx_stream.flags |= CIP_WRONG_DBS;
+ }
end:
return err;
}
@@ -480,8 +486,8 @@ int snd_oxfw_stream_parse_format(u8 *format,
}
}
- if (formation->pcm > AMDTP_MAX_CHANNELS_FOR_PCM ||
- formation->midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+ if (formation->pcm > AM824_MAX_CHANNELS_FOR_PCM ||
+ formation->midi > AM824_MAX_CHANNELS_FOR_MIDI)
return -ENOSYS;
return 0;
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 8c6ce019f437..d606e3a9ce97 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -19,6 +19,8 @@
#define VENDOR_BEHRINGER 0x001564
#define VENDOR_LACIE 0x00d04b
+#define MODEL_SATELLITE 0x00200f
+
#define SPECIFIER_1394TA 0x00a02d
#define VERSION_AVC 0x010001
@@ -129,6 +131,31 @@ static void oxfw_card_free(struct snd_card *card)
mutex_destroy(&oxfw->mutex);
}
+static void detect_quirks(struct snd_oxfw *oxfw)
+{
+ struct fw_device *fw_dev = fw_parent_device(oxfw->unit);
+ struct fw_csr_iterator it;
+ int key, val;
+ int vendor, model;
+
+ /* Seek from Root Directory of Config ROM. */
+ vendor = model = 0;
+ fw_csr_iterator_init(&it, fw_dev->config_rom + 5);
+ while (fw_csr_iterator_next(&it, &key, &val)) {
+ if (key == CSR_VENDOR)
+ vendor = val;
+ else if (key == CSR_MODEL)
+ model = val;
+ }
+
+ /*
+ * Mackie Onyx Satellite with base station has a quirk to report a wrong
+ * value in 'dbs' field of CIP header against its format information.
+ */
+ if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE)
+ oxfw->wrong_dbs = true;
+}
+
static int oxfw_probe(struct fw_unit *unit,
const struct ieee1394_device_id *id)
{
@@ -157,6 +184,8 @@ static int oxfw_probe(struct fw_unit *unit,
if (err < 0)
goto error;
+ detect_quirks(oxfw);
+
err = name_card(oxfw);
if (err < 0)
goto error;
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index cace5ad4fe76..8392c424ad1d 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -28,7 +28,7 @@
#include "../fcp.h"
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
struct device_info {
@@ -49,6 +49,7 @@ struct snd_oxfw {
struct mutex mutex;
spinlock_t lock;
+ bool wrong_dbs;
bool has_output;
u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
diff --git a/sound/firewire/tascam/Makefile b/sound/firewire/tascam/Makefile
new file mode 100644
index 000000000000..0fc955d5bd15
--- /dev/null
+++ b/sound/firewire/tascam/Makefile
@@ -0,0 +1,4 @@
+snd-firewire-tascam-objs := tascam-proc.o amdtp-tascam.o tascam-stream.o \
+ tascam-pcm.o tascam-hwdep.o tascam-transaction.o \
+ tascam-midi.o tascam.o
+obj-$(CONFIG_SND_FIREWIRE_TASCAM) += snd-firewire-tascam.o
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
new file mode 100644
index 000000000000..9dd0fccd5ccc
--- /dev/null
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -0,0 +1,243 @@
+/*
+ * amdtp-tascam.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "tascam.h"
+
+#define AMDTP_FMT_TSCM_TX 0x1e
+#define AMDTP_FMT_TSCM_RX 0x3e
+
+struct amdtp_tscm {
+ unsigned int pcm_channels;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+};
+
+int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate)
+{
+ struct amdtp_tscm *p = s->protocol;
+ unsigned int data_channels;
+
+ if (amdtp_stream_running(s))
+ return -EBUSY;
+
+ data_channels = p->pcm_channels;
+
+ /* Packets in in-stream have extra 2 data channels. */
+ if (s->direction == AMDTP_IN_STREAM)
+ data_channels += 2;
+
+ return amdtp_stream_set_parameters(s, rate, data_channels);
+}
+
+static void write_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32(*src);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32(*src << 16);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ /* The first data channel is for event counter. */
+ buffer += 1;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[c]);
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_tscm *p = s->protocol;
+ unsigned int channels, i, c;
+
+ channels = p->pcm_channels;
+
+ for (i = 0; i < data_blocks; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[c] = 0x00000000;
+ buffer += s->data_block_quadlets;
+ }
+}
+
+int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ /*
+ * Our implementation allows this protocol to deliver 24 bit sample in
+ * 32bit data channel.
+ */
+ err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ if (err < 0)
+ return err;
+
+ return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_tscm *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
+ struct snd_pcm_substream *pcm;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (data_blocks > 0 && pcm)
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+
+ /* A place holder for control messages. */
+
+ return data_blocks;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
+ struct snd_pcm_substream *pcm;
+
+ /* This field is not used. */
+ *syt = 0x0000;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm)
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ else
+ write_pcm_silence(s, buffer, data_blocks);
+
+ return data_blocks;
+}
+
+int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, unsigned int pcm_channels)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+ struct amdtp_tscm *p;
+ unsigned int fmt;
+ int err;
+
+ if (dir == AMDTP_IN_STREAM) {
+ fmt = AMDTP_FMT_TSCM_TX;
+ process_data_blocks = process_tx_data_blocks;
+ } else {
+ fmt = AMDTP_FMT_TSCM_RX;
+ process_data_blocks = process_rx_data_blocks;
+ }
+
+ err = amdtp_stream_init(s, unit, dir,
+ CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt,
+ process_data_blocks, sizeof(struct amdtp_tscm));
+ if (err < 0)
+ return 0;
+
+ /* Use fixed value for FDF field. */
+ s->fdf = 0x00;
+
+ /* This protocol uses fixed number of data channels for PCM samples. */
+ p = s->protocol;
+ p->pcm_channels = pcm_channels;
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c
new file mode 100644
index 000000000000..131267c3a042
--- /dev/null
+++ b/sound/firewire/tascam/tascam-hwdep.c
@@ -0,0 +1,201 @@
+/*
+ * tascam-hwdep.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node information
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ */
+
+#include "tascam.h"
+
+static long hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
+ long count)
+{
+ union snd_firewire_event event;
+
+ memset(&event, 0, sizeof(event));
+
+ event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+ event.lock_status.status = (tscm->dev_lock_count > 0);
+ tscm->dev_lock_changed = false;
+
+ count = min_t(long, count, sizeof(event.lock_status));
+
+ if (copy_to_user(buf, &event, count))
+ return -EFAULT;
+
+ return count;
+}
+
+static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count,
+ loff_t *offset)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+ DEFINE_WAIT(wait);
+ union snd_firewire_event event;
+
+ spin_lock_irq(&tscm->lock);
+
+ while (!tscm->dev_lock_changed) {
+ prepare_to_wait(&tscm->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&tscm->lock);
+ schedule();
+ finish_wait(&tscm->hwdep_wait, &wait);
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ spin_lock_irq(&tscm->lock);
+ }
+
+ memset(&event, 0, sizeof(event));
+ count = hwdep_read_locked(tscm, buf, count);
+ spin_unlock_irq(&tscm->lock);
+
+ return count;
+}
+
+static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file,
+ poll_table *wait)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+ unsigned int events;
+
+ poll_wait(file, &tscm->hwdep_wait, wait);
+
+ spin_lock_irq(&tscm->lock);
+ if (tscm->dev_lock_changed)
+ events = POLLIN | POLLRDNORM;
+ else
+ events = 0;
+ spin_unlock_irq(&tscm->lock);
+
+ return events;
+}
+
+static int hwdep_get_info(struct snd_tscm *tscm, void __user *arg)
+{
+ struct fw_device *dev = fw_parent_device(tscm->unit);
+ struct snd_firewire_get_info info;
+
+ memset(&info, 0, sizeof(info));
+ info.type = SNDRV_FIREWIRE_TYPE_TASCAM;
+ info.card = dev->card->index;
+ *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+ *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+ strlcpy(info.device_name, dev_name(&dev->device),
+ sizeof(info.device_name));
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+static int hwdep_lock(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ if (tscm->dev_lock_count == 0) {
+ tscm->dev_lock_count = -1;
+ err = 0;
+ } else {
+ err = -EBUSY;
+ }
+
+ spin_unlock_irq(&tscm->lock);
+
+ return err;
+}
+
+static int hwdep_unlock(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ if (tscm->dev_lock_count == -1) {
+ tscm->dev_lock_count = 0;
+ err = 0;
+ } else {
+ err = -EBADFD;
+ }
+
+ spin_unlock_irq(&tscm->lock);
+
+ return err;
+}
+
+static int hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+
+ spin_lock_irq(&tscm->lock);
+ if (tscm->dev_lock_count == -1)
+ tscm->dev_lock_count = 0;
+ spin_unlock_irq(&tscm->lock);
+
+ return 0;
+}
+
+static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+
+ switch (cmd) {
+ case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+ return hwdep_get_info(tscm, (void __user *)arg);
+ case SNDRV_FIREWIRE_IOCTL_LOCK:
+ return hwdep_lock(tscm);
+ case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+ return hwdep_unlock(tscm);
+ default:
+ return -ENOIOCTLCMD;
+ }
+}
+
+#ifdef CONFIG_COMPAT
+static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return hwdep_ioctl(hwdep, file, cmd,
+ (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+ .read = hwdep_read,
+ .release = hwdep_release,
+ .poll = hwdep_poll,
+ .ioctl = hwdep_ioctl,
+ .ioctl_compat = hwdep_compat_ioctl,
+};
+
+int snd_tscm_create_hwdep_device(struct snd_tscm *tscm)
+{
+ struct snd_hwdep *hwdep;
+ int err;
+
+ err = snd_hwdep_new(tscm->card, "Tascam", 0, &hwdep);
+ if (err < 0)
+ return err;
+
+ strcpy(hwdep->name, "Tascam");
+ hwdep->iface = SNDRV_HWDEP_IFACE_FW_TASCAM;
+ hwdep->ops = hwdep_ops;
+ hwdep->private_data = tscm;
+ hwdep->exclusive = true;
+
+ return err;
+}
diff --git a/sound/firewire/tascam/tascam-midi.c b/sound/firewire/tascam/tascam-midi.c
new file mode 100644
index 000000000000..41f842079d9d
--- /dev/null
+++ b/sound/firewire/tascam/tascam-midi.c
@@ -0,0 +1,135 @@
+/*
+ * tascam-midi.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+static int midi_capture_open(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_playback_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+
+ /* Initialize internal status. */
+ tscm->running_status[substream->number] = 0;
+ tscm->on_sysex[substream->number] = 0;
+ return 0;
+}
+
+static int midi_capture_close(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_playback_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+
+ snd_fw_async_midi_port_finish(&tscm->out_ports[substream->number]);
+
+ return 0;
+}
+
+static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+ struct snd_tscm *tscm = substrm->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&tscm->lock, flags);
+
+ if (up)
+ tscm->tx_midi_substreams[substrm->number] = substrm;
+ else
+ tscm->tx_midi_substreams[substrm->number] = NULL;
+
+ spin_unlock_irqrestore(&tscm->lock, flags);
+}
+
+static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+ struct snd_tscm *tscm = substrm->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&tscm->lock, flags);
+
+ if (up)
+ snd_fw_async_midi_port_run(&tscm->out_ports[substrm->number],
+ substrm);
+
+ spin_unlock_irqrestore(&tscm->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_capture_ops = {
+ .open = midi_capture_open,
+ .close = midi_capture_close,
+ .trigger = midi_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_playback_ops = {
+ .open = midi_playback_open,
+ .close = midi_playback_close,
+ .trigger = midi_playback_trigger,
+};
+
+int snd_tscm_create_midi_devices(struct snd_tscm *tscm)
+{
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_str *stream;
+ struct snd_rawmidi_substream *subs;
+ int err;
+
+ err = snd_rawmidi_new(tscm->card, tscm->card->driver, 0,
+ tscm->spec->midi_playback_ports,
+ tscm->spec->midi_capture_ports,
+ &rmidi);
+ if (err < 0)
+ return err;
+
+ snprintf(rmidi->name, sizeof(rmidi->name),
+ "%s MIDI", tscm->card->shortname);
+ rmidi->private_data = tscm;
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &midi_capture_ops);
+ stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+
+ /* Set port names for MIDI input. */
+ list_for_each_entry(subs, &stream->substreams, list) {
+ /* TODO: support virtual MIDI ports. */
+ if (subs->number < tscm->spec->midi_capture_ports) {
+ /* Hardware MIDI ports. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ tscm->card->shortname, subs->number + 1);
+ }
+ }
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &midi_playback_ops);
+ stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+
+ /* Set port names for MIDI ourput. */
+ list_for_each_entry(subs, &stream->substreams, list) {
+ if (subs->number < tscm->spec->midi_playback_ports) {
+ /* Hardware MIDI ports only. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ tscm->card->shortname, subs->number + 1);
+ }
+ }
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
new file mode 100644
index 000000000000..380d3db969a5
--- /dev/null
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -0,0 +1,312 @@
+/*
+ * tascam-pcm.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+static void set_buffer_params(struct snd_pcm_hardware *hw)
+{
+ hw->period_bytes_min = 4 * hw->channels_min;
+ hw->period_bytes_max = hw->period_bytes_min * 2048;
+ hw->buffer_bytes_max = hw->period_bytes_max * 2;
+
+ hw->periods_min = 2;
+ hw->periods_max = UINT_MAX;
+}
+
+static int pcm_init_hw_params(struct snd_tscm *tscm,
+ struct snd_pcm_substream *substream)
+{
+ static const struct snd_pcm_hardware hardware = {
+ .info = SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_JOINT_DUPLEX |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 18,
+ };
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct amdtp_stream *stream;
+ unsigned int pcm_channels;
+
+ runtime->hw = hardware;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
+ stream = &tscm->tx_stream;
+ pcm_channels = tscm->spec->pcm_capture_analog_channels;
+ } else {
+ runtime->hw.formats =
+ SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32;
+ stream = &tscm->rx_stream;
+ pcm_channels = tscm->spec->pcm_playback_analog_channels;
+ }
+
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ runtime->hw.channels_min = runtime->hw.channels_max = pcm_channels;
+
+ set_buffer_params(&runtime->hw);
+
+ return amdtp_tscm_add_pcm_hw_constraints(stream, runtime);
+}
+
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ enum snd_tscm_clock clock;
+ unsigned int rate;
+ int err;
+
+ err = snd_tscm_stream_lock_try(tscm);
+ if (err < 0)
+ goto end;
+
+ err = pcm_init_hw_params(tscm, substream);
+ if (err < 0)
+ goto err_locked;
+
+ err = snd_tscm_stream_get_clock(tscm, &clock);
+ if (clock != SND_TSCM_CLOCK_INTERNAL ||
+ amdtp_stream_pcm_running(&tscm->rx_stream) ||
+ amdtp_stream_pcm_running(&tscm->tx_stream)) {
+ err = snd_tscm_stream_get_rate(tscm, &rate);
+ if (err < 0)
+ goto err_locked;
+ substream->runtime->hw.rate_min = rate;
+ substream->runtime->hw.rate_max = rate;
+ }
+
+ snd_pcm_set_sync(substream);
+end:
+ return err;
+err_locked:
+ snd_tscm_stream_lock_release(tscm);
+ return err;
+}
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ snd_tscm_stream_lock_release(tscm);
+
+ return 0;
+}
+
+static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&tscm->mutex);
+ tscm->substreams_counter++;
+ mutex_unlock(&tscm->mutex);
+ }
+
+ amdtp_tscm_set_pcm_format(&tscm->tx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&tscm->mutex);
+ tscm->substreams_counter++;
+ mutex_unlock(&tscm->mutex);
+ }
+
+ amdtp_tscm_set_pcm_format(&tscm->rx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ mutex_lock(&tscm->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ tscm->substreams_counter--;
+
+ snd_tscm_stream_stop_duplex(tscm);
+
+ mutex_unlock(&tscm->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ mutex_lock(&tscm->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ tscm->substreams_counter--;
+
+ snd_tscm_stream_stop_duplex(tscm);
+
+ mutex_unlock(&tscm->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&tscm->mutex);
+
+ err = snd_tscm_stream_start_duplex(tscm, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&tscm->tx_stream);
+
+ mutex_unlock(&tscm->mutex);
+
+ return err;
+}
+
+static int pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&tscm->mutex);
+
+ err = snd_tscm_stream_start_duplex(tscm, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&tscm->rx_stream);
+
+ mutex_unlock(&tscm->mutex);
+
+ return err;
+}
+
+static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&tscm->tx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&tscm->tx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&tscm->rx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&tscm->rx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_tscm *tscm = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&tscm->tx_stream);
+}
+
+static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_tscm *tscm = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&tscm->rx_stream);
+}
+
+static struct snd_pcm_ops pcm_capture_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_capture_hw_params,
+ .hw_free = pcm_capture_hw_free,
+ .prepare = pcm_capture_prepare,
+ .trigger = pcm_capture_trigger,
+ .pointer = pcm_capture_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops pcm_playback_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_playback_hw_params,
+ .hw_free = pcm_playback_hw_free,
+ .prepare = pcm_playback_prepare,
+ .trigger = pcm_playback_trigger,
+ .pointer = pcm_playback_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+};
+
+int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(tscm->card, tscm->card->driver, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = tscm;
+ snprintf(pcm->name, sizeof(pcm->name),
+ "%s PCM", tscm->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-proc.c b/sound/firewire/tascam/tascam-proc.c
new file mode 100644
index 000000000000..bfd4a4c06914
--- /dev/null
+++ b/sound/firewire/tascam/tascam-proc.c
@@ -0,0 +1,88 @@
+/*
+ * tascam-proc.h - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "./tascam.h"
+
+static void proc_read_firmware(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_tscm *tscm = entry->private_data;
+ __be32 data;
+ unsigned int reg, fpga, arm, hw;
+ int err;
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_REGISTER,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ reg = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_FPGA,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ fpga = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_ARM,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ arm = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_HW,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ hw = be32_to_cpu(data);
+
+ snd_iprintf(buffer, "Register: %d (0x%08x)\n", reg & 0xffff, reg);
+ snd_iprintf(buffer, "FPGA: %d (0x%08x)\n", fpga & 0xffff, fpga);
+ snd_iprintf(buffer, "ARM: %d (0x%08x)\n", arm & 0xffff, arm);
+ snd_iprintf(buffer, "Hardware: %d (0x%08x)\n", hw >> 16, hw);
+}
+
+static void add_node(struct snd_tscm *tscm, struct snd_info_entry *root,
+ const char *name,
+ void (*op)(struct snd_info_entry *e,
+ struct snd_info_buffer *b))
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_card_entry(tscm->card, name, root);
+ if (entry == NULL)
+ return;
+
+ snd_info_set_text_ops(entry, tscm, op);
+ if (snd_info_register(entry) < 0)
+ snd_info_free_entry(entry);
+}
+
+void snd_tscm_proc_init(struct snd_tscm *tscm)
+{
+ struct snd_info_entry *root;
+
+ /*
+ * All nodes are automatically removed at snd_card_disconnect(),
+ * by following to link list.
+ */
+ root = snd_info_create_card_entry(tscm->card, "firewire",
+ tscm->card->proc_root);
+ if (root == NULL)
+ return;
+ root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(root) < 0) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ add_node(tscm, root, "firmware", proc_read_firmware);
+}
diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
new file mode 100644
index 000000000000..0e6dd5c61f53
--- /dev/null
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -0,0 +1,496 @@
+/*
+ * tascam-stream.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/delay.h>
+#include "tascam.h"
+
+#define CALLBACK_TIMEOUT 500
+
+static int get_clock(struct snd_tscm *tscm, u32 *data)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+ &reg, sizeof(reg), 0);
+ if (err >= 0)
+ *data = be32_to_cpu(reg);
+
+ return err;
+}
+
+static int set_clock(struct snd_tscm *tscm, unsigned int rate,
+ enum snd_tscm_clock clock)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+ data &= 0x0000ffff;
+
+ if (rate > 0) {
+ data &= 0x000000ff;
+ /* Base rate. */
+ if ((rate % 44100) == 0) {
+ data |= 0x00000100;
+ /* Multiplier. */
+ if (rate / 44100 == 2)
+ data |= 0x00008000;
+ } else if ((rate % 48000) == 0) {
+ data |= 0x00000200;
+ /* Multiplier. */
+ if (rate / 48000 == 2)
+ data |= 0x00008000;
+ } else {
+ return -EAGAIN;
+ }
+ }
+
+ if (clock != INT_MAX) {
+ data &= 0x0000ff00;
+ data |= clock + 1;
+ }
+
+ reg = cpu_to_be32(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ if (data & 0x00008000)
+ reg = cpu_to_be32(0x0000001a);
+ else
+ reg = cpu_to_be32(0x0000000d);
+
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MULTIPLEX_MODE,
+ &reg, sizeof(reg), 0);
+}
+
+int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate)
+{
+ u32 data = 0x0;
+ unsigned int trials = 0;
+ int err;
+
+ while (data == 0x0 || trials++ < 5) {
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+
+ data = (data & 0xff000000) >> 24;
+ }
+
+ /* Check base rate. */
+ if ((data & 0x0f) == 0x01)
+ *rate = 44100;
+ else if ((data & 0x0f) == 0x02)
+ *rate = 48000;
+ else
+ return -EAGAIN;
+
+ /* Check multiplier. */
+ if ((data & 0xf0) == 0x80)
+ *rate *= 2;
+ else if ((data & 0xf0) != 0x00)
+ return -EAGAIN;
+
+ return err;
+}
+
+int snd_tscm_stream_get_clock(struct snd_tscm *tscm, enum snd_tscm_clock *clock)
+{
+ u32 data;
+ int err;
+
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+
+ *clock = ((data & 0x00ff0000) >> 16) - 1;
+ if (*clock < 0 || *clock > SND_TSCM_CLOCK_ADAT)
+ return -EIO;
+
+ return 0;
+}
+
+static int enable_data_channels(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ u32 data;
+ unsigned int i;
+ int err;
+
+ data = 0;
+ for (i = 0; i < tscm->spec->pcm_capture_analog_channels; ++i)
+ data |= BIT(i);
+ if (tscm->spec->has_adat)
+ data |= 0x0000ff00;
+ if (tscm->spec->has_spdif)
+ data |= 0x00030000;
+
+ reg = cpu_to_be32(data);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_TX_PCM_CHANNELS,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = 0;
+ for (i = 0; i < tscm->spec->pcm_playback_analog_channels; ++i)
+ data |= BIT(i);
+ if (tscm->spec->has_adat)
+ data |= 0x0000ff00;
+ if (tscm->spec->has_spdif)
+ data |= 0x00030000;
+
+ reg = cpu_to_be32(data);
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_RX_PCM_CHANNELS,
+ &reg, sizeof(reg), 0);
+}
+
+static int set_stream_formats(struct snd_tscm *tscm, unsigned int rate)
+{
+ __be32 reg;
+ int err;
+
+ /* Set an option for unknown purpose. */
+ reg = cpu_to_be32(0x00200000);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ err = enable_data_channels(tscm);
+ if (err < 0)
+ return err;
+
+ return set_clock(tscm, rate, INT_MAX);
+}
+
+static void finish_session(struct snd_tscm *tscm)
+{
+ __be32 reg;
+
+ reg = 0;
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
+ &reg, sizeof(reg), 0);
+
+ reg = 0;
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON,
+ &reg, sizeof(reg), 0);
+
+}
+
+static int begin_session(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ int err;
+
+ reg = cpu_to_be32(0x00000001);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ reg = cpu_to_be32(0x00000001);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Set an option for unknown purpose. */
+ reg = cpu_to_be32(0x00002000);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Start multiplexing PCM samples on packets. */
+ reg = cpu_to_be32(0x00000001);
+ return snd_fw_transaction(tscm->unit,
+ TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_ON,
+ &reg, sizeof(reg), 0);
+}
+
+static void release_resources(struct snd_tscm *tscm)
+{
+ __be32 reg;
+
+ /* Unregister channels. */
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH,
+ &reg, sizeof(reg), 0);
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN,
+ &reg, sizeof(reg), 0);
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH,
+ &reg, sizeof(reg), 0);
+
+ /* Release isochronous resources. */
+ fw_iso_resources_free(&tscm->tx_resources);
+ fw_iso_resources_free(&tscm->rx_resources);
+}
+
+static int keep_resources(struct snd_tscm *tscm, unsigned int rate)
+{
+ __be32 reg;
+ int err;
+
+ /* Keep resources for in-stream. */
+ err = amdtp_tscm_set_parameters(&tscm->tx_stream, rate);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&tscm->tx_resources,
+ amdtp_stream_get_max_payload(&tscm->tx_stream),
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ /* Keep resources for out-stream. */
+ err = amdtp_tscm_set_parameters(&tscm->rx_stream, rate);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&tscm->rx_resources,
+ amdtp_stream_get_max_payload(&tscm->rx_stream),
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ return err;
+
+ /* Register the isochronous channel for transmitting stream. */
+ reg = cpu_to_be32(tscm->tx_resources.channel);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ /* Unknown */
+ reg = cpu_to_be32(0x00000002);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ /* Register the isochronous channel for receiving stream. */
+ reg = cpu_to_be32(tscm->rx_resources.channel);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ release_resources(tscm);
+ return err;
+}
+
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm)
+{
+ unsigned int pcm_channels;
+ int err;
+
+ /* For out-stream. */
+ err = fw_iso_resources_init(&tscm->rx_resources, tscm->unit);
+ if (err < 0)
+ return err;
+ pcm_channels = tscm->spec->pcm_playback_analog_channels;
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ err = amdtp_tscm_init(&tscm->rx_stream, tscm->unit, AMDTP_OUT_STREAM,
+ pcm_channels);
+ if (err < 0)
+ return err;
+
+ /* For in-stream. */
+ err = fw_iso_resources_init(&tscm->tx_resources, tscm->unit);
+ if (err < 0)
+ return err;
+ pcm_channels = tscm->spec->pcm_capture_analog_channels;
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ err = amdtp_tscm_init(&tscm->tx_stream, tscm->unit, AMDTP_IN_STREAM,
+ pcm_channels);
+ if (err < 0)
+ amdtp_stream_destroy(&tscm->rx_stream);
+
+ return 0;
+}
+
+/* At bus reset, streaming is stopped and some registers are clear. */
+void snd_tscm_stream_update_duplex(struct snd_tscm *tscm)
+{
+ amdtp_stream_pcm_abort(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->tx_stream);
+
+ amdtp_stream_pcm_abort(&tscm->rx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+}
+
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
+void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm)
+{
+ amdtp_stream_destroy(&tscm->rx_stream);
+ amdtp_stream_destroy(&tscm->tx_stream);
+
+ fw_iso_resources_destroy(&tscm->rx_resources);
+ fw_iso_resources_destroy(&tscm->tx_resources);
+}
+
+int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
+{
+ unsigned int curr_rate;
+ int err;
+
+ if (tscm->substreams_counter == 0)
+ return 0;
+
+ err = snd_tscm_stream_get_rate(tscm, &curr_rate);
+ if (err < 0)
+ return err;
+ if (curr_rate != rate ||
+ amdtp_streaming_error(&tscm->tx_stream) ||
+ amdtp_streaming_error(&tscm->rx_stream)) {
+ finish_session(tscm);
+
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ release_resources(tscm);
+ }
+
+ if (!amdtp_stream_running(&tscm->tx_stream)) {
+ amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE,
+ &tscm->tx_stream, &tscm->rx_stream);
+ err = keep_resources(tscm, rate);
+ if (err < 0)
+ goto error;
+
+ err = set_stream_formats(tscm, rate);
+ if (err < 0)
+ goto error;
+
+ err = begin_session(tscm);
+ if (err < 0)
+ goto error;
+
+ err = amdtp_stream_start(&tscm->tx_stream,
+ tscm->tx_resources.channel,
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&tscm->tx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ if (!amdtp_stream_running(&tscm->rx_stream)) {
+ err = amdtp_stream_start(&tscm->rx_stream,
+ tscm->rx_resources.channel,
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&tscm->rx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ return 0;
+error:
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ finish_session(tscm);
+ release_resources(tscm);
+
+ return err;
+}
+
+void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm)
+{
+ if (tscm->substreams_counter > 0)
+ return;
+
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ finish_session(tscm);
+ release_resources(tscm);
+}
+
+void snd_tscm_stream_lock_changed(struct snd_tscm *tscm)
+{
+ tscm->dev_lock_changed = true;
+ wake_up(&tscm->hwdep_wait);
+}
+
+int snd_tscm_stream_lock_try(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ /* user land lock this */
+ if (tscm->dev_lock_count < 0) {
+ err = -EBUSY;
+ goto end;
+ }
+
+ /* this is the first time */
+ if (tscm->dev_lock_count++ == 0)
+ snd_tscm_stream_lock_changed(tscm);
+ err = 0;
+end:
+ spin_unlock_irq(&tscm->lock);
+ return err;
+}
+
+void snd_tscm_stream_lock_release(struct snd_tscm *tscm)
+{
+ spin_lock_irq(&tscm->lock);
+
+ if (WARN_ON(tscm->dev_lock_count <= 0))
+ goto end;
+ if (--tscm->dev_lock_count == 0)
+ snd_tscm_stream_lock_changed(tscm);
+end:
+ spin_unlock_irq(&tscm->lock);
+}
diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c
new file mode 100644
index 000000000000..1c9a88be55c8
--- /dev/null
+++ b/sound/firewire/tascam/tascam-transaction.c
@@ -0,0 +1,293 @@
+/*
+ * tascam-transaction.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+/*
+ * When return minus value, given argument is not MIDI status.
+ * When return 0, given argument is a beginning of system exclusive.
+ * When return the others, given argument is MIDI data.
+ */
+static inline int calculate_message_bytes(u8 status)
+{
+ switch (status) {
+ case 0xf6: /* Tune request. */
+ case 0xf8: /* Timing clock. */
+ case 0xfa: /* Start. */
+ case 0xfb: /* Continue. */
+ case 0xfc: /* Stop. */
+ case 0xfe: /* Active sensing. */
+ case 0xff: /* System reset. */
+ return 1;
+ case 0xf1: /* MIDI time code quarter frame. */
+ case 0xf3: /* Song select. */
+ return 2;
+ case 0xf2: /* Song position pointer. */
+ return 3;
+ case 0xf0: /* Exclusive. */
+ return 0;
+ case 0xf7: /* End of exclusive. */
+ break;
+ case 0xf4: /* Undefined. */
+ case 0xf5: /* Undefined. */
+ case 0xf9: /* Undefined. */
+ case 0xfd: /* Undefined. */
+ break;
+ default:
+ switch (status & 0xf0) {
+ case 0x80: /* Note on. */
+ case 0x90: /* Note off. */
+ case 0xa0: /* Polyphonic key pressure. */
+ case 0xb0: /* Control change and Mode change. */
+ case 0xe0: /* Pitch bend change. */
+ return 3;
+ case 0xc0: /* Program change. */
+ case 0xd0: /* Channel pressure. */
+ return 2;
+ default:
+ break;
+ }
+ break;
+ }
+
+ return -EINVAL;
+}
+
+static int fill_message(struct snd_rawmidi_substream *substream, u8 *buf)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+ unsigned int port = substream->number;
+ unsigned int len;
+ unsigned int i;
+ u8 status;
+ int consume;
+
+ buf[0] = buf[1] = buf[2] = buf[3] = 0x00;
+
+ len = snd_rawmidi_transmit_peek(substream, buf + 1, 3);
+ if (len == 0)
+ return 0;
+
+ /* On exclusive message. */
+ if (tscm->on_sysex[port]) {
+ /* Seek the end of exclusives. */
+ for (i = 1; i < 4 || i < len; ++i) {
+ if (buf[i] == 0xf7) {
+ tscm->on_sysex[port] = false;
+ break;
+ }
+ }
+
+ /* At the end of exclusive message, use label 0x07. */
+ if (!tscm->on_sysex[port]) {
+ len = i;
+ buf[0] = (port << 4) | 0x07;
+ /* During exclusive message, use label 0x04. */
+ } else if (len == 3) {
+ buf[0] = (port << 4) | 0x04;
+ /* We need to fill whole 3 bytes. Go to next change. */
+ } else {
+ len = 0;
+ }
+ } else {
+ /* The beginning of exclusives. */
+ if (buf[1] == 0xf0) {
+ /* Transfer it in next chance in another condition. */
+ tscm->on_sysex[port] = true;
+ return 0;
+ } else {
+ /* On running-status. */
+ if ((buf[1] & 0x80) != 0x80)
+ status = tscm->running_status[port];
+ else
+ status = buf[1];
+
+ /* Calculate consume bytes. */
+ consume = calculate_message_bytes(status);
+ if (consume <= 0)
+ return 0;
+
+ /* On running-status. */
+ if ((buf[1] & 0x80) != 0x80) {
+ buf[3] = buf[2];
+ buf[2] = buf[1];
+ buf[1] = tscm->running_status[port];
+ consume--;
+ } else {
+ tscm->running_status[port] = buf[1];
+ }
+
+ /* Confirm length. */
+ if (len < consume)
+ return 0;
+ if (len > consume)
+ len = consume;
+ }
+
+ buf[0] = (port << 4) | (buf[1] >> 4);
+ }
+
+ return len;
+}
+
+static void handle_midi_tx(struct fw_card *card, struct fw_request *request,
+ int tcode, int destination, int source,
+ int generation, unsigned long long offset,
+ void *data, size_t length, void *callback_data)
+{
+ struct snd_tscm *tscm = callback_data;
+ u32 *buf = (u32 *)data;
+ unsigned int messages;
+ unsigned int i;
+ unsigned int port;
+ struct snd_rawmidi_substream *substream;
+ u8 *b;
+ int bytes;
+
+ if (offset != tscm->async_handler.offset)
+ goto end;
+
+ messages = length / 8;
+ for (i = 0; i < messages; i++) {
+ b = (u8 *)(buf + i * 2);
+
+ port = b[0] >> 4;
+ /* TODO: support virtual MIDI ports. */
+ if (port > tscm->spec->midi_capture_ports)
+ goto end;
+
+ /* Assume the message length. */
+ bytes = calculate_message_bytes(b[1]);
+ /* On MIDI data or exclusives. */
+ if (bytes <= 0) {
+ /* Seek the end of exclusives. */
+ for (bytes = 1; bytes < 4; bytes++) {
+ if (b[bytes] == 0xf7)
+ break;
+ }
+ if (bytes == 4)
+ bytes = 3;
+ }
+
+ substream = ACCESS_ONCE(tscm->tx_midi_substreams[port]);
+ if (substream != NULL)
+ snd_rawmidi_receive(substream, b + 1, bytes);
+ }
+end:
+ fw_send_response(card, request, RCODE_COMPLETE);
+}
+
+int snd_tscm_transaction_register(struct snd_tscm *tscm)
+{
+ static const struct fw_address_region resp_register_region = {
+ .start = 0xffffe0000000ull,
+ .end = 0xffffe000ffffull,
+ };
+ unsigned int i;
+ int err;
+
+ /*
+ * Usually, two quadlets are transferred by one transaction. The first
+ * quadlet has MIDI messages, the rest includes timestamp.
+ * Sometimes, 8 set of the data is transferred by a block transaction.
+ */
+ tscm->async_handler.length = 8 * 8;
+ tscm->async_handler.address_callback = handle_midi_tx;
+ tscm->async_handler.callback_data = tscm;
+
+ err = fw_core_add_address_handler(&tscm->async_handler,
+ &resp_register_region);
+ if (err < 0)
+ return err;
+
+ err = snd_tscm_transaction_reregister(tscm);
+ if (err < 0)
+ goto error;
+
+ for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++) {
+ err = snd_fw_async_midi_port_init(
+ &tscm->out_ports[i], tscm->unit,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_RX_QUAD,
+ 4, fill_message);
+ if (err < 0)
+ goto error;
+ }
+
+ return err;
+error:
+ fw_core_remove_address_handler(&tscm->async_handler);
+ return err;
+}
+
+/* At bus reset, these registers are cleared. */
+int snd_tscm_transaction_reregister(struct snd_tscm *tscm)
+{
+ struct fw_device *device = fw_parent_device(tscm->unit);
+ __be32 reg;
+ int err;
+
+ /* Register messaging address. Block transaction is not allowed. */
+ reg = cpu_to_be32((device->card->node_id << 16) |
+ (tscm->async_handler.offset >> 32));
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ reg = cpu_to_be32(tscm->async_handler.offset);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Turn on messaging. */
+ reg = cpu_to_be32(0x00000001);
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Turn on FireWire LED. */
+ reg = cpu_to_be32(0x0001008e);
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER,
+ &reg, sizeof(reg), 0);
+}
+
+void snd_tscm_transaction_unregister(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ unsigned int i;
+
+ /* Turn off FireWire LED. */
+ reg = cpu_to_be32(0x0000008e);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER,
+ &reg, sizeof(reg), 0);
+
+ /* Turn off messaging. */
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON,
+ &reg, sizeof(reg), 0);
+
+ /* Unregister the address. */
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI,
+ &reg, sizeof(reg), 0);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO,
+ &reg, sizeof(reg), 0);
+
+ fw_core_remove_address_handler(&tscm->async_handler);
+ for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++)
+ snd_fw_async_midi_port_destroy(&tscm->out_ports[i]);
+}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
new file mode 100644
index 000000000000..c6747a45795b
--- /dev/null
+++ b/sound/firewire/tascam/tascam.c
@@ -0,0 +1,210 @@
+/*
+ * tascam.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+MODULE_DESCRIPTION("TASCAM FireWire series Driver");
+MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>");
+MODULE_LICENSE("GPL v2");
+
+static struct snd_tscm_spec model_specs[] = {
+ {
+ .name = "FW-1884",
+ .has_adat = true,
+ .has_spdif = true,
+ .pcm_capture_analog_channels = 8,
+ .pcm_playback_analog_channels = 8,
+ .midi_capture_ports = 4,
+ .midi_playback_ports = 4,
+ .is_controller = true,
+ },
+ {
+ .name = "FW-1082",
+ .has_adat = false,
+ .has_spdif = true,
+ .pcm_capture_analog_channels = 8,
+ .pcm_playback_analog_channels = 2,
+ .midi_capture_ports = 2,
+ .midi_playback_ports = 2,
+ .is_controller = true,
+ },
+ /* FW-1804 may be supported. */
+};
+
+static int identify_model(struct snd_tscm *tscm)
+{
+ struct fw_device *fw_dev = fw_parent_device(tscm->unit);
+ const u32 *config_rom = fw_dev->config_rom;
+ char model[8];
+ unsigned int i;
+ u8 c;
+
+ if (fw_dev->config_rom_length < 30) {
+ dev_err(&tscm->unit->device,
+ "Configuration ROM is too short.\n");
+ return -ENODEV;
+ }
+
+ /* Pick up model name from certain addresses. */
+ for (i = 0; i < 8; i++) {
+ c = config_rom[28 + i / 4] >> (24 - 8 * (i % 4));
+ if (c == '\0')
+ break;
+ model[i] = c;
+ }
+ model[i] = '\0';
+
+ for (i = 0; i < ARRAY_SIZE(model_specs); i++) {
+ if (strcmp(model, model_specs[i].name) == 0) {
+ tscm->spec = &model_specs[i];
+ break;
+ }
+ }
+ if (tscm->spec == NULL)
+ return -ENODEV;
+
+ strcpy(tscm->card->driver, "FW-TASCAM");
+ strcpy(tscm->card->shortname, model);
+ strcpy(tscm->card->mixername, model);
+ snprintf(tscm->card->longname, sizeof(tscm->card->longname),
+ "TASCAM %s, GUID %08x%08x at %s, S%d", model,
+ cpu_to_be32(fw_dev->config_rom[3]),
+ cpu_to_be32(fw_dev->config_rom[4]),
+ dev_name(&tscm->unit->device), 100 << fw_dev->max_speed);
+
+ return 0;
+}
+
+static void tscm_card_free(struct snd_card *card)
+{
+ struct snd_tscm *tscm = card->private_data;
+
+ snd_tscm_transaction_unregister(tscm);
+ snd_tscm_stream_destroy_duplex(tscm);
+
+ fw_unit_put(tscm->unit);
+
+ mutex_destroy(&tscm->mutex);
+}
+
+static int snd_tscm_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_card *card;
+ struct snd_tscm *tscm;
+ int err;
+
+ /* create card */
+ err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
+ sizeof(struct snd_tscm), &card);
+ if (err < 0)
+ return err;
+ card->private_free = tscm_card_free;
+
+ /* initialize myself */
+ tscm = card->private_data;
+ tscm->card = card;
+ tscm->unit = fw_unit_get(unit);
+
+ mutex_init(&tscm->mutex);
+ spin_lock_init(&tscm->lock);
+ init_waitqueue_head(&tscm->hwdep_wait);
+
+ err = identify_model(tscm);
+ if (err < 0)
+ goto error;
+
+ snd_tscm_proc_init(tscm);
+
+ err = snd_tscm_stream_init_duplex(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_pcm_devices(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_transaction_register(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_midi_devices(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_hwdep_device(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ dev_set_drvdata(&unit->device, tscm);
+
+ return err;
+error:
+ snd_card_free(card);
+ return err;
+}
+
+static void snd_tscm_update(struct fw_unit *unit)
+{
+ struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+
+ snd_tscm_transaction_reregister(tscm);
+
+ mutex_lock(&tscm->mutex);
+ snd_tscm_stream_update_duplex(tscm);
+ mutex_unlock(&tscm->mutex);
+}
+
+static void snd_tscm_remove(struct fw_unit *unit)
+{
+ struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(tscm->card);
+}
+
+static const struct ieee1394_device_id snd_tscm_id_table[] = {
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ },
+ /* FE-08 requires reverse-engineering because it just has faders. */
+ {}
+};
+MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table);
+
+static struct fw_driver tscm_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "snd-firewire-tascam",
+ .bus = &fw_bus_type,
+ },
+ .probe = snd_tscm_probe,
+ .update = snd_tscm_update,
+ .remove = snd_tscm_remove,
+ .id_table = snd_tscm_id_table,
+};
+
+static int __init snd_tscm_init(void)
+{
+ return driver_register(&tscm_driver.driver);
+}
+
+static void __exit snd_tscm_exit(void)
+{
+ driver_unregister(&tscm_driver.driver);
+}
+
+module_init(snd_tscm_init);
+module_exit(snd_tscm_exit);
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
new file mode 100644
index 000000000000..2d028d2bd3bd
--- /dev/null
+++ b/sound/firewire/tascam/tascam.h
@@ -0,0 +1,147 @@
+/*
+ * tascam.h - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_TASCAM_H_INCLUDED
+#define SOUND_TASCAM_H_INCLUDED
+
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/mutex.h>
+#include <linux/slab.h>
+#include <linux/compat.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+
+#include "../lib.h"
+#include "../amdtp-stream.h"
+#include "../iso-resources.h"
+
+struct snd_tscm_spec {
+ const char *const name;
+ bool has_adat;
+ bool has_spdif;
+ unsigned int pcm_capture_analog_channels;
+ unsigned int pcm_playback_analog_channels;
+ unsigned int midi_capture_ports;
+ unsigned int midi_playback_ports;
+ bool is_controller;
+};
+
+#define TSCM_MIDI_IN_PORT_MAX 4
+#define TSCM_MIDI_OUT_PORT_MAX 4
+
+struct snd_tscm {
+ struct snd_card *card;
+ struct fw_unit *unit;
+
+ struct mutex mutex;
+ spinlock_t lock;
+
+ const struct snd_tscm_spec *spec;
+
+ struct fw_iso_resources tx_resources;
+ struct fw_iso_resources rx_resources;
+ struct amdtp_stream tx_stream;
+ struct amdtp_stream rx_stream;
+ unsigned int substreams_counter;
+
+ int dev_lock_count;
+ bool dev_lock_changed;
+ wait_queue_head_t hwdep_wait;
+
+ /* For MIDI message incoming transactions. */
+ struct fw_address_handler async_handler;
+ struct snd_rawmidi_substream *tx_midi_substreams[TSCM_MIDI_IN_PORT_MAX];
+
+ /* For MIDI message outgoing transactions. */
+ struct snd_fw_async_midi_port out_ports[TSCM_MIDI_OUT_PORT_MAX];
+ u8 running_status[TSCM_MIDI_OUT_PORT_MAX];
+ bool on_sysex[TSCM_MIDI_OUT_PORT_MAX];
+
+ /* For control messages. */
+ struct snd_firewire_tascam_status *status;
+};
+
+#define TSCM_ADDR_BASE 0xffff00000000ull
+
+#define TSCM_OFFSET_FIRMWARE_REGISTER 0x0000
+#define TSCM_OFFSET_FIRMWARE_FPGA 0x0004
+#define TSCM_OFFSET_FIRMWARE_ARM 0x0008
+#define TSCM_OFFSET_FIRMWARE_HW 0x000c
+
+#define TSCM_OFFSET_ISOC_TX_CH 0x0200
+#define TSCM_OFFSET_UNKNOWN 0x0204
+#define TSCM_OFFSET_START_STREAMING 0x0208
+#define TSCM_OFFSET_ISOC_RX_CH 0x020c
+#define TSCM_OFFSET_ISOC_RX_ON 0x0210 /* Little conviction. */
+#define TSCM_OFFSET_TX_PCM_CHANNELS 0x0214
+#define TSCM_OFFSET_RX_PCM_CHANNELS 0x0218
+#define TSCM_OFFSET_MULTIPLEX_MODE 0x021c
+#define TSCM_OFFSET_ISOC_TX_ON 0x0220
+/* Unknown 0x0224 */
+#define TSCM_OFFSET_CLOCK_STATUS 0x0228
+#define TSCM_OFFSET_SET_OPTION 0x022c
+
+#define TSCM_OFFSET_MIDI_TX_ON 0x0300
+#define TSCM_OFFSET_MIDI_TX_ADDR_HI 0x0304
+#define TSCM_OFFSET_MIDI_TX_ADDR_LO 0x0308
+
+#define TSCM_OFFSET_LED_POWER 0x0404
+
+#define TSCM_OFFSET_MIDI_RX_QUAD 0x4000
+
+enum snd_tscm_clock {
+ SND_TSCM_CLOCK_INTERNAL = 0,
+ SND_TSCM_CLOCK_WORD = 1,
+ SND_TSCM_CLOCK_SPDIF = 2,
+ SND_TSCM_CLOCK_ADAT = 3,
+};
+
+int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, unsigned int pcm_channels);
+int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate);
+int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
+
+int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate);
+int snd_tscm_stream_get_clock(struct snd_tscm *tscm,
+ enum snd_tscm_clock *clock);
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm);
+void snd_tscm_stream_update_duplex(struct snd_tscm *tscm);
+void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm);
+int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate);
+void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm);
+
+void snd_tscm_stream_lock_changed(struct snd_tscm *tscm);
+int snd_tscm_stream_lock_try(struct snd_tscm *tscm);
+void snd_tscm_stream_lock_release(struct snd_tscm *tscm);
+
+int snd_tscm_transaction_register(struct snd_tscm *tscm);
+int snd_tscm_transaction_reregister(struct snd_tscm *tscm);
+void snd_tscm_transaction_unregister(struct snd_tscm *tscm);
+
+void snd_tscm_proc_init(struct snd_tscm *tscm);
+
+int snd_tscm_create_pcm_devices(struct snd_tscm *tscm);
+
+int snd_tscm_create_midi_devices(struct snd_tscm *tscm);
+
+int snd_tscm_create_hwdep_device(struct snd_tscm *tscm);
+
+#endif
diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c
index 33ba77dd32f2..cb89ec7c8147 100644
--- a/sound/hda/ext/hdac_ext_stream.c
+++ b/sound/hda/ext/hdac_ext_stream.c
@@ -227,7 +227,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_setup);
void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
int stream)
{
- snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 0);
+ snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 1 << stream);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_link_set_stream_id);
@@ -385,14 +385,13 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type)
break;
case HDAC_EXT_STREAM_TYPE_HOST:
- if (stream->decoupled) {
+ if (stream->decoupled && !stream->link_locked)
snd_hdac_ext_stream_decouple(ebus, stream, false);
- snd_hdac_stream_release(&stream->hstream);
- }
+ snd_hdac_stream_release(&stream->hstream);
break;
case HDAC_EXT_STREAM_TYPE_LINK:
- if (stream->decoupled)
+ if (stream->decoupled && !stream->hstream.opened)
snd_hdac_ext_stream_decouple(ebus, stream, false);
spin_lock_irq(&bus->reg_lock);
stream->link_locked = 0;
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index db96042a497f..b3b0ad289df1 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -952,3 +952,84 @@ bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid,
return true;
}
EXPORT_SYMBOL_GPL(snd_hdac_is_supported_format);
+
+static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm);
+ unsigned int res;
+
+ if (snd_hdac_exec_verb(hdac, cmd, flags, &res))
+ return -1;
+
+ return res;
+}
+
+static int codec_write(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm);
+
+ return snd_hdac_exec_verb(hdac, cmd, flags, NULL);
+}
+
+/**
+ * snd_hdac_codec_read - send a command and get the response
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @flags: optional bit flags
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command and read the corresponding response.
+ *
+ * Returns the obtained response value, or -1 for an error.
+ */
+int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ return codec_read(hdac, nid, flags, verb, parm);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_read);
+
+/**
+ * snd_hdac_codec_write - send a single command without waiting for response
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @flags: optional bit flags
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ return codec_write(hdac, nid, flags, verb, parm);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_write);
+
+/*
+ * snd_hdac_check_power_state: check whether the actual power state matches
+ * with the target state
+ *
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @target_state: target state to check for
+ *
+ * Return true if state matches, false if not
+ */
+bool snd_hdac_check_power_state(struct hdac_device *hdac,
+ hda_nid_t nid, unsigned int target_state)
+{
+ unsigned int state = codec_read(hdac, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+
+ if (state & AC_PWRST_ERROR)
+ return true;
+ state = (state >> 4) & 0x0f;
+ return (state == target_state);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_check_power_state);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 37f43a1b34ef..2eeaf5ea20f9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -91,50 +91,6 @@ static int codec_exec_verb(struct hdac_device *dev, unsigned int cmd,
}
/**
- * snd_hda_codec_read - send a command and get the response
- * @codec: the HDA codec
- * @nid: NID to send the command
- * @flags: optional bit flags
- * @verb: the verb to send
- * @parm: the parameter for the verb
- *
- * Send a single command and read the corresponding response.
- *
- * Returns the obtained response value, or -1 for an error.
- */
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
- int flags,
- unsigned int verb, unsigned int parm)
-{
- unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm);
- unsigned int res;
- if (snd_hdac_exec_verb(&codec->core, cmd, flags, &res))
- return -1;
- return res;
-}
-EXPORT_SYMBOL_GPL(snd_hda_codec_read);
-
-/**
- * snd_hda_codec_write - send a single command without waiting for response
- * @codec: the HDA codec
- * @nid: NID to send the command
- * @flags: optional bit flags
- * @verb: the verb to send
- * @parm: the parameter for the verb
- *
- * Send a single command without waiting for response.
- *
- * Returns 0 if successful, or a negative error code.
- */
-int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
- unsigned int verb, unsigned int parm)
-{
- unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm);
- return snd_hdac_exec_verb(&codec->core, cmd, flags, NULL);
-}
-EXPORT_SYMBOL_GPL(snd_hda_codec_write);
-
-/**
* snd_hda_sequence_write - sequence writes
* @codec: the HDA codec
* @seq: VERB array to send
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2970413f18a0..95991e463abb 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -309,11 +309,21 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec);
/*
* low level functions
*/
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
+static inline unsigned int
+snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
int flags,
- unsigned int verb, unsigned int parm);
-int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
- unsigned int verb, unsigned int parm);
+ unsigned int verb, unsigned int parm)
+{
+ return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm);
+}
+
+static inline int
+snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
+ unsigned int verb, unsigned int parm)
+{
+ return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm);
+}
+
#define snd_hda_param_read(codec, nid, param) \
snd_hdac_read_parm(&(codec)->core, nid, param)
#define snd_hda_get_sub_nodes(codec, nid, start_nid) \
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 4a21c2199e02..d0e066e4c985 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -681,12 +681,7 @@ static inline bool
snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid,
unsigned int target_state)
{
- unsigned int state = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_POWER_STATE, 0);
- if (state & AC_PWRST_ERROR)
- return true;
- state = (state >> 4) & 0x0f;
- return (state == target_state);
+ return snd_hdac_check_power_state(&codec->core, nid, target_state);
}
unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index acbfbe087ee8..3a2d4a5a1714 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1775,6 +1775,16 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
return non_pcm;
}
+/* There is a fixed mapping between audio pin node and display port
+ * on current Intel platforms:
+ * Pin Widget 5 - PORT B (port = 1 in i915 driver)
+ * Pin Widget 6 - PORT C (port = 2 in i915 driver)
+ * Pin Widget 7 - PORT D (port = 3 in i915 driver)
+ */
+static int intel_pin2port(hda_nid_t pin_nid)
+{
+ return pin_nid - 4;
+}
/*
* HDMI callbacks
@@ -1791,6 +1801,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
int pin_idx = hinfo_to_pin_index(codec, hinfo);
struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
hda_nid_t pin_nid = per_pin->pin_nid;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct i915_audio_component *acomp = codec->bus->core.audio_component;
bool non_pcm;
int pinctl;
@@ -1807,6 +1819,13 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx);
}
+ /* Call sync_audio_rate to set the N/CTS/M manually if necessary */
+ /* Todo: add DP1.2 MST audio support later */
+ if (acomp && acomp->ops && acomp->ops->sync_audio_rate)
+ acomp->ops->sync_audio_rate(acomp->dev,
+ intel_pin2port(pin_nid),
+ runtime->rate);
+
non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
mutex_lock(&per_pin->lock);
per_pin->channels = substream->runtime->channels;
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 9bba275b4c9b..2875b4f6d8c9 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5112,6 +5112,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
dev_err(hdsp->card->dev,
"too short firmware size %d (expected %d)\n",
(int)fw->size, HDSP_FIRMWARE_SIZE);
+ release_firmware(fw);
return -EINVAL;
}
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 225bfda414e9..7de792b06007 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -58,6 +58,7 @@ source "sound/soc/sh/Kconfig"
source "sound/soc/sirf/Kconfig"
source "sound/soc/spear/Kconfig"
source "sound/soc/sti/Kconfig"
+source "sound/soc/sunxi/Kconfig"
source "sound/soc/tegra/Kconfig"
source "sound/soc/txx9/Kconfig"
source "sound/soc/ux500/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 134aca150a50..af0a5714e107 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -40,6 +40,7 @@ obj-$(CONFIG_SND_SOC) += sh/
obj-$(CONFIG_SND_SOC) += sirf/
obj-$(CONFIG_SND_SOC) += spear/
obj-$(CONFIG_SND_SOC) += sti/
+obj-$(CONFIG_SND_SOC) += sunxi/
obj-$(CONFIG_SND_SOC) += tegra/
obj-$(CONFIG_SND_SOC) += txx9/
obj-$(CONFIG_SND_SOC) += ux500/
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index aa354e1c6ff7..1933bcd46cca 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -176,6 +176,7 @@ static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = {
{ .compatible = "atmel,asoc-wm8904", },
{ }
};
+MODULE_DEVICE_TABLE(of, atmel_asoc_wm8904_dt_ids);
#endif
static struct platform_driver atmel_asoc_wm8904_driver = {
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
index 452f404abfd2..e97c32798e98 100644
--- a/sound/soc/au1x/db1000.c
+++ b/sound/soc/au1x/db1000.c
@@ -38,14 +38,7 @@ static int db1000_audio_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &db1000_ac97;
card->dev = &pdev->dev;
- return snd_soc_register_card(card);
-}
-
-static int db1000_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- snd_soc_unregister_card(card);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
static struct platform_driver db1000_audio_driver = {
@@ -54,7 +47,6 @@ static struct platform_driver db1000_audio_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = db1000_audio_probe,
- .remove = db1000_audio_remove,
};
module_platform_driver(db1000_audio_driver);
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 8c907ebea189..5c73061d912a 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -178,14 +178,7 @@ static int db1200_audio_probe(struct platform_device *pdev)
card = db1200_cards[pid->driver_data];
card->dev = &pdev->dev;
- return snd_soc_register_card(card);
-}
-
-static int db1200_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- snd_soc_unregister_card(card);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
static struct platform_driver db1200_audio_driver = {
@@ -195,7 +188,6 @@ static struct platform_driver db1200_audio_driver = {
},
.id_table = db1200_pids,
.probe = db1200_audio_probe,
- .remove = db1200_audio_remove,
};
module_platform_driver(db1200_audio_driver);
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index 5bf1501e5e3c..864df2616e10 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -87,27 +87,18 @@ static int bf5xx_ad1836_driver_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "Failed to register card\n");
return ret;
}
-static int bf5xx_ad1836_driver_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver bf5xx_ad1836_driver = {
.driver = {
.name = "bfin-snd-ad1836",
.pm = &snd_soc_pm_ops,
},
.probe = bf5xx_ad1836_driver_probe,
- .remove = bf5xx_ad1836_driver_remove,
};
module_platform_driver(bf5xx_ad1836_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
index 523baf5820d7..72ac78988426 100644
--- a/sound/soc/blackfin/bfin-eval-adau1373.c
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -154,16 +154,7 @@ static int bfin_eval_adau1373_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1373);
-}
-
-static int bfin_eval_adau1373_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373);
}
static struct platform_driver bfin_eval_adau1373_driver = {
@@ -172,7 +163,6 @@ static struct platform_driver bfin_eval_adau1373_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1373_probe,
- .remove = bfin_eval_adau1373_remove,
};
module_platform_driver(bfin_eval_adau1373_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
index f9e926dfd4ef..5c67f72cf9a9 100644
--- a/sound/soc/blackfin/bfin-eval-adau1701.c
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -94,16 +94,7 @@ static int bfin_eval_adau1701_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1701);
-}
-
-static int bfin_eval_adau1701_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701);
}
static struct platform_driver bfin_eval_adau1701_driver = {
@@ -112,7 +103,6 @@ static struct platform_driver bfin_eval_adau1701_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1701_probe,
- .remove = bfin_eval_adau1701_remove,
};
module_platform_driver(bfin_eval_adau1701_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
index 27eee66afdb2..1037477d10b2 100644
--- a/sound/soc/blackfin/bfin-eval-adav80x.c
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -119,16 +119,7 @@ static int bfin_eval_adav80x_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adav80x);
-}
-
-static int bfin_eval_adav80x_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x);
}
static const struct platform_device_id bfin_eval_adav80x_ids[] = {
@@ -144,7 +135,6 @@ static struct platform_driver bfin_eval_adav80x_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adav80x_probe,
- .remove = bfin_eval_adav80x_remove,
.id_table = bfin_eval_adav80x_ids,
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0c9733ecd17f..70e5a75901aa 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
select SND_SOC_AK4554
+ select SND_SOC_AK4613 if I2C
select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
@@ -79,7 +80,6 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX9877 if I2C
select SND_SOC_MC13783 if MFD_MC13XXX
select SND_SOC_ML26124 if I2C
- select SND_SOC_HDMI_CODEC
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM1792A if SPI_MASTER
select SND_SOC_PCM3008
@@ -319,6 +319,10 @@ config SND_SOC_AK4535
config SND_SOC_AK4554
tristate "AKM AK4554 CODEC"
+config SND_SOC_AK4613
+ tristate "AKM AK4613 CODEC"
+ depends on I2C
+
config SND_SOC_AK4641
tristate
@@ -442,9 +446,6 @@ config SND_SOC_BT_SCO
config SND_SOC_DMIC
tristate
-config SND_SOC_HDMI_CODEC
- tristate "HDMI stub CODEC"
-
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4a32077954ae..be1491acb6ae 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4554-objs := ak4554.o
+snd-soc-ak4613-objs := ak4613.o
snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
@@ -72,7 +73,6 @@ snd-soc-max98925-objs := max98925.o
snd-soc-max9850-objs := max9850.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
-snd-soc-hdmi-codec-objs := hdmi.o
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm1792a-codec-objs := pcm1792a.o
snd-soc-pcm3008-objs := pcm3008.o
@@ -216,6 +216,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o
+obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o
obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
@@ -264,7 +265,6 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
-obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
new file mode 100644
index 000000000000..07a266460ec3
--- /dev/null
+++ b/sound/soc/codecs/ak4613.c
@@ -0,0 +1,497 @@
+/*
+ * ak4613.c -- Asahi Kasei ALSA Soc Audio driver
+ *
+ * Copyright (C) 2015 Renesas Electronics Corporation
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on ak4642.c by Kuninori Morimoto
+ * Based on wm8731.c by Richard Purdie
+ * Based on ak4535.c by Richard Purdie
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#define PW_MGMT1 0x00 /* Power Management 1 */
+#define PW_MGMT2 0x01 /* Power Management 2 */
+#define PW_MGMT3 0x02 /* Power Management 3 */
+#define CTRL1 0x03 /* Control 1 */
+#define CTRL2 0x04 /* Control 2 */
+#define DEMP1 0x05 /* De-emphasis1 */
+#define DEMP2 0x06 /* De-emphasis2 */
+#define OFD 0x07 /* Overflow Detect */
+#define ZRD 0x08 /* Zero Detect */
+#define ICTRL 0x09 /* Input Control */
+#define OCTRL 0x0a /* Output Control */
+#define LOUT1 0x0b /* LOUT1 Volume Control */
+#define ROUT1 0x0c /* ROUT1 Volume Control */
+#define LOUT2 0x0d /* LOUT2 Volume Control */
+#define ROUT2 0x0e /* ROUT2 Volume Control */
+#define LOUT3 0x0f /* LOUT3 Volume Control */
+#define ROUT3 0x10 /* ROUT3 Volume Control */
+#define LOUT4 0x11 /* LOUT4 Volume Control */
+#define ROUT4 0x12 /* ROUT4 Volume Control */
+#define LOUT5 0x13 /* LOUT5 Volume Control */
+#define ROUT5 0x14 /* ROUT5 Volume Control */
+#define LOUT6 0x15 /* LOUT6 Volume Control */
+#define ROUT6 0x16 /* ROUT6 Volume Control */
+
+/* PW_MGMT1 */
+#define RSTN BIT(0)
+#define PMDAC BIT(1)
+#define PMADC BIT(2)
+#define PMVR BIT(3)
+
+/* PW_MGMT2 */
+#define PMAD_ALL 0x7
+
+/* PW_MGMT3 */
+#define PMDA_ALL 0x3f
+
+/* CTRL1 */
+#define DIF0 BIT(3)
+#define DIF1 BIT(4)
+#define DIF2 BIT(5)
+#define TDM0 BIT(6)
+#define TDM1 BIT(7)
+#define NO_FMT (0xff)
+#define FMT_MASK (0xf8)
+
+/* CTRL2 */
+#define DFS_NORMAL_SPEED (0 << 2)
+#define DFS_DOUBLE_SPEED (1 << 2)
+#define DFS_QUAD_SPEED (2 << 2)
+
+struct ak4613_priv {
+ struct mutex lock;
+
+ unsigned int fmt;
+ u8 fmt_ctrl;
+ int cnt;
+};
+
+struct ak4613_formats {
+ unsigned int width;
+ unsigned int fmt;
+};
+
+struct ak4613_interface {
+ struct ak4613_formats capture;
+ struct ak4613_formats playback;
+};
+
+/*
+ * Playback Volume
+ *
+ * max : 0x00 : 0 dB
+ * ( 0.5 dB step )
+ * min : 0xFE : -127.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new ak4613_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6,
+ 0, 0xFF, 1, out_tlv),
+};
+
+static const struct reg_default ak4613_reg[] = {
+ { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 },
+ { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 },
+ { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 },
+ { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 },
+ { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 },
+ { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 },
+};
+
+#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3)
+#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt }
+static const struct ak4613_interface ak4613_iface[] = {
+ /* capture */ /* playback */
+ [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) },
+ [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) },
+ [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) },
+ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) },
+ [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) },
+};
+
+static const struct regmap_config ak4613_regmap_cfg = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = 0x16,
+ .reg_defaults = ak4613_reg,
+ .num_reg_defaults = ARRAY_SIZE(ak4613_reg),
+};
+
+static const struct of_device_id ak4613_of_match[] = {
+ { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg },
+ {},
+};
+MODULE_DEVICE_TABLE(of, ak4613_of_match);
+
+static const struct i2c_device_id ak4613_i2c_id[] = {
+ { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id);
+
+static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = {
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("LOUT3"),
+ SND_SOC_DAPM_OUTPUT("LOUT4"),
+ SND_SOC_DAPM_OUTPUT("LOUT5"),
+ SND_SOC_DAPM_OUTPUT("LOUT6"),
+
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT3"),
+ SND_SOC_DAPM_OUTPUT("ROUT4"),
+ SND_SOC_DAPM_OUTPUT("ROUT5"),
+ SND_SOC_DAPM_OUTPUT("ROUT6"),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("LIN1"),
+ SND_SOC_DAPM_INPUT("LIN2"),
+
+ SND_SOC_DAPM_INPUT("RIN1"),
+ SND_SOC_DAPM_INPUT("RIN2"),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0),
+ SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0),
+ SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0),
+ SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0),
+ SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0),
+ SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0),
+ SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0),
+};
+
+static const struct snd_soc_dapm_route ak4613_intercon[] = {
+ {"LOUT1", NULL, "DAC1"},
+ {"LOUT2", NULL, "DAC2"},
+ {"LOUT3", NULL, "DAC3"},
+ {"LOUT4", NULL, "DAC4"},
+ {"LOUT5", NULL, "DAC5"},
+ {"LOUT6", NULL, "DAC6"},
+
+ {"ROUT1", NULL, "DAC1"},
+ {"ROUT2", NULL, "DAC2"},
+ {"ROUT3", NULL, "DAC3"},
+ {"ROUT4", NULL, "DAC4"},
+ {"ROUT5", NULL, "DAC5"},
+ {"ROUT6", NULL, "DAC6"},
+
+ {"DAC1", NULL, "Playback"},
+ {"DAC2", NULL, "Playback"},
+ {"DAC3", NULL, "Playback"},
+ {"DAC4", NULL, "Playback"},
+ {"DAC5", NULL, "Playback"},
+ {"DAC6", NULL, "Playback"},
+
+ {"Capture", NULL, "ADC1"},
+ {"Capture", NULL, "ADC2"},
+
+ {"ADC1", NULL, "LIN1"},
+ {"ADC2", NULL, "LIN2"},
+
+ {"ADC1", NULL, "RIN1"},
+ {"ADC2", NULL, "RIN2"},
+};
+
+static void ak4613_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = codec->dev;
+
+ mutex_lock(&priv->lock);
+ priv->cnt--;
+ if (priv->cnt < 0) {
+ dev_err(dev, "unexpected counter error\n");
+ priv->cnt = 0;
+ }
+ if (!priv->cnt)
+ priv->fmt_ctrl = NO_FMT;
+ mutex_unlock(&priv->lock);
+}
+
+static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ fmt &= SND_SOC_DAIFMT_FORMAT_MASK;
+
+ switch (fmt) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_I2S:
+ priv->fmt = fmt;
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ak4613_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+ const struct ak4613_formats *fmts;
+ struct device *dev = codec->dev;
+ unsigned int width = params_width(params);
+ unsigned int fmt = priv->fmt;
+ unsigned int rate;
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ int i, ret;
+ u8 fmt_ctrl, ctrl2;
+
+ rate = params_rate(params);
+ switch (rate) {
+ case 32000:
+ case 44100:
+ case 48000:
+ ctrl2 = DFS_NORMAL_SPEED;
+ break;
+ case 88200:
+ case 96000:
+ ctrl2 = DFS_DOUBLE_SPEED;
+ break;
+ case 176400:
+ case 192000:
+ ctrl2 = DFS_QUAD_SPEED;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /*
+ * FIXME
+ *
+ * It doesn't support TDM at this point
+ */
+ fmt_ctrl = NO_FMT;
+ for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) {
+ fmts = (is_play) ? &ak4613_iface[i].playback :
+ &ak4613_iface[i].capture;
+
+ if (fmts->fmt != fmt)
+ continue;
+
+ if (fmt == SND_SOC_DAIFMT_RIGHT_J) {
+ if (fmts->width != width)
+ continue;
+ } else {
+ if (fmts->width < width)
+ continue;
+ }
+
+ fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i);
+ break;
+ }
+
+ ret = -EINVAL;
+ if (fmt_ctrl == NO_FMT)
+ goto hw_params_end;
+
+ mutex_lock(&priv->lock);
+ if ((priv->fmt_ctrl == NO_FMT) ||
+ (priv->fmt_ctrl == fmt_ctrl)) {
+ priv->fmt_ctrl = fmt_ctrl;
+ priv->cnt++;
+ ret = 0;
+ }
+ mutex_unlock(&priv->lock);
+
+ if (ret < 0)
+ goto hw_params_end;
+
+ snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl);
+ snd_soc_write(codec, CTRL2, ctrl2);
+
+hw_params_end:
+ if (ret < 0)
+ dev_warn(dev, "unsupported data width/format combination\n");
+
+ return ret;
+}
+
+static int ak4613_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 mgmt1 = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ mgmt1 |= RSTN;
+ /* fall through */
+ case SND_SOC_BIAS_PREPARE:
+ mgmt1 |= PMADC | PMDAC;
+ /* fall through */
+ case SND_SOC_BIAS_STANDBY:
+ mgmt1 |= PMVR;
+ /* fall through */
+ case SND_SOC_BIAS_OFF:
+ default:
+ break;
+ }
+
+ snd_soc_write(codec, PW_MGMT1, mgmt1);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops ak4613_dai_ops = {
+ .shutdown = ak4613_dai_shutdown,
+ .set_fmt = ak4613_dai_set_fmt,
+ .hw_params = ak4613_dai_hw_params,
+};
+
+#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_64000 |\
+ SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000 |\
+ SNDRV_PCM_RATE_176400 |\
+ SNDRV_PCM_RATE_192000)
+#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver ak4613_dai = {
+ .name = "ak4613-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AK4613_PCM_RATE,
+ .formats = AK4613_PCM_FMTBIT,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AK4613_PCM_RATE,
+ .formats = AK4613_PCM_FMTBIT,
+ },
+ .ops = &ak4613_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int ak4613_resume(struct snd_soc_codec *codec)
+{
+ struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+ regcache_mark_dirty(regmap);
+ return regcache_sync(regmap);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+ .resume = ak4613_resume,
+ .set_bias_level = ak4613_set_bias_level,
+ .controls = ak4613_snd_controls,
+ .num_controls = ARRAY_SIZE(ak4613_snd_controls),
+ .dapm_widgets = ak4613_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets),
+ .dapm_routes = ak4613_intercon,
+ .num_dapm_routes = ARRAY_SIZE(ak4613_intercon),
+};
+
+static int ak4613_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &i2c->dev;
+ struct device_node *np = dev->of_node;
+ const struct regmap_config *regmap_cfg;
+ struct regmap *regmap;
+ struct ak4613_priv *priv;
+
+ regmap_cfg = NULL;
+ if (np) {
+ const struct of_device_id *of_id;
+
+ of_id = of_match_device(ak4613_of_match, dev);
+ if (of_id)
+ regmap_cfg = of_id->data;
+ } else {
+ regmap_cfg = (const struct regmap_config *)id->driver_data;
+ }
+
+ if (!regmap_cfg)
+ return -EINVAL;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->fmt_ctrl = NO_FMT;
+ priv->cnt = 0;
+
+ mutex_init(&priv->lock);
+
+ i2c_set_clientdata(i2c, priv);
+
+ regmap = devm_regmap_init_i2c(i2c, regmap_cfg);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ return snd_soc_register_codec(dev, &soc_codec_dev_ak4613,
+ &ak4613_dai, 1);
+}
+
+static int ak4613_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver ak4613_i2c_driver = {
+ .driver = {
+ .name = "ak4613-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = ak4613_of_match,
+ },
+ .probe = ak4613_i2c_probe,
+ .remove = ak4613_i2c_remove,
+ .id_table = ak4613_i2c_id,
+};
+
+module_i2c_driver(ak4613_i2c_driver);
+
+MODULE_DESCRIPTION("Soc AK4613 driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4a90143d0e90..cda27c22812a 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -23,6 +23,8 @@
* AK4648 is tested.
*/
+#include <linux/clk.h>
+#include <linux/clk-provider.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
@@ -128,11 +130,8 @@
#define I2S (3 << 0)
/* MD_CTL2 */
-#define FS0 (1 << 0)
-#define FS1 (1 << 1)
-#define FS2 (1 << 2)
-#define FS3 (1 << 5)
-#define FS_MASK (FS0 | FS1 | FS2 | FS3)
+#define FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2))
+#define PSs(val) ((val & 0x3) << 6)
/* MD_CTL3 */
#define BST1 (1 << 3)
@@ -147,6 +146,7 @@ struct ak4642_drvdata {
struct ak4642_priv {
const struct ak4642_drvdata *drvdata;
+ struct clk *mcko;
};
/*
@@ -430,56 +430,56 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
+static int ak4642_set_mcko(struct snd_soc_codec *codec,
+ u32 frequency)
+{
+ u32 fs_list[] = {
+ [0] = 8000,
+ [1] = 12000,
+ [2] = 16000,
+ [3] = 24000,
+ [4] = 7350,
+ [5] = 11025,
+ [6] = 14700,
+ [7] = 22050,
+ [10] = 32000,
+ [11] = 48000,
+ [14] = 29400,
+ [15] = 44100,
+ };
+ u32 ps_list[] = {
+ [0] = 256,
+ [1] = 128,
+ [2] = 64,
+ [3] = 32
+ };
+ int ps, fs;
+
+ for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) {
+ for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) {
+ if (frequency == ps_list[ps] * fs_list[fs]) {
+ snd_soc_write(codec, MD_CTL2,
+ PSs(ps) | FSs(fs));
+ return 0;
+ }
+ }
+ }
+
+ return 0;
+}
+
static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
- u8 rate;
+ struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+ u32 rate = clk_get_rate(priv->mcko);
- switch (params_rate(params)) {
- case 7350:
- rate = FS2;
- break;
- case 8000:
- rate = 0;
- break;
- case 11025:
- rate = FS2 | FS0;
- break;
- case 12000:
- rate = FS0;
- break;
- case 14700:
- rate = FS2 | FS1;
- break;
- case 16000:
- rate = FS1;
- break;
- case 22050:
- rate = FS2 | FS1 | FS0;
- break;
- case 24000:
- rate = FS1 | FS0;
- break;
- case 29400:
- rate = FS3 | FS2 | FS1;
- break;
- case 32000:
- rate = FS3 | FS1;
- break;
- case 44100:
- rate = FS3 | FS2 | FS1 | FS0;
- break;
- case 48000:
- rate = FS3 | FS1 | FS0;
- break;
- default:
- return -EINVAL;
- }
- snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
+ if (!rate)
+ rate = params_rate(params) * 256;
- return 0;
+ return ak4642_set_mcko(codec, rate);
}
static int ak4642_set_bias_level(struct snd_soc_codec *codec,
@@ -532,7 +532,18 @@ static int ak4642_resume(struct snd_soc_codec *codec)
return 0;
}
+static int ak4642_probe(struct snd_soc_codec *codec)
+{
+ struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ if (priv->mcko)
+ ak4642_set_mcko(codec, clk_get_rate(priv->mcko));
+
+ return 0;
+}
+
static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+ .probe = ak4642_probe,
.resume = ak4642_resume,
.set_bias_level = ak4642_set_bias_level,
.controls = ak4642_snd_controls,
@@ -580,19 +591,54 @@ static const struct ak4642_drvdata ak4648_drvdata = {
.extended_frequencies = 1,
};
+#ifdef CONFIG_COMMON_CLK
+static struct clk *ak4642_of_parse_mcko(struct device *dev)
+{
+ struct device_node *np = dev->of_node;
+ struct clk *clk;
+ const char *clk_name = np->name;
+ const char *parent_clk_name = NULL;
+ u32 rate;
+
+ if (of_property_read_u32(np, "clock-frequency", &rate))
+ return NULL;
+
+ if (of_property_read_bool(np, "clocks"))
+ parent_clk_name = of_clk_get_parent_name(np, 0);
+
+ of_property_read_string(np, "clock-output-names", &clk_name);
+
+ clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
+ (parent_clk_name) ? 0 : CLK_IS_ROOT,
+ rate);
+ if (!IS_ERR(clk))
+ of_clk_add_provider(np, of_clk_src_simple_get, clk);
+
+ return clk;
+}
+#else
+#define ak4642_of_parse_mcko(d) 0
+#endif
+
static const struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
- struct device_node *np = i2c->dev.of_node;
+ struct device *dev = &i2c->dev;
+ struct device_node *np = dev->of_node;
const struct ak4642_drvdata *drvdata = NULL;
struct regmap *regmap;
struct ak4642_priv *priv;
+ struct clk *mcko = NULL;
if (np) {
const struct of_device_id *of_id;
- of_id = of_match_device(ak4642_of_match, &i2c->dev);
+ mcko = ak4642_of_parse_mcko(dev);
+ if (IS_ERR(mcko))
+ mcko = NULL;
+
+ of_id = of_match_device(ak4642_of_match, dev);
if (of_id)
drvdata = of_id->data;
} else {
@@ -600,15 +646,16 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
}
if (!drvdata) {
- dev_err(&i2c->dev, "Unknown device type\n");
+ dev_err(dev, "Unknown device type\n");
return -EINVAL;
}
- priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
priv->drvdata = drvdata;
+ priv->mcko = mcko;
i2c_set_clientdata(i2c, priv);
@@ -616,7 +663,7 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
if (IS_ERR(regmap))
return PTR_ERR(regmap);
- return snd_soc_register_codec(&i2c->dev,
+ return snd_soc_register_codec(dev,
&soc_codec_dev_ak4642, &ak4642_dai, 1);
}
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 8a2221ab3d10..ac21b85ff75f 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
0x4f5, 0x0da);
}
break;
+ default:
+ break;
}
return 0;
@@ -689,6 +691,15 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
ARIZONA_IN_VU, val);
}
+bool arizona_input_analog(struct snd_soc_codec *codec, int shift)
+{
+ unsigned int reg = ARIZONA_IN1L_CONTROL + ((shift / 2) * 8);
+ unsigned int val = snd_soc_read(codec, reg);
+
+ return !(val & ARIZONA_IN1_MODE_MASK);
+}
+EXPORT_SYMBOL_GPL(arizona_input_analog);
+
int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
int event)
{
@@ -725,6 +736,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES);
if (reg == 0)
arizona_in_set_vu(codec, 0);
+ break;
+ default:
+ break;
}
return 0;
@@ -806,6 +820,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
break;
}
break;
+ default:
+ break;
}
return 0;
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index ada0a418ff4b..7b68d05a0939 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -294,4 +294,6 @@ extern int arizona_init_dai(struct arizona_priv *priv, int dai);
int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
bool diff);
+extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift);
+
#endif
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
deleted file mode 100644
index bd42ad34e004..000000000000
--- a/sound/soc/codecs/hdmi.c
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * ALSA SoC codec driver for HDMI audio codecs.
- * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <linux/of.h>
-#include <linux/of_device.h>
-
-#define DRV_NAME "hdmi-audio-codec"
-
-static const struct snd_soc_dapm_widget hdmi_widgets[] = {
- SND_SOC_DAPM_INPUT("RX"),
- SND_SOC_DAPM_OUTPUT("TX"),
-};
-
-static const struct snd_soc_dapm_route hdmi_routes[] = {
- { "Capture", NULL, "RX" },
- { "TX", NULL, "Playback" },
-};
-
-static struct snd_soc_dai_driver hdmi_codec_dai = {
- .name = "hdmi-hifi",
- .playback = {
- .stream_name = "Playback",
- .channels_min = 2,
- .channels_max = 8,
- .rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
- .sig_bits = 24,
- },
- .capture = {
- .stream_name = "Capture",
- .channels_min = 2,
- .channels_max = 2,
- .rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE,
- },
-
-};
-
-#ifdef CONFIG_OF
-static const struct of_device_id hdmi_audio_codec_ids[] = {
- { .compatible = "linux,hdmi-audio", },
- { }
-};
-MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids);
-#endif
-
-static struct snd_soc_codec_driver hdmi_codec = {
- .dapm_widgets = hdmi_widgets,
- .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
- .dapm_routes = hdmi_routes,
- .num_dapm_routes = ARRAY_SIZE(hdmi_routes),
- .ignore_pmdown_time = true,
-};
-
-static int hdmi_codec_probe(struct platform_device *pdev)
-{
- return snd_soc_register_codec(&pdev->dev, &hdmi_codec,
- &hdmi_codec_dai, 1);
-}
-
-static int hdmi_codec_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_codec(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver hdmi_codec_driver = {
- .driver = {
- .name = DRV_NAME,
- .of_match_table = of_match_ptr(hdmi_audio_codec_ids),
- },
-
- .probe = hdmi_codec_probe,
- .remove = hdmi_codec_remove,
-};
-
-module_platform_driver(hdmi_codec_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("ASoC generic HDMI codec driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 5c101af0ac63..080cc1ce3963 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2829,6 +2829,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
snd_soc_dapm_sync(dapm);
rt5645->jack_type = SND_JACK_HEADPHONE;
}
+ if (rt5645->pdata.jd_invert)
+ regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+ RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
} else { /* jack out */
rt5645->jack_type = 0;
@@ -2847,6 +2850,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
snd_soc_dapm_disable_pin(dapm, "LDO2");
snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
snd_soc_dapm_sync(dapm);
+ if (rt5645->pdata.jd_invert)
+ regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+ RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR);
}
return rt5645->jack_type;
@@ -3212,6 +3218,32 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
{ }
};
+static struct rt5645_platform_data buddy_platform_data = {
+ .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5,
+ .dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
+ .jd_mode = 3,
+ .jd_invert = true,
+};
+
+static int buddy_quirk_cb(const struct dmi_system_id *id)
+{
+ rt5645_pdata = &buddy_platform_data;
+
+ return 1;
+}
+
+static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = {
+ {
+ .ident = "Chrome Buddy",
+ .callback = buddy_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"),
+ },
+ },
+ { }
+};
+
+
static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev)
{
rt5645->pdata.in2_diff = device_property_read_bool(dev,
@@ -3244,7 +3276,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
if (pdata)
rt5645->pdata = *pdata;
- else if (dmi_check_system(dmi_platform_intel_braswell))
+ else if (dmi_check_system(dmi_platform_intel_braswell) ||
+ dmi_check_system(dmi_platform_intel_broadwell))
rt5645->pdata = *rt5645_pdata;
else
rt5645_parse_dt(rt5645, &i2c->dev);
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 8c964cfb120d..61bc8ab77646 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -779,8 +779,6 @@
#define RT5645_PWR_CLS_D_R_BIT 9
#define RT5645_PWR_CLS_D_L (0x1 << 8)
#define RT5645_PWR_CLS_D_L_BIT 8
-#define RT5645_PWR_ADC_R (0x1 << 1)
-#define RT5645_PWR_ADC_R_BIT 1
#define RT5645_PWR_DAC_L2 (0x1 << 7)
#define RT5645_PWR_DAC_L2_BIT 7
#define RT5645_PWR_DAC_R2 (0x1 << 6)
@@ -1628,6 +1626,10 @@
#define RT5645_OT_P_NOR (0x0 << 10)
#define RT5645_OT_P_INV (0x1 << 10)
#define RT5645_IRQ_JD_1_1_EN (0x1 << 9)
+#define RT5645_JD_1_1_MASK (0x1 << 7)
+#define RT5645_JD_1_1_SFT 7
+#define RT5645_JD_1_1_NOR (0x0 << 7)
+#define RT5645_JD_1_1_INV (0x1 << 7)
/* IRQ Control 2 (0xbe) */
#define RT5645_IRQ_MB1_OC_MASK (0x1 << 15)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 8739126a1f6f..a564759845f9 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -80,6 +80,7 @@ struct aic3x_priv {
unsigned int sysclk;
unsigned int dai_fmt;
unsigned int tdm_delay;
+ unsigned int slot_width;
struct list_head list;
int master;
int gpio_reset;
@@ -1025,10 +1026,14 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
u16 d, pll_d = 1;
int clk;
+ int width = aic3x->slot_width;
+
+ if (!width)
+ width = params_width(params);
/* select data word length */
data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
- switch (params_width(params)) {
+ switch (width) {
case 16:
break;
case 20:
@@ -1170,12 +1175,16 @@ static int aic3x_prepare(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int delay = 0;
+ int width = aic3x->slot_width;
+
+ if (!width)
+ width = substream->runtime->sample_bits;
/* TDM slot selection only valid in DSP_A/_B mode */
if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A)
- delay += (aic3x->tdm_delay + 1);
+ delay += (aic3x->tdm_delay*width + 1);
else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B)
- delay += aic3x->tdm_delay;
+ delay += aic3x->tdm_delay*width;
/* Configure data delay */
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
@@ -1296,7 +1305,20 @@ static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- aic3x->tdm_delay = lsb * slot_width;
+ switch (slot_width) {
+ case 16:
+ case 20:
+ case 24:
+ case 32:
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported slot width %d\n", slot_width);
+ return -EINVAL;
+ }
+
+
+ aic3x->tdm_delay = lsb;
+ aic3x->slot_width = slot_width;
/* DOUT in high-impedance on inactive bit clocks */
snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 9756578fc752..c04c0bc6f58a 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -38,6 +38,12 @@
struct wm5110_priv {
struct arizona_priv core;
struct arizona_fll fll[2];
+
+ unsigned int in_value;
+ int in_pre_pending;
+ int in_post_pending;
+
+ unsigned int in_pga_cache[6];
};
static const struct wm_adsp_region wm5110_dsp1_regions[] = {
@@ -428,6 +434,127 @@ err:
return ret;
}
+static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct snd_soc_card *card = dapm->card;
+ int ret;
+
+ /*
+ * PGA Volume is also used as part of the enable sequence, so
+ * usage of it should be avoided whilst that is running.
+ */
+ mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_get_volsw_range(kcontrol, ucontrol);
+
+ mutex_unlock(&card->dapm_mutex);
+
+ return ret;
+}
+
+static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct snd_soc_card *card = dapm->card;
+ int ret;
+
+ /*
+ * PGA Volume is also used as part of the enable sequence, so
+ * usage of it should be avoided whilst that is running.
+ */
+ mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_put_volsw_range(kcontrol, ucontrol);
+
+ mutex_unlock(&card->dapm_mutex);
+
+ return ret;
+}
+
+static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ unsigned int reg, mask;
+ struct reg_sequence analog_seq[] = {
+ { 0x80, 0x3 },
+ { 0x35d, 0 },
+ { 0x80, 0x0 },
+ };
+
+ reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4);
+ mask = ARIZONA_IN1L_PGA_VOL_MASK;
+
+ switch (event) {
+ case SND_SOC_DAPM_WILL_PMU:
+ wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2);
+ wm5110->in_pre_pending++;
+ wm5110->in_post_pending++;
+ return 0;
+ case SND_SOC_DAPM_PRE_PMU:
+ wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg);
+
+ snd_soc_update_bits(codec, reg, mask,
+ 0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT);
+
+ wm5110->in_pre_pending--;
+ if (wm5110->in_pre_pending == 0) {
+ analog_seq[1].def = wm5110->in_value;
+ regmap_multi_reg_write_bypassed(arizona->regmap,
+ analog_seq,
+ ARRAY_SIZE(analog_seq));
+
+ msleep(55);
+
+ wm5110->in_value = 0;
+ }
+
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, reg, mask,
+ wm5110->in_pga_cache[w->shift]);
+
+ wm5110->in_post_pending--;
+ if (wm5110->in_post_pending == 0)
+ regmap_multi_reg_write_bypassed(arizona->regmap,
+ analog_seq,
+ ARRAY_SIZE(analog_seq));
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int wm5110_in_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+
+ switch (arizona->rev) {
+ case 0 ... 4:
+ if (arizona_input_analog(codec, w->shift))
+ wm5110_in_analog_ev(w, kcontrol, event);
+
+ break;
+ default:
+ break;
+ }
+
+ return arizona_in_ev(w, kcontrol, event);
+}
+
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
@@ -454,18 +581,24 @@ SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]),
SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]),
SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]),
-SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
- ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
- ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
- ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
- ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
- ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
- ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
+ ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
+ ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
+ ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
+ ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum),
@@ -896,29 +1029,35 @@ SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7d45d98a861f..4495a40a9468 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -80,12 +80,13 @@ struct davinci_mcasp {
/* McASP specific data */
int tdm_slots;
+ u32 tdm_mask[2];
+ int slot_width;
u8 op_mode;
u8 num_serializer;
u8 *serial_dir;
u8 version;
u8 bclk_div;
- u16 bclk_lrclk_ratio;
int streams;
u32 irq_request[2];
int dma_request[2];
@@ -556,8 +557,21 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
mcasp->bclk_div = div;
break;
- case 2: /* BCLK/LRCLK ratio */
- mcasp->bclk_lrclk_ratio = div;
+ case 2: /*
+ * BCLK/LRCLK ratio descries how many bit-clock cycles
+ * fit into one frame. The clock ratio is given for a
+ * full period of data (for I2S format both left and
+ * right channels), so it has to be divided by number
+ * of tdm-slots (for I2S - divided by 2).
+ * Instead of storing this ratio, we calculate a new
+ * tdm_slot width by dividing the the ratio by the
+ * number of configured tdm slots.
+ */
+ mcasp->slot_width = div / mcasp->tdm_slots;
+ if (div % mcasp->tdm_slots)
+ dev_warn(mcasp->dev,
+ "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots",
+ __func__, div, mcasp->tdm_slots);
break;
default:
@@ -596,12 +610,92 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
return 0;
}
+/* All serializers must have equal number of channels */
+static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream,
+ int serializers)
+{
+ struct snd_pcm_hw_constraint_list *cl = &mcasp->chconstr[stream];
+ unsigned int *list = (unsigned int *) cl->list;
+ int slots = mcasp->tdm_slots;
+ int i, count = 0;
+
+ if (mcasp->tdm_mask[stream])
+ slots = hweight32(mcasp->tdm_mask[stream]);
+
+ for (i = 2; i <= slots; i++)
+ list[count++] = i;
+
+ for (i = 2; i <= serializers; i++)
+ list[count++] = i*slots;
+
+ cl->count = count;
+
+ return 0;
+}
+
+static int davinci_mcasp_set_ch_constraints(struct davinci_mcasp *mcasp)
+{
+ int rx_serializers = 0, tx_serializers = 0, ret, i;
+
+ for (i = 0; i < mcasp->num_serializer; i++)
+ if (mcasp->serial_dir[i] == TX_MODE)
+ tx_serializers++;
+ else if (mcasp->serial_dir[i] == RX_MODE)
+ rx_serializers++;
+
+ ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_PLAYBACK,
+ tx_serializers);
+ if (ret)
+ return ret;
+
+ ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_CAPTURE,
+ rx_serializers);
+
+ return ret;
+}
+
+
+static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask,
+ unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(mcasp->dev,
+ "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n",
+ __func__, tx_mask, rx_mask, slots, slot_width);
+
+ if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) {
+ dev_err(mcasp->dev,
+ "Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n",
+ tx_mask, rx_mask, slots);
+ return -EINVAL;
+ }
+
+ if (slot_width &&
+ (slot_width < 8 || slot_width > 32 || slot_width % 4 != 0)) {
+ dev_err(mcasp->dev, "%s: Unsupported slot_width %d\n",
+ __func__, slot_width);
+ return -EINVAL;
+ }
+
+ mcasp->tdm_slots = slots;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask;
+ mcasp->slot_width = slot_width;
+
+ return davinci_mcasp_set_ch_constraints(mcasp);
+}
+
static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
- int word_length)
+ int sample_width)
{
u32 fmt;
- u32 tx_rotate = (word_length / 4) & 0x7;
- u32 mask = (1ULL << word_length) - 1;
+ u32 tx_rotate = (sample_width / 4) & 0x7;
+ u32 mask = (1ULL << sample_width) - 1;
+ u32 slot_width = sample_width;
+
/*
* For captured data we should not rotate, inversion and masking is
* enoguh to get the data to the right position:
@@ -614,28 +708,23 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
u32 rx_rotate = 0;
/*
- * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()
- * callback, take it into account here. That allows us to for example
- * send 32 bits per channel to the codec, while only 16 of them carry
- * audio payload.
- * The clock ratio is given for a full period of data (for I2S format
- * both left and right channels), so it has to be divided by number of
- * tdm-slots (for I2S - divided by 2).
+ * Setting the tdm slot width either with set_clkdiv() or
+ * set_tdm_slot() allows us to for example send 32 bits per
+ * channel to the codec, while only 16 of them carry audio
+ * payload.
*/
- if (mcasp->bclk_lrclk_ratio) {
- u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots;
-
+ if (mcasp->slot_width) {
/*
- * When we have more bclk then it is needed for the data, we
- * need to use the rotation to move the received samples to have
- * correct alignment.
+ * When we have more bclk then it is needed for the
+ * data, we need to use the rotation to move the
+ * received samples to have correct alignment.
*/
- rx_rotate = (slot_length - word_length) / 4;
- word_length = slot_length;
+ slot_width = mcasp->slot_width;
+ rx_rotate = (slot_width - sample_width) / 4;
}
/* mapping of the XSSZ bit-field as described in the datasheet */
- fmt = (word_length >> 1) - 1;
+ fmt = (slot_width >> 1) - 1;
if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt),
@@ -776,33 +865,50 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
/*
* If more than one serializer is needed, then use them with
- * their specified tdm_slots count. Otherwise, one serializer
- * can cope with the transaction using as many slots as channels
- * in the stream, requires channels symmetry
+ * all the specified tdm_slots. Otherwise, one serializer can
+ * cope with the transaction using just as many slots as there
+ * are channels in the stream.
*/
- active_serializers = (channels + total_slots - 1) / total_slots;
- if (active_serializers == 1)
- active_slots = channels;
- else
- active_slots = total_slots;
-
- for (i = 0; i < active_slots; i++)
- mask |= (1 << i);
+ if (mcasp->tdm_mask[stream]) {
+ active_slots = hweight32(mcasp->tdm_mask[stream]);
+ active_serializers = (channels + active_slots - 1) /
+ active_slots;
+ if (active_serializers == 1) {
+ active_slots = channels;
+ for (i = 0; i < total_slots; i++) {
+ if ((1 << i) & mcasp->tdm_mask[stream]) {
+ mask |= (1 << i);
+ if (--active_slots <= 0)
+ break;
+ }
+ }
+ }
+ } else {
+ active_serializers = (channels + total_slots - 1) / total_slots;
+ if (active_serializers == 1)
+ active_slots = channels;
+ else
+ active_slots = total_slots;
+ for (i = 0; i < active_slots; i++)
+ mask |= (1 << i);
+ }
mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
if (!mcasp->dat_port)
busel = TXSEL;
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
- FSXMOD(total_slots), FSXMOD(0x1FF));
-
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
- FSRMOD(total_slots), FSRMOD(0x1FF));
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+ FSXMOD(total_slots), FSXMOD(0x1FF));
+ } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+ FSRMOD(total_slots), FSRMOD(0x1FF));
+ }
return 0;
}
@@ -922,6 +1028,9 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
int sbits = params_width(params);
int ppm, div;
+ if (mcasp->slot_width)
+ sbits = mcasp->slot_width;
+
div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots,
&ppm);
if (ppm)
@@ -1027,6 +1136,9 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
struct snd_interval range;
int i;
+ if (rd->mcasp->slot_width)
+ sbits = rd->mcasp->slot_width;
+
snd_interval_any(&range);
range.empty = 1;
@@ -1069,10 +1181,14 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
if (snd_mask_test(fmt, i)) {
- uint bclk_freq = snd_pcm_format_width(i)*slots*rate;
+ uint sbits = snd_pcm_format_width(i);
int ppm;
- davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm);
+ if (rd->mcasp->slot_width)
+ sbits = rd->mcasp->slot_width;
+
+ davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate,
+ &ppm);
if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
snd_mask_set(&nfmt, i);
count++;
@@ -1094,6 +1210,10 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
&mcasp->ruledata[substream->stream];
u32 max_channels = 0;
int i, dir;
+ int tdm_slots = mcasp->tdm_slots;
+
+ if (mcasp->tdm_mask[substream->stream])
+ tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
mcasp->substreams[substream->stream] = substream;
@@ -1114,7 +1234,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
max_channels++;
}
ruledata->serializers = max_channels;
- max_channels *= mcasp->tdm_slots;
+ max_channels *= tdm_slots;
/*
* If the already active stream has less channels than the calculated
* limnit based on the seirializers * tdm_slots, we need to use that as
@@ -1124,15 +1244,25 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
*/
if (mcasp->channels && mcasp->channels < max_channels)
max_channels = mcasp->channels;
+ /*
+ * But we can always allow channels upto the amount of
+ * the available tdm_slots.
+ */
+ if (max_channels < tdm_slots)
+ max_channels = tdm_slots;
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
2, max_channels);
- if (mcasp->chconstr[substream->stream].count)
- snd_pcm_hw_constraint_list(substream->runtime,
- 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- &mcasp->chconstr[substream->stream]);
+ snd_pcm_hw_constraint_list(substream->runtime,
+ 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &mcasp->chconstr[substream->stream]);
+
+ if (mcasp->slot_width)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ 8, mcasp->slot_width);
/*
* If we rely on implicit BCLK divider setting we should
@@ -1184,6 +1314,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
.set_fmt = davinci_mcasp_set_dai_fmt,
.set_clkdiv = davinci_mcasp_set_clkdiv,
.set_sysclk = davinci_mcasp_set_sysclk,
+ .set_tdm_slot = davinci_mcasp_set_tdm_slot,
};
static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
@@ -1514,59 +1645,6 @@ nodata:
return pdata;
}
-/* All serializers must have equal number of channels */
-static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp,
- struct snd_pcm_hw_constraint_list *cl,
- int serializers)
-{
- unsigned int *list;
- int i, count = 0;
-
- if (serializers <= 1)
- return 0;
-
- list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
- (mcasp->tdm_slots + serializers - 2),
- GFP_KERNEL);
- if (!list)
- return -ENOMEM;
-
- for (i = 2; i <= mcasp->tdm_slots; i++)
- list[count++] = i;
-
- for (i = 2; i <= serializers; i++)
- list[count++] = i*mcasp->tdm_slots;
-
- cl->count = count;
- cl->list = list;
-
- return 0;
-}
-
-
-static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp)
-{
- int rx_serializers = 0, tx_serializers = 0, ret, i;
-
- for (i = 0; i < mcasp->num_serializer; i++)
- if (mcasp->serial_dir[i] == TX_MODE)
- tx_serializers++;
- else if (mcasp->serial_dir[i] == RX_MODE)
- rx_serializers++;
-
- ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
- SNDRV_PCM_STREAM_PLAYBACK],
- tx_serializers);
- if (ret)
- return ret;
-
- ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
- SNDRV_PCM_STREAM_CAPTURE],
- rx_serializers);
-
- return ret;
-}
-
enum {
PCM_EDMA,
PCM_SDMA,
@@ -1783,7 +1861,28 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
}
- ret = davinci_mcasp_init_ch_constraints(mcasp);
+ /* Allocate memory for long enough list for all possible
+ * scenarios. Maximum number tdm slots is 32 and there cannot
+ * be more serializers than given in the configuration. The
+ * serializer directions could be taken into account, but it
+ * would make code much more complex and save only couple of
+ * bytes.
+ */
+ mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list =
+ devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+ (32 + mcasp->num_serializer - 2),
+ GFP_KERNEL);
+
+ mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list =
+ devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+ (32 + mcasp->num_serializer - 2),
+ GFP_KERNEL);
+
+ if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
+ !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list)
+ return -ENOMEM;
+
+ ret = davinci_mcasp_set_ch_constraints(mcasp);
if (ret)
goto err;
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 96f55ae75c71..0901d5e20df2 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -593,6 +593,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-wm8960", },
{}
};
+MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
static struct platform_driver fsl_asoc_card_driver = {
.probe = fsl_asoc_card_probe,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a18fd92c4a85..9366b5a42e1d 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -801,6 +801,7 @@ static const struct of_device_id fsl_sai_ids[] = {
{ .compatible = "fsl,imx6sx-sai", },
{ /* sentinel */ }
};
+MODULE_DEVICE_TABLE(of, fsl_sai_ids);
static struct platform_driver fsl_sai_driver = {
.probe = fsl_sai_probe,
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 3ff76d419436..54c33204541f 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -151,7 +151,9 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
}
if (set->slots) {
- ret = snd_soc_dai_set_tdm_slot(dai, 0, 0,
+ ret = snd_soc_dai_set_tdm_slot(dai,
+ set->tx_slot_mask,
+ set->rx_slot_mask,
set->slots,
set->slot_width);
if (ret && ret != -ENOTSUPP) {
@@ -243,7 +245,9 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
return ret;
/* Parse TDM slot */
- ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width);
+ ret = snd_soc_of_parse_tdm_slot(np, &dai->tx_slot_mask,
+ &dai->rx_slot_mask,
+ &dai->slots, &dai->slot_width);
if (ret)
return ret;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 683e50116152..5e9c316c142a 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -368,23 +368,6 @@ static void sst_media_close(struct snd_pcm_substream *substream,
kfree(stream);
}
-static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai,
- struct snd_pcm_substream *substream)
-{
- struct sst_data *sst = snd_soc_dai_get_drvdata(dai);
- struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
- struct sst_runtime_stream *stream =
- substream->runtime->private_data;
- u32 str_id = stream->stream_info.str_id;
- unsigned int pipe_id;
-
- pipe_id = map[str_id].device_id;
-
- dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n",
- pipe_id, str_id);
- return pipe_id;
-}
-
static int sst_media_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 8bafaf6ceab1..3f8a1e10bed0 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -266,18 +266,11 @@ static int broadwell_audio_probe(struct platform_device *pdev)
{
broadwell_rt286.dev = &pdev->dev;
- return snd_soc_register_card(&broadwell_rt286);
-}
-
-static int broadwell_audio_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&broadwell_rt286);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286);
}
static struct platform_driver broadwell_audio = {
.probe = broadwell_audio_probe,
- .remove = broadwell_audio_remove,
.driver = {
.name = "broadwell-audio",
},
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 7d617bf493bc..bea26730873c 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -510,17 +510,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
},
},
{
- .name = "DMIC23 Pin",
- .ops = &skl_dmic_dai_ops,
- .capture = {
- .stream_name = "DMIC23 Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
- },
-},
-{
.name = "HD-Codec Pin",
.ops = &skl_link_dai_ops,
.playback = {
@@ -538,28 +527,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
-{
- .name = "HD-Codec-SPK Pin",
- .ops = &skl_link_dai_ops,
- .playback = {
- .stream_name = "HD-Codec-SPK Tx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
-},
-{
- .name = "HD-Codec-AMIC Pin",
- .ops = &skl_link_dai_ops,
- .capture = {
- .stream_name = "HD-Codec-AMIC Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
-},
};
static int skl_platform_open(struct snd_pcm_substream *substream)
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index b05fb1c1a848..794a3499e567 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -485,6 +485,7 @@ static const struct of_device_id jz4740_of_matches[] = {
{ .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 },
{ /* sentinel */ }
};
+MODULE_DEVICE_TABLE(of, jz4740_of_matches);
#endif
static int jz4740_i2s_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
index de7563bdc5c2..e0304d544f26 100644
--- a/sound/soc/kirkwood/armada-370-db.c
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -130,6 +130,7 @@ static const struct of_device_id a370db_dt_ids[] = {
{ .compatible = "marvell,a370db-audio" },
{ },
};
+MODULE_DEVICE_TABLE(of, a370db_dt_ids);
static struct platform_driver a370db_driver = {
.driver = {
diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c
index 684e8a78bed0..71a1a35047ba 100644
--- a/sound/soc/mediatek/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173-max98090.c
@@ -179,21 +179,13 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
}
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
return ret;
}
-static int mt8173_max98090_dev_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static const struct of_device_id mt8173_max98090_dt_match[] = {
{ .compatible = "mediatek,mt8173-max98090", },
{ }
@@ -209,7 +201,6 @@ static struct platform_driver mt8173_max98090_driver = {
#endif
},
.probe = mt8173_max98090_dev_probe,
- .remove = mt8173_max98090_dev_remove,
};
module_platform_driver(mt8173_max98090_driver);
diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
index 86cf9752f18a..50ba538eccb3 100644
--- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -246,21 +246,13 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
return ret;
}
-static int mt8173_rt5650_rt5676_dev_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static const struct of_device_id mt8173_rt5650_rt5676_dt_match[] = {
{ .compatible = "mediatek,mt8173-rt5650-rt5676", },
{ }
@@ -276,7 +268,6 @@ static struct platform_driver mt8173_rt5650_rt5676_driver = {
#endif
},
.probe = mt8173_rt5650_rt5676_dev_probe,
- .remove = mt8173_rt5650_rt5676_dev_remove,
};
module_platform_driver(mt8173_rt5650_rt5676_driver);
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 6e6fce6a14ba..2b23ffbac6b1 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
ret);
@@ -154,12 +154,8 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
static int mxs_sgtl5000_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
mxs_saif_put_mclk(0);
- snd_soc_unregister_card(card);
-
return 0;
}
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index 2b26318bc200..6147e86e9b0f 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -116,26 +116,19 @@ static int brownstone_probe(struct platform_device *pdev)
int ret;
brownstone.dev = &pdev->dev;
- ret = snd_soc_register_card(&brownstone);
+ ret = devm_snd_soc_register_card(&pdev->dev, &brownstone);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int brownstone_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&brownstone);
- return 0;
-}
-
static struct platform_driver mmp_driver = {
.driver = {
.name = "brownstone-audio",
.pm = &snd_soc_pm_ops,
},
.probe = brownstone_probe,
- .remove = brownstone_remove,
};
module_platform_driver(mmp_driver);
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 3580d10c9f28..c97dc13d3608 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -295,28 +295,19 @@ static int corgi_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int corgi_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver corgi_driver = {
.driver = {
.name = "corgi-audio",
.pm = &snd_soc_pm_ops,
},
.probe = corgi_probe,
- .remove = corgi_remove,
};
module_platform_driver(corgi_driver);
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index d72e124a3676..1de876529aa1 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -138,7 +138,7 @@ static int e740_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -149,10 +149,7 @@ static int e740_probe(struct platform_device *pdev)
static int e740_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 48f2d7c2e68c..b7eb7cd5df7d 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -120,7 +120,7 @@ static int e750_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -131,10 +131,7 @@ static int e750_probe(struct platform_device *pdev)
static int e750_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 45d4bd46fff6..41bf71466a7b 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -119,7 +119,7 @@ static int e800_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -130,10 +130,7 @@ static int e800_probe(struct platform_device *pdev)
static int e800_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 9f8be7cd567e..ecbf2873b7ff 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -193,7 +193,7 @@ static int hx4700_audio_probe(struct platform_device *pdev)
return ret;
snd_soc_card_hx4700.dev = &pdev->dev;
- ret = snd_soc_register_card(&snd_soc_card_hx4700);
+ ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700);
if (ret)
gpio_free_array(hx4700_audio_gpios,
ARRAY_SIZE(hx4700_audio_gpios));
@@ -203,8 +203,6 @@ static int hx4700_audio_probe(struct platform_device *pdev)
static int hx4700_audio_remove(struct platform_device *pdev)
{
- snd_soc_unregister_card(&snd_soc_card_hx4700);
-
gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
gpio_set_value(GPIO107_HX4700_SPK_nSD, 0);
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
index 29fabbfd21f1..9d0e40771ef5 100644
--- a/sound/soc/pxa/imote2.c
+++ b/sound/soc/pxa/imote2.c
@@ -72,28 +72,19 @@ static int imote2_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int imote2_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver imote2_driver = {
.driver = {
.name = "imote2-audio",
.pm = &snd_soc_pm_ops,
},
.probe = imote2_probe,
- .remove = imote2_remove,
};
module_platform_driver(imote2_driver);
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index a9615a574546..29bc60e85e92 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -181,7 +181,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev)
return -ENODEV;
mioa701.dev = &pdev->dev;
- rc = snd_soc_register_card(&mioa701);
+ rc = devm_snd_soc_register_card(&pdev->dev, &mioa701);
if (!rc)
dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
"lead to overheating and possible destruction of your device."
@@ -189,17 +189,8 @@ static int mioa701_wm9713_probe(struct platform_device *pdev)
return rc;
}
-static int mioa701_wm9713_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver mioa701_wm9713_driver = {
.probe = mioa701_wm9713_probe,
- .remove = mioa701_wm9713_remove,
.driver = {
.name = "mioa701-wm9713",
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index c20bbc042425..4e74d9573f03 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -140,22 +140,15 @@ static int palm27x_asoc_probe(struct platform_device *pdev)
palm27x_asoc.dev = &pdev->dev;
- ret = snd_soc_register_card(&palm27x_asoc);
+ ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int palm27x_asoc_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&palm27x_asoc);
- return 0;
-}
-
static struct platform_driver palm27x_wm9712_driver = {
.probe = palm27x_asoc_probe,
- .remove = palm27x_asoc_remove,
.driver = {
.name = "palm27x-asoc",
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 80b457ac522a..84d0e2e50808 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -267,28 +267,19 @@ static int poodle_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int poodle_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver poodle_driver = {
.driver = {
.name = "poodle-audio",
.pm = &snd_soc_pm_ops,
},
.probe = poodle_probe,
- .remove = poodle_remove,
};
module_platform_driver(poodle_driver);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 3da485ec1de7..da03fad1b9cd 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -809,6 +809,7 @@ static const struct of_device_id pxa_ssp_of_ids[] = {
{ .compatible = "mrvl,pxa-ssp-dai" },
{}
};
+MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids);
#endif
static int asoc_ssp_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 831ee37d2e3e..29a3fdbb7b59 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -132,6 +132,7 @@ static const struct of_device_id snd_soc_pxa_audio_match[] = {
{ .compatible = "mrvl,pxa-pcm-audio" },
{ }
};
+MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match);
#endif
static struct platform_driver pxa_pcm_driver = {
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 461123ad5ff2..b00222620fd0 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -305,7 +305,7 @@ static int spitz_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -322,9 +322,6 @@ err1:
static int spitz_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
gpio_free(spitz_mic_gpio);
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f59f566551ef..49518dd642aa 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -233,7 +233,7 @@ static int tosa_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -244,10 +244,7 @@ static int tosa_probe(struct platform_device *pdev)
static int tosa_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free(TOSA_GPIO_L_MUTE);
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index 1753c7d9e760..65c20f779177 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -128,7 +128,7 @@ static int ttc_dkb_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -136,22 +136,12 @@ static int ttc_dkb_probe(struct platform_device *pdev)
return ret;
}
-static int ttc_dkb_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
-}
-
static struct platform_driver ttc_dkb_driver = {
.driver = {
.name = "ttc-dkb-audio",
.pm = &snd_soc_pm_ops,
},
.probe = ttc_dkb_probe,
- .remove = ttc_dkb_remove,
};
module_platform_driver(ttc_dkb_driver);
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 97bc2023f08a..e5101e0d2d37 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -438,7 +438,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev)
if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) {
dev_err(&pdev->dev,
"%s() error getting mi2s-bit-clk: %ld\n",
- __func__, PTR_ERR(drvdata->mi2s_bit_clk[i]));
+ __func__,
+ PTR_ERR(drvdata->mi2s_bit_clk[dai_id]));
return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]);
}
}
diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig
index 58bae8e2cf5f..570905709d3a 100644
--- a/sound/soc/rockchip/Kconfig
+++ b/sound/soc/rockchip/Kconfig
@@ -17,7 +17,7 @@ config SND_SOC_ROCKCHIP_I2S
config SND_SOC_ROCKCHIP_MAX98090
tristate "ASoC support for Rockchip boards using a MAX98090 codec"
- depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB
+ depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP
select SND_SOC_ROCKCHIP_I2S
select SND_SOC_MAX98090
select SND_SOC_TS3A227E
@@ -27,7 +27,7 @@ config SND_SOC_ROCKCHIP_MAX98090
config SND_SOC_ROCKCHIP_RT5645
tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec"
- depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB
+ depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP
select SND_SOC_ROCKCHIP_I2S
select SND_SOC_RT5645
help
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 07114b0b0dc1..6ca90aaf141f 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU
config SND_SOC_RCAR
tristate "R-Car series SRU/SCU/SSIU/SSI support"
depends on DMA_OF
+ depends on COMMON_CLK
select SND_SIMPLE_CARD
select REGMAP_MMIO
help
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index fefc881dbac2..c4ebbb7a7b6f 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -7,7 +7,7 @@
* License. See the file "COPYING" in the main directory of this archive
* for more details.
*/
-#include <linux/sh_clk.h>
+#include <linux/clk-provider.h>
#include "rsnd.h"
#define CLKA 0
@@ -16,12 +16,26 @@
#define CLKI 3
#define CLKMAX 4
+#define CLKOUT 0
+#define CLKOUT1 1
+#define CLKOUT2 2
+#define CLKOUT3 3
+#define CLKOUTMAX 4
+
+#define BRRx_MASK(x) (0x3FF & x)
+
+static struct rsnd_mod_ops adg_ops = {
+ .name = "adg",
+};
+
struct rsnd_adg {
struct clk *clk[CLKMAX];
+ struct clk *clkout[CLKOUTMAX];
+ struct clk_onecell_data onecell;
+ struct rsnd_mod mod;
- int rbga_rate_for_441khz_div_6; /* RBGA */
- int rbgb_rate_for_48khz_div_6; /* RBGB */
- u32 ckr;
+ int rbga_rate_for_441khz; /* RBGA */
+ int rbgb_rate_for_48khz; /* RBGB */
};
#define for_each_rsnd_clk(pos, adg, i) \
@@ -29,8 +43,28 @@ struct rsnd_adg {
(i < CLKMAX) && \
((pos) = adg->clk[i]); \
i++)
+#define for_each_rsnd_clkout(pos, adg, i) \
+ for (i = 0; \
+ (i < CLKOUTMAX) && \
+ ((pos) = adg->clkout[i]); \
+ i++)
#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
+static u32 rsnd_adg_calculate_rbgx(unsigned long div)
+{
+ int i, ratio;
+
+ if (!div)
+ return 0;
+
+ for (i = 3; i >= 0; i--) {
+ ratio = 2 << (i * 2);
+ if (0 == (div % ratio))
+ return (u32)((i << 8) | ((div / ratio) - 1));
+ }
+
+ return ~0;
+}
static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
{
@@ -60,6 +94,9 @@ static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io)
{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
int id = rsnd_mod_id(mod);
int shift = (id % 2) ? 16 : 0;
u32 mask, val;
@@ -69,21 +106,26 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod,
val = val << shift;
mask = 0xffff << shift;
- rsnd_mod_bset(mod, CMDOUT_TIMSEL, mask, val);
+ rsnd_mod_bset(adg_mod, CMDOUT_TIMSEL, mask, val);
return 0;
}
-static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod,
+static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io,
u32 timsel)
{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
int is_play = rsnd_io_is_play(io);
- int id = rsnd_mod_id(mod);
+ int id = rsnd_mod_id(src_mod);
int shift = (id % 2) ? 16 : 0;
u32 mask, ws;
u32 in, out;
+ rsnd_mod_confirm_src(src_mod);
+
ws = rsnd_adg_ssi_ws_timing_gen2(io);
in = (is_play) ? timsel : ws;
@@ -95,37 +137,38 @@ static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod,
switch (id / 2) {
case 0:
- rsnd_mod_bset(mod, SRCIN_TIMSEL0, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL0, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL0, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL0, mask, out);
break;
case 1:
- rsnd_mod_bset(mod, SRCIN_TIMSEL1, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL1, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL1, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL1, mask, out);
break;
case 2:
- rsnd_mod_bset(mod, SRCIN_TIMSEL2, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL2, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL2, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL2, mask, out);
break;
case 3:
- rsnd_mod_bset(mod, SRCIN_TIMSEL3, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL3, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL3, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL3, mask, out);
break;
case 4:
- rsnd_mod_bset(mod, SRCIN_TIMSEL4, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL4, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL4, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL4, mask, out);
break;
}
return 0;
}
-int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
+int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io,
unsigned int src_rate,
unsigned int dst_rate)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod);
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
struct device *dev = rsnd_priv_to_dev(priv);
int idx, sel, div, step, ret;
u32 val, en;
@@ -134,10 +177,12 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */
clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */
clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */
- adg->rbga_rate_for_441khz_div_6,/* 0011: RBGA */
- adg->rbgb_rate_for_48khz_div_6, /* 0100: RBGB */
+ adg->rbga_rate_for_441khz, /* 0011: RBGA */
+ adg->rbgb_rate_for_48khz, /* 0100: RBGB */
};
+ rsnd_mod_confirm_src(src_mod);
+
min = ~0;
val = 0;
en = 0;
@@ -175,25 +220,27 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
return -EIO;
}
- ret = rsnd_adg_set_src_timsel_gen2(mod, io, val);
+ ret = rsnd_adg_set_src_timsel_gen2(src_mod, io, val);
if (ret < 0) {
dev_err(dev, "timsel error\n");
return ret;
}
- rsnd_mod_bset(mod, DIV_EN, en, en);
+ rsnd_mod_bset(adg_mod, DIV_EN, en, en);
dev_dbg(dev, "convert rate %d <-> %d\n", src_rate, dst_rate);
return 0;
}
-int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
+int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io)
{
u32 val = rsnd_adg_ssi_ws_timing_gen2(io);
- return rsnd_adg_set_src_timsel_gen2(mod, io, val);
+ rsnd_mod_confirm_src(src_mod);
+
+ return rsnd_adg_set_src_timsel_gen2(src_mod, io, val);
}
int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
@@ -202,6 +249,7 @@ int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
unsigned int dst_rate)
{
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
struct device *dev = rsnd_priv_to_dev(priv);
int idx, sel, div, shift;
u32 mask, val;
@@ -211,8 +259,8 @@ int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
clk_get_rate(adg->clk[CLKB]), /* 001: CLKB */
clk_get_rate(adg->clk[CLKC]), /* 010: CLKC */
0, /* 011: MLBCLK (not used) */
- adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */
- adg->rbgb_rate_for_48khz_div_6, /* 101: RBGB */
+ adg->rbga_rate_for_441khz, /* 100: RBGA */
+ adg->rbgb_rate_for_48khz, /* 101: RBGB */
};
/* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */
@@ -238,13 +286,13 @@ find_rate:
switch (id / 4) {
case 0:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL3, mask, val);
break;
case 1:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL4, mask, val);
break;
case 2:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL5, mask, val);
break;
}
@@ -257,12 +305,17 @@ find_rate:
return 0;
}
-static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
+static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val)
{
- int id = rsnd_mod_id(mod);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
+ int id = rsnd_mod_id(ssi_mod);
int shift = (id % 4) * 8;
u32 mask = 0xFF << shift;
+ rsnd_mod_confirm_ssi(ssi_mod);
+
val = val << shift;
/*
@@ -274,13 +327,13 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
switch (id / 4) {
case 0:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL0, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL0, mask, val);
break;
case 1:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL1, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL1, mask, val);
break;
case 2:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL2, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val);
break;
}
}
@@ -326,14 +379,14 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
}
/*
- * find 1/6 clock from BRGA/BRGB
+ * find divided clock from BRGA/BRGB
*/
- if (rate == adg->rbga_rate_for_441khz_div_6) {
+ if (rate == adg->rbga_rate_for_441khz) {
data = 0x10;
goto found_clock;
}
- if (rate == adg->rbgb_rate_for_48khz_div_6) {
+ if (rate == adg->rbgb_rate_for_48khz) {
data = 0x20;
goto found_clock;
}
@@ -342,29 +395,60 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
found_clock:
- /* see rsnd_adg_ssi_clk_init() */
- rsnd_mod_bset(mod, SSICKR, 0x00FF0000, adg->ckr);
- rsnd_mod_write(mod, BRRA, 0x00000002); /* 1/6 */
- rsnd_mod_write(mod, BRRB, 0x00000002); /* 1/6 */
-
/*
* This "mod" = "ssi" here.
* we can get "ssi id" from mod
*/
rsnd_adg_set_ssi_clk(mod, data);
- dev_dbg(dev, "ADG: ssi%d selects clk%d = %d",
- rsnd_mod_id(mod), i, rate);
+ dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod),
+ data, rate);
return 0;
}
-static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
+static void rsnd_adg_get_clkin(struct rsnd_priv *priv,
+ struct rsnd_adg *adg)
{
+ struct device *dev = rsnd_priv_to_dev(priv);
struct clk *clk;
- unsigned long rate;
- u32 ckr;
+ static const char * const clk_name[] = {
+ [CLKA] = "clk_a",
+ [CLKB] = "clk_b",
+ [CLKC] = "clk_c",
+ [CLKI] = "clk_i",
+ };
int i;
+
+ for (i = 0; i < CLKMAX; i++) {
+ clk = devm_clk_get(dev, clk_name[i]);
+ adg->clk[i] = IS_ERR(clk) ? NULL : clk;
+ }
+
+ for_each_rsnd_clk(clk, adg, i)
+ dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+}
+
+static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
+ struct rsnd_adg *adg)
+{
+ struct clk *clk;
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct device_node *np = dev->of_node;
+ u32 ckr, rbgx, rbga, rbgb;
+ u32 rate, req_rate, div;
+ uint32_t count = 0;
+ unsigned long req_48kHz_rate, req_441kHz_rate;
+ int i;
+ const char *parent_clk_name = NULL;
+ static const char * const clkout_name[] = {
+ [CLKOUT] = "audio_clkout",
+ [CLKOUT1] = "audio_clkout1",
+ [CLKOUT2] = "audio_clkout2",
+ [CLKOUT3] = "audio_clkout3",
+ };
int brg_table[] = {
[CLKA] = 0x0,
[CLKB] = 0x1,
@@ -372,19 +456,34 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
[CLKI] = 0x2,
};
+ of_property_read_u32(np, "#clock-cells", &count);
+
+ /*
+ * ADG supports BRRA/BRRB output only
+ * this means all clkout0/1/2/3 will be same rate
+ */
+ of_property_read_u32(np, "clock-frequency", &req_rate);
+ req_48kHz_rate = 0;
+ req_441kHz_rate = 0;
+ if (0 == (req_rate % 44100))
+ req_441kHz_rate = req_rate;
+ if (0 == (req_rate % 48000))
+ req_48kHz_rate = req_rate;
+
/*
* This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC
* have 44.1kHz or 48kHz base clocks for now.
*
* SSI itself can divide parent clock by 1/1 - 1/16
- * So, BRGA outputs 44.1kHz base parent clock 1/32,
- * and, BRGB outputs 48.0kHz base parent clock 1/32 here.
* see
* rsnd_adg_ssi_clk_try_start()
+ * rsnd_ssi_master_clk_start()
*/
ckr = 0;
- adg->rbga_rate_for_441khz_div_6 = 0;
- adg->rbgb_rate_for_48khz_div_6 = 0;
+ rbga = 2; /* default 1/6 */
+ rbgb = 2; /* default 1/6 */
+ adg->rbga_rate_for_441khz = 0;
+ adg->rbgb_rate_for_48khz = 0;
for_each_rsnd_clk(clk, adg, i) {
rate = clk_get_rate(clk);
@@ -392,19 +491,86 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
continue;
/* RBGA */
- if (!adg->rbga_rate_for_441khz_div_6 && (0 == rate % 44100)) {
- adg->rbga_rate_for_441khz_div_6 = rate / 6;
- ckr |= brg_table[i] << 20;
+ if (!adg->rbga_rate_for_441khz && (0 == rate % 44100)) {
+ div = 6;
+ if (req_441kHz_rate)
+ div = rate / req_441kHz_rate;
+ rbgx = rsnd_adg_calculate_rbgx(div);
+ if (BRRx_MASK(rbgx) == rbgx) {
+ rbga = rbgx;
+ adg->rbga_rate_for_441khz = rate / div;
+ ckr |= brg_table[i] << 20;
+ if (req_441kHz_rate)
+ parent_clk_name = __clk_get_name(clk);
+ }
}
/* RBGB */
- if (!adg->rbgb_rate_for_48khz_div_6 && (0 == rate % 48000)) {
- adg->rbgb_rate_for_48khz_div_6 = rate / 6;
- ckr |= brg_table[i] << 16;
+ if (!adg->rbgb_rate_for_48khz && (0 == rate % 48000)) {
+ div = 6;
+ if (req_48kHz_rate)
+ div = rate / req_48kHz_rate;
+ rbgx = rsnd_adg_calculate_rbgx(div);
+ if (BRRx_MASK(rbgx) == rbgx) {
+ rbgb = rbgx;
+ adg->rbgb_rate_for_48khz = rate / div;
+ ckr |= brg_table[i] << 16;
+ if (req_48kHz_rate) {
+ parent_clk_name = __clk_get_name(clk);
+ ckr |= 0x80000000;
+ }
+ }
+ }
+ }
+
+ /*
+ * ADG supports BRRA/BRRB output only.
+ * this means all clkout0/1/2/3 will be * same rate
+ */
+
+ /*
+ * for clkout
+ */
+ if (!count) {
+ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT],
+ parent_clk_name,
+ (parent_clk_name) ?
+ 0 : CLK_IS_ROOT, req_rate);
+ if (!IS_ERR(clk)) {
+ adg->clkout[CLKOUT] = clk;
+ of_clk_add_provider(np, of_clk_src_simple_get, clk);
+ }
+ }
+ /*
+ * for clkout0/1/2/3
+ */
+ else {
+ for (i = 0; i < CLKOUTMAX; i++) {
+ clk = clk_register_fixed_rate(dev, clkout_name[i],
+ parent_clk_name,
+ (parent_clk_name) ?
+ 0 : CLK_IS_ROOT,
+ req_rate);
+ if (!IS_ERR(clk)) {
+ adg->onecell.clks = adg->clkout;
+ adg->onecell.clk_num = CLKOUTMAX;
+
+ adg->clkout[i] = clk;
+
+ of_clk_add_provider(np, of_clk_src_onecell_get,
+ &adg->onecell);
+ }
}
}
- adg->ckr = ckr;
+ rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr);
+ rsnd_mod_write(adg_mod, BRRA, rbga);
+ rsnd_mod_write(adg_mod, BRRB, rbgb);
+
+ for_each_rsnd_clkout(clk, adg, i)
+ dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+ dev_dbg(dev, "SSICKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n",
+ ckr, rbga, rbgb);
}
int rsnd_adg_probe(struct platform_device *pdev,
@@ -413,8 +579,6 @@ int rsnd_adg_probe(struct platform_device *pdev,
{
struct rsnd_adg *adg;
struct device *dev = rsnd_priv_to_dev(priv);
- struct clk *clk;
- int i;
adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL);
if (!adg) {
@@ -422,15 +586,16 @@ int rsnd_adg_probe(struct platform_device *pdev,
return -ENOMEM;
}
- adg->clk[CLKA] = devm_clk_get(dev, "clk_a");
- adg->clk[CLKB] = devm_clk_get(dev, "clk_b");
- adg->clk[CLKC] = devm_clk_get(dev, "clk_c");
- adg->clk[CLKI] = devm_clk_get(dev, "clk_i");
-
- for_each_rsnd_clk(clk, adg, i)
- dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+ /*
+ * ADG is special module.
+ * Use ADG mod without rsnd_mod_init() to make debug easy
+ * for rsnd_write/rsnd_read
+ */
+ adg->mod.ops = &adg_ops;
+ adg->mod.priv = priv;
- rsnd_adg_ssi_clk_init(priv, adg);
+ rsnd_adg_get_clkin(priv, adg);
+ rsnd_adg_get_clkout(priv, adg);
priv->adg = adg;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index f3feed5ce9b6..eec294da81e3 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -110,6 +110,7 @@ static const struct rsnd_of_data rsnd_of_data_gen2 = {
static const struct of_device_id rsnd_of_match[] = {
{ .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 },
{ .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 },
+ { .compatible = "renesas,rcar_sound-gen3", .data = &rsnd_of_data_gen2 }, /* gen2 compatible */
{},
};
MODULE_DEVICE_TABLE(of, rsnd_of_match);
@@ -126,6 +127,17 @@ MODULE_DEVICE_TABLE(of, rsnd_of_match);
#define rsnd_info_id(priv, io, name) \
((io)->info->name - priv->info->name##_info)
+void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type)
+{
+ if (mod->type != type) {
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ dev_warn(dev, "%s[%d] is not your expected module\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
+ }
+}
+
/*
* rsnd_mod functions
*/
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 05498bba5874..a3e7c716e1f7 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -66,7 +66,7 @@ struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_ctu_nr(priv)))
id = 0;
- return &((struct rsnd_ctu *)(priv->ctu) + id)->mod;
+ return rsnd_mod_get((struct rsnd_ctu *)(priv->ctu) + id);
}
static void rsnd_of_parse_ctu(struct platform_device *pdev,
@@ -150,7 +150,7 @@ int rsnd_ctu_probe(struct platform_device *pdev,
ctu->info = &info->ctu_info[i];
- ret = rsnd_mod_init(priv, &ctu->mod, &rsnd_ctu_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops,
clk, RSND_MOD_CTU, i);
if (ret)
return ret;
@@ -166,6 +166,6 @@ void rsnd_ctu_remove(struct platform_device *pdev,
int i;
for_each_rsnd_ctu(ctu, priv, i) {
- rsnd_mod_quit(&ctu->mod);
+ rsnd_mod_quit(rsnd_mod_get(ctu));
}
}
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index 57796387d482..8d8eee6350c9 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -282,7 +282,7 @@ struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_dvc_nr(priv)))
id = 0;
- return &((struct rsnd_dvc *)(priv->dvc) + id)->mod;
+ return rsnd_mod_get((struct rsnd_dvc *)(priv->dvc) + id);
}
static void rsnd_of_parse_dvc(struct platform_device *pdev,
@@ -361,7 +361,7 @@ int rsnd_dvc_probe(struct platform_device *pdev,
dvc->info = &info->dvc_info[i];
- ret = rsnd_mod_init(priv, &dvc->mod, &rsnd_dvc_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops,
clk, RSND_MOD_DVC, i);
if (ret)
return ret;
@@ -377,6 +377,6 @@ void rsnd_dvc_remove(struct platform_device *pdev,
int i;
for_each_rsnd_dvc(dvc, priv, i) {
- rsnd_mod_quit(&dvc->mod);
+ rsnd_mod_quit(rsnd_mod_get(dvc));
}
}
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 0d5c102db6f5..8544403ffb26 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -99,7 +99,7 @@ struct rsnd_mod *rsnd_mix_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_mix_nr(priv)))
id = 0;
- return &((struct rsnd_mix *)(priv->mix) + id)->mod;
+ return rsnd_mod_get((struct rsnd_mix *)(priv->mix) + id);
}
static void rsnd_of_parse_mix(struct platform_device *pdev,
@@ -179,7 +179,7 @@ int rsnd_mix_probe(struct platform_device *pdev,
mix->info = &info->mix_info[i];
- ret = rsnd_mod_init(priv, &mix->mod, &rsnd_mix_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops,
clk, RSND_MOD_MIX, i);
if (ret)
return ret;
@@ -195,6 +195,6 @@ void rsnd_mix_remove(struct platform_device *pdev,
int i;
for_each_rsnd_mix(mix, priv, i) {
- rsnd_mod_quit(&mix->mod);
+ rsnd_mod_quit(rsnd_mod_get(mix));
}
}
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 7a0e52b4640a..e4068d78616c 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -214,6 +214,7 @@ struct rsnd_dma {
};
#define rsnd_dma_to_dmaen(dma) (&(dma)->dma.en)
#define rsnd_dma_to_dmapp(dma) (&(dma)->dma.pp)
+#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
void rsnd_dma_start(struct rsnd_dai_stream *io, struct rsnd_dma *dma);
void rsnd_dma_stop(struct rsnd_dai_stream *io, struct rsnd_dma *dma);
@@ -225,8 +226,6 @@ int rsnd_dma_probe(struct platform_device *pdev,
struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node,
struct rsnd_mod *mod, char *name);
-#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
-
/*
* R-Car sound mod
*/
@@ -332,6 +331,7 @@ struct rsnd_mod {
#define rsnd_mod_id(mod) ((mod) ? (mod)->id : -1)
#define rsnd_mod_hw_start(mod) clk_enable((mod)->clk)
#define rsnd_mod_hw_stop(mod) clk_disable((mod)->clk)
+#define rsnd_mod_get(ip) (&(ip)->mod)
int rsnd_mod_init(struct rsnd_priv *priv,
struct rsnd_mod *mod,
@@ -627,4 +627,15 @@ void rsnd_dvc_remove(struct platform_device *pdev,
struct rsnd_priv *priv);
struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id);
+#ifdef DEBUG
+void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type);
+#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI)
+#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC)
+#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC)
+#else
+#define rsnd_mod_confirm_ssi(mssi)
+#define rsnd_mod_confirm_src(msrc)
+#define rsnd_mod_confirm_dvc(mdvc)
+#endif
+
#endif
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 89a18e102feb..ca7a20f03c9b 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -918,11 +918,10 @@ static void rsnd_src_reconvert_update(struct rsnd_dai_stream *io,
rsnd_mod_write(mod, SRC_IFSVR, fsrate);
}
-static int rsnd_src_pcm_new(struct rsnd_mod *mod,
+static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct snd_soc_pcm_runtime *rtd)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct rsnd_src *src = rsnd_mod_to_src(mod);
int ret;
@@ -932,12 +931,6 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod,
*/
/*
- * Gen1 is not supported
- */
- if (rsnd_is_gen1(priv))
- return 0;
-
- /*
* SRC sync convert needs clock master
*/
if (!rsnd_rdai_is_clk_master(rdai))
@@ -975,7 +968,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = {
.start = rsnd_src_start_gen2,
.stop = rsnd_src_stop_gen2,
.hw_params = rsnd_src_hw_params,
- .pcm_new = rsnd_src_pcm_new,
+ .pcm_new = rsnd_src_pcm_new_gen2,
};
struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
@@ -983,7 +976,7 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_src_nr(priv)))
id = 0;
- return &((struct rsnd_src *)(priv->src) + id)->mod;
+ return rsnd_mod_get((struct rsnd_src *)(priv->src) + id);
}
static void rsnd_of_parse_src(struct platform_device *pdev,
@@ -1078,7 +1071,7 @@ int rsnd_src_probe(struct platform_device *pdev,
src->info = &info->src_info[i];
- ret = rsnd_mod_init(priv, &src->mod, ops, clk, RSND_MOD_SRC, i);
+ ret = rsnd_mod_init(priv, rsnd_mod_get(src), ops, clk, RSND_MOD_SRC, i);
if (ret)
return ret;
}
@@ -1093,6 +1086,6 @@ void rsnd_src_remove(struct platform_device *pdev,
int i;
for_each_rsnd_src(src, priv, i) {
- rsnd_mod_quit(&src->mod);
+ rsnd_mod_quit(rsnd_mod_get(src));
}
}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index d45b9a7e324e..5e05f9422073 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -128,10 +128,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
struct rsnd_priv *priv = rsnd_io_to_priv(io);
struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
struct device *dev = rsnd_priv_to_dev(priv);
- int i, j, ret;
- int adg_clk_div_table[] = {
- 1, 6, /* see adg.c */
- };
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+ int j, ret;
int ssi_clk_mul_table[] = {
1, 2, 4, 8, 16, 6, 12,
};
@@ -141,28 +139,25 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
/*
* Find best clock, and try to start ADG
*/
- for (i = 0; i < ARRAY_SIZE(adg_clk_div_table); i++) {
- for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
-
- /*
- * this driver is assuming that
- * system word is 64fs (= 2 x 32bit)
- * see rsnd_ssi_init()
- */
- main_rate = rate / adg_clk_div_table[i]
- * 32 * 2 * ssi_clk_mul_table[j];
-
- ret = rsnd_adg_ssi_clk_try_start(&ssi->mod, main_rate);
- if (0 == ret) {
- ssi->cr_clk = FORCE | SWL_32 |
- SCKD | SWSD | CKDV(j);
-
- dev_dbg(dev, "%s[%d] outputs %u Hz\n",
- rsnd_mod_name(&ssi->mod),
- rsnd_mod_id(&ssi->mod), rate);
-
- return 0;
- }
+ for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
+
+ /*
+ * this driver is assuming that
+ * system word is 64fs (= 2 x 32bit)
+ * see rsnd_ssi_init()
+ */
+ main_rate = rate * 32 * 2 * ssi_clk_mul_table[j];
+
+ ret = rsnd_adg_ssi_clk_try_start(mod, main_rate);
+ if (0 == ret) {
+ ssi->cr_clk = FORCE | SWL_32 |
+ SCKD | SWSD | CKDV(j);
+
+ dev_dbg(dev, "%s[%d] outputs %u Hz\n",
+ rsnd_mod_name(mod),
+ rsnd_mod_id(mod), rate);
+
+ return 0;
}
}
@@ -172,8 +167,10 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi)
{
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
ssi->cr_clk = 0;
- rsnd_adg_ssi_clk_stop(&ssi->mod);
+ rsnd_adg_ssi_clk_stop(mod);
}
static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
@@ -182,11 +179,12 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
struct rsnd_priv *priv = rsnd_io_to_priv(io);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
u32 cr_mode;
u32 cr;
if (0 == ssi->usrcnt) {
- rsnd_mod_hw_start(&ssi->mod);
+ rsnd_mod_hw_start(mod);
if (rsnd_rdai_is_clk_master(rdai)) {
struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi);
@@ -198,7 +196,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
}
}
- if (rsnd_ssi_is_dma_mode(&ssi->mod)) {
+ if (rsnd_ssi_is_dma_mode(mod)) {
cr_mode = UIEN | OIEN | /* over/under run */
DMEN; /* DMA : enable DMA */
} else {
@@ -210,24 +208,25 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
cr_mode |
EN;
- rsnd_mod_write(&ssi->mod, SSICR, cr);
+ rsnd_mod_write(mod, SSICR, cr);
/* enable WS continue */
if (rsnd_rdai_is_clk_master(rdai))
- rsnd_mod_write(&ssi->mod, SSIWSR, CONT);
+ rsnd_mod_write(mod, SSIWSR, CONT);
/* clear error status */
- rsnd_mod_write(&ssi->mod, SSISR, 0);
+ rsnd_mod_write(mod, SSISR, 0);
ssi->usrcnt++;
dev_dbg(dev, "%s[%d] hw started\n",
- rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod));
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
}
static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct device *dev = rsnd_priv_to_dev(priv);
u32 cr;
@@ -247,15 +246,15 @@ static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
cr = ssi->cr_own |
ssi->cr_clk;
- rsnd_mod_write(&ssi->mod, SSICR, cr | EN);
- rsnd_ssi_status_check(&ssi->mod, DIRQ);
+ rsnd_mod_write(mod, SSICR, cr | EN);
+ rsnd_ssi_status_check(mod, DIRQ);
/*
* disable SSI,
* and, wait idle state
*/
- rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */
- rsnd_ssi_status_check(&ssi->mod, IIRQ);
+ rsnd_mod_write(mod, SSICR, cr); /* disabled all */
+ rsnd_ssi_status_check(mod, IIRQ);
if (rsnd_rdai_is_clk_master(rdai)) {
struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi);
@@ -266,13 +265,13 @@ static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
rsnd_ssi_master_clk_stop(ssi);
}
- rsnd_mod_hw_stop(&ssi->mod);
+ rsnd_mod_hw_stop(mod);
ssi->chan = 0;
}
dev_dbg(dev, "%s[%d] hw stopped\n",
- rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod));
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
}
/*
@@ -371,7 +370,7 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
/* It will be removed on rsnd_ssi_hw_stop */
ssi->chan = chan;
if (ssi_parent)
- return rsnd_ssi_hw_params(&ssi_parent->mod, io,
+ return rsnd_ssi_hw_params(rsnd_mod_get(ssi_parent), io,
substream, params);
return 0;
@@ -379,12 +378,14 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status)
{
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
/* under/over flow error */
if (status & (UIRQ | OIRQ)) {
ssi->err++;
/* clear error status */
- rsnd_mod_write(&ssi->mod, SSISR, 0);
+ rsnd_mod_write(mod, SSISR, 0);
}
}
@@ -656,7 +657,7 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv)))
id = 0;
- return &((struct rsnd_ssi *)(priv->ssi) + id)->mod;
+ return rsnd_mod_get((struct rsnd_ssi *)(priv->ssi) + id);
}
int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
@@ -668,10 +669,12 @@ int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
static void rsnd_ssi_parent_setup(struct rsnd_priv *priv, struct rsnd_ssi *ssi)
{
- if (!rsnd_ssi_is_pin_sharing(&ssi->mod))
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
+ if (!rsnd_ssi_is_pin_sharing(mod))
return;
- switch (rsnd_mod_id(&ssi->mod)) {
+ switch (rsnd_mod_id(mod)) {
case 1:
case 2:
ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 0));
@@ -794,7 +797,8 @@ int rsnd_ssi_probe(struct platform_device *pdev,
else if (rsnd_ssi_pio_available(ssi))
ops = &rsnd_ssi_pio_ops;
- ret = rsnd_mod_init(priv, &ssi->mod, ops, clk, RSND_MOD_SSI, i);
+ ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk,
+ RSND_MOD_SSI, i);
if (ret)
return ret;
@@ -811,6 +815,6 @@ void rsnd_ssi_remove(struct platform_device *pdev,
int i;
for_each_rsnd_ssi(ssi, priv, i) {
- rsnd_mod_quit(&ssi->mod);
+ rsnd_mod_quit(rsnd_mod_get(ssi));
}
}
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index abb0d956231c..76b2ab8c2b4a 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -738,7 +738,7 @@ static int siu_probe(struct platform_device *pdev)
struct siu_info *info;
int ret;
- info = kmalloc(sizeof(*info), GFP_KERNEL);
+ info = devm_kmalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
if (!info)
return -ENOMEM;
siu_i2s_data = info;
@@ -746,7 +746,7 @@ static int siu_probe(struct platform_device *pdev)
ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev);
if (ret)
- goto ereqfw;
+ return ret;
/*
* Loaded firmware is "const" - read only, but we have to modify it in
@@ -757,89 +757,52 @@ static int siu_probe(struct platform_device *pdev)
release_firmware(fw_entry);
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- ret = -ENODEV;
- goto egetres;
- }
+ if (!res)
+ return -ENODEV;
- region = request_mem_region(res->start, resource_size(res),
- pdev->name);
+ region = devm_request_mem_region(&pdev->dev, res->start,
+ resource_size(res), pdev->name);
if (!region) {
dev_err(&pdev->dev, "SIU region already claimed\n");
- ret = -EBUSY;
- goto ereqmemreg;
+ return -EBUSY;
}
- ret = -ENOMEM;
- info->pram = ioremap(res->start, PRAM_SIZE);
+ info->pram = devm_ioremap(&pdev->dev, res->start, PRAM_SIZE);
if (!info->pram)
- goto emappram;
- info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE);
+ return -ENOMEM;
+ info->xram = devm_ioremap(&pdev->dev, res->start + XRAM_OFFSET,
+ XRAM_SIZE);
if (!info->xram)
- goto emapxram;
- info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE);
+ return -ENOMEM;
+ info->yram = devm_ioremap(&pdev->dev, res->start + YRAM_OFFSET,
+ YRAM_SIZE);
if (!info->yram)
- goto emapyram;
- info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) -
- REG_OFFSET);
+ return -ENOMEM;
+ info->reg = devm_ioremap(&pdev->dev, res->start + REG_OFFSET,
+ resource_size(res) - REG_OFFSET);
if (!info->reg)
- goto emapreg;
+ return -ENOMEM;
dev_set_drvdata(&pdev->dev, info);
/* register using ARRAY version so we can keep dai name */
- ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component,
- &siu_i2s_dai, 1);
+ ret = devm_snd_soc_register_component(&pdev->dev, &siu_i2s_component,
+ &siu_i2s_dai, 1);
if (ret < 0)
- goto edaiinit;
+ return ret;
- ret = snd_soc_register_platform(&pdev->dev, &siu_platform);
+ ret = devm_snd_soc_register_platform(&pdev->dev, &siu_platform);
if (ret < 0)
- goto esocregp;
+ return ret;
pm_runtime_enable(&pdev->dev);
- return ret;
-
-esocregp:
- snd_soc_unregister_component(&pdev->dev);
-edaiinit:
- iounmap(info->reg);
-emapreg:
- iounmap(info->yram);
-emapyram:
- iounmap(info->xram);
-emapxram:
- iounmap(info->pram);
-emappram:
- release_mem_region(res->start, resource_size(res));
-ereqmemreg:
-egetres:
-ereqfw:
- kfree(info);
-
- return ret;
+ return 0;
}
static int siu_remove(struct platform_device *pdev)
{
- struct siu_info *info = dev_get_drvdata(&pdev->dev);
- struct resource *res;
-
pm_runtime_disable(&pdev->dev);
-
- snd_soc_unregister_platform(&pdev->dev);
- snd_soc_unregister_component(&pdev->dev);
-
- iounmap(info->reg);
- iounmap(info->yram);
- iounmap(info->xram);
- iounmap(info->pram);
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (res)
- release_mem_region(res->start, resource_size(res));
- kfree(info);
-
return 0;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6173d15236c3..3b471f9c98c6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3291,13 +3291,38 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
+static int snd_soc_of_get_slot_mask(struct device_node *np,
+ const char *prop_name,
+ unsigned int *mask)
+{
+ u32 val;
+ const __be32 *of_slot_mask = of_get_property(np, prop_name, &val);
+ int i;
+
+ if (!of_slot_mask)
+ return 0;
+ val /= sizeof(u32);
+ for (i = 0; i < val; i++)
+ if (be32_to_cpup(&of_slot_mask[i]))
+ *mask |= (1 << i);
+
+ return val;
+}
+
int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *tx_mask,
+ unsigned int *rx_mask,
unsigned int *slots,
unsigned int *slot_width)
{
u32 val;
int ret;
+ if (tx_mask)
+ snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", tx_mask);
+ if (rx_mask)
+ snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", rx_mask);
+
if (of_property_read_bool(np, "dai-tdm-slot-num")) {
ret = of_property_read_u32(np, "dai-tdm-slot-num", &val);
if (ret)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 70e4b9d8bdcd..317395824cd7 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -34,6 +34,24 @@
#define DPCM_MAX_BE_USERS 8
+/*
+ * snd_soc_dai_stream_valid() - check if a DAI supports the given stream
+ *
+ * Returns true if the DAI supports the indicated stream type.
+ */
+static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_pcm_stream *codec_stream;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ codec_stream = &dai->driver->playback;
+ else
+ codec_stream = &dai->driver->capture;
+
+ /* If the codec specifies any rate at all, it supports the stream. */
+ return codec_stream->rates;
+}
+
/**
* snd_soc_runtime_activate() - Increment active count for PCM runtime components
* @rtd: ASoC PCM runtime that is activated
@@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
/* first calculate min/max only for CODECs in the DAI link */
for (i = 0; i < rtd->num_codecs; i++) {
+
+ /*
+ * Skip CODECs which don't support the current stream type.
+ * Otherwise, since the rate, channel, and format values will
+ * zero in that case, we would have no usable settings left,
+ * causing the resulting setup to fail.
+ * At least one CODEC should match, otherwise we should have
+ * bailed out on a higher level, since there would be no
+ * CODEC to support the transfer direction in that case.
+ */
+ if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
+ substream->stream))
+ continue;
+
codec_dai_drv = rtd->codec_dais[i]->driver;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
codec_stream = &codec_dai_drv->playback;
@@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
struct snd_pcm_hw_params codec_params;
+ /*
+ * Skip CODECs which don't support the current stream type,
+ * the idea being that if a CODEC is not used for the currently
+ * set up transfer direction, it should not need to be
+ * configured, especially since the configuration used might
+ * not even be supported by that CODEC. There may be cases
+ * however where a CODEC needs to be set up although it is
+ * actually not being used for the transfer, e.g. if a
+ * capture-only CODEC is acting as an LRCLK and/or BCLK master
+ * for the DAI link including a playback-only CODEC.
+ * If this becomes necessary, we will have to augment the
+ * machine driver setup with information on how to act, so
+ * we can do the right thing here.
+ */
+ if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
+ continue;
+
/* copy params for each codec */
codec_params = *params;
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
new file mode 100644
index 000000000000..84c72ec6ad73
--- /dev/null
+++ b/sound/soc/sunxi/Kconfig
@@ -0,0 +1,11 @@
+menu "Allwinner SoC Audio support"
+
+config SND_SUN4I_CODEC
+ tristate "Allwinner A10 Codec Support"
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ select REGMAP_MMIO
+ help
+ Select Y or M to add support for the Codec embedded in the Allwinner
+ A10 and affiliated SoCs.
+
+endmenu
diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile
new file mode 100644
index 000000000000..ea8a08c881d6
--- /dev/null
+++ b/sound/soc/sunxi/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
+
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
new file mode 100644
index 000000000000..8d59d83b5aa4
--- /dev/null
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -0,0 +1,719 @@
+/*
+ * Copyright 2014 Emilio López <emilio@elopez.com.ar>
+ * Copyright 2014 Jon Smirl <jonsmirl@gmail.com>
+ * Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com>
+ *
+ * Based on the Allwinner SDK driver, released under the GPL.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/of_address.h>
+#include <linux/clk.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/dmaengine_pcm.h>
+
+/* Codec DAC register offsets and bit fields */
+#define SUN4I_CODEC_DAC_DPC (0x00)
+#define SUN4I_CODEC_DAC_DPC_EN_DA (31)
+#define SUN4I_CODEC_DAC_DPC_DVOL (12)
+#define SUN4I_CODEC_DAC_FIFOC (0x04)
+#define SUN4I_CODEC_DAC_FIFOC_DAC_FS (29)
+#define SUN4I_CODEC_DAC_FIFOC_FIR_VERSION (28)
+#define SUN4I_CODEC_DAC_FIFOC_SEND_LASAT (26)
+#define SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE (24)
+#define SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT (21)
+#define SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL (8)
+#define SUN4I_CODEC_DAC_FIFOC_MONO_EN (6)
+#define SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS (5)
+#define SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN (4)
+#define SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH (0)
+#define SUN4I_CODEC_DAC_FIFOS (0x08)
+#define SUN4I_CODEC_DAC_TXDATA (0x0c)
+#define SUN4I_CODEC_DAC_ACTL (0x10)
+#define SUN4I_CODEC_DAC_ACTL_DACAENR (31)
+#define SUN4I_CODEC_DAC_ACTL_DACAENL (30)
+#define SUN4I_CODEC_DAC_ACTL_MIXEN (29)
+#define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15)
+#define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14)
+#define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13)
+#define SUN4I_CODEC_DAC_ACTL_DACPAS (8)
+#define SUN4I_CODEC_DAC_ACTL_MIXPAS (7)
+#define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6)
+#define SUN4I_CODEC_DAC_ACTL_PA_VOL (0)
+#define SUN4I_CODEC_DAC_TUNE (0x14)
+#define SUN4I_CODEC_DAC_DEBUG (0x18)
+
+/* Codec ADC register offsets and bit fields */
+#define SUN4I_CODEC_ADC_FIFOC (0x1c)
+#define SUN4I_CODEC_ADC_FIFOC_EN_AD (28)
+#define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE (24)
+#define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL (8)
+#define SUN4I_CODEC_ADC_FIFOC_MONO_EN (7)
+#define SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS (6)
+#define SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN (4)
+#define SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH (0)
+#define SUN4I_CODEC_ADC_FIFOS (0x20)
+#define SUN4I_CODEC_ADC_RXDATA (0x24)
+#define SUN4I_CODEC_ADC_ACTL (0x28)
+#define SUN4I_CODEC_ADC_ACTL_ADC_R_EN (31)
+#define SUN4I_CODEC_ADC_ACTL_ADC_L_EN (30)
+#define SUN4I_CODEC_ADC_ACTL_PREG1EN (29)
+#define SUN4I_CODEC_ADC_ACTL_PREG2EN (28)
+#define SUN4I_CODEC_ADC_ACTL_VMICEN (27)
+#define SUN4I_CODEC_ADC_ACTL_VADCG (20)
+#define SUN4I_CODEC_ADC_ACTL_ADCIS (17)
+#define SUN4I_CODEC_ADC_ACTL_PA_EN (4)
+#define SUN4I_CODEC_ADC_ACTL_DDE (3)
+#define SUN4I_CODEC_ADC_DEBUG (0x2c)
+
+/* Other various ADC registers */
+#define SUN4I_CODEC_DAC_TXCNT (0x30)
+#define SUN4I_CODEC_ADC_RXCNT (0x34)
+#define SUN4I_CODEC_AC_SYS_VERI (0x38)
+#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c)
+
+struct sun4i_codec {
+ struct device *dev;
+ struct regmap *regmap;
+ struct clk *clk_apb;
+ struct clk *clk_module;
+
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
+};
+
+static void sun4i_codec_start_playback(struct sun4i_codec *scodec)
+{
+ /*
+ * FIXME: according to the BSP, we might need to drive a PA
+ * GPIO high here on some boards
+ */
+
+ /* Flush TX FIFO */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH),
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH));
+
+ /* Enable DAC DRQ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN),
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN));
+}
+
+static void sun4i_codec_stop_playback(struct sun4i_codec *scodec)
+{
+ /*
+ * FIXME: according to the BSP, we might need to drive a PA
+ * GPIO low here on some boards
+ */
+
+ /* Disable DAC DRQ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN),
+ 0);
+}
+
+static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ sun4i_codec_start_playback(scodec);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sun4i_codec_stop_playback(scodec);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int sun4i_codec_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+ u32 val;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ /* Flush the TX FIFO */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH),
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH));
+
+ /* Set TX FIFO Empty Trigger Level */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 0x3f << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL,
+ 0xf << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL);
+
+ if (substream->runtime->rate > 32000)
+ /* Use 64 bits FIR filter */
+ val = 0;
+ else
+ /* Use 32 bits FIR filter */
+ val = BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION);
+
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION),
+ val);
+
+ /* Send zeros when we have an underrun */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT),
+ 0);
+
+ return 0;
+}
+
+static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params)
+{
+ unsigned int rate = params_rate(params);
+
+ switch (rate) {
+ case 176400:
+ case 88200:
+ case 44100:
+ case 33075:
+ case 22050:
+ case 14700:
+ case 11025:
+ case 7350:
+ return 22579200;
+
+ case 192000:
+ case 96000:
+ case 48000:
+ case 32000:
+ case 24000:
+ case 16000:
+ case 12000:
+ case 8000:
+ return 24576000;
+
+ default:
+ return 0;
+ }
+}
+
+static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params)
+{
+ unsigned int rate = params_rate(params);
+
+ switch (rate) {
+ case 192000:
+ case 176400:
+ return 6;
+
+ case 96000:
+ case 88200:
+ return 7;
+
+ case 48000:
+ case 44100:
+ return 0;
+
+ case 32000:
+ case 33075:
+ return 1;
+
+ case 24000:
+ case 22050:
+ return 2;
+
+ case 16000:
+ case 14700:
+ return 3;
+
+ case 12000:
+ case 11025:
+ return 4;
+
+ case 8000:
+ case 7350:
+ return 5;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static int sun4i_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+ unsigned long clk_freq;
+ int hwrate;
+ u32 val;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ clk_freq = sun4i_codec_get_mod_freq(params);
+ if (!clk_freq)
+ return -EINVAL;
+
+ if (clk_set_rate(scodec->clk_module, clk_freq))
+ return -EINVAL;
+
+ hwrate = sun4i_codec_get_hw_rate(params);
+ if (hwrate < 0)
+ return hwrate;
+
+ /* Set DAC sample rate */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 7 << SUN4I_CODEC_DAC_FIFOC_DAC_FS,
+ hwrate << SUN4I_CODEC_DAC_FIFOC_DAC_FS);
+
+ /* Set the number of channels we want to use */
+ if (params_channels(params) == 1)
+ val = BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN);
+ else
+ val = 0;
+
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN),
+ val);
+
+ /* Set the number of sample bits to either 16 or 24 bits */
+ if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) {
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS),
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS));
+
+ /* Set TX FIFO mode to padding the LSBs with 0 */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE),
+ 0);
+
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ } else {
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS),
+ 0);
+
+ /* Set TX FIFO mode to repeat the MSB */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE),
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE));
+
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ }
+
+ return 0;
+}
+
+static int sun4i_codec_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ /*
+ * Stop issuing DRQ when we have room for less than 16 samples
+ * in our TX FIFO
+ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT,
+ 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT);
+
+ return clk_prepare_enable(scodec->clk_module);
+}
+
+static void sun4i_codec_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ clk_disable_unprepare(scodec->clk_module);
+}
+
+static const struct snd_soc_dai_ops sun4i_codec_dai_ops = {
+ .startup = sun4i_codec_startup,
+ .shutdown = sun4i_codec_shutdown,
+ .trigger = sun4i_codec_trigger,
+ .hw_params = sun4i_codec_hw_params,
+ .prepare = sun4i_codec_prepare,
+};
+
+static struct snd_soc_dai_driver sun4i_codec_dai = {
+ .name = "Codec",
+ .ops = &sun4i_codec_dai_ops,
+ .playback = {
+ .stream_name = "Codec Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .rates = SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000 |
+ SNDRV_PCM_RATE_KNOT,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .sig_bits = 24,
+ },
+};
+
+/*** Codec ***/
+static const struct snd_kcontrol_new sun4i_codec_pa_mute =
+ SOC_DAPM_SINGLE("Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0);
+
+static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1);
+
+static const struct snd_kcontrol_new sun4i_codec_widgets[] = {
+ SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0,
+ sun4i_codec_pa_volume_scale),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Right DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACPAS, 1, 0),
+ SOC_DAPM_SINGLE("Mixer Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = {
+ /* Digital parts of the DACs */
+ SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC,
+ SUN4I_CODEC_DAC_DPC_EN_DA, 0,
+ NULL, 0),
+
+ /* Analog parts of the DACs */
+ SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACAENL, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACAENR, 0),
+
+ /* Mixers */
+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+ sun4i_codec_left_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+ sun4i_codec_right_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_right_mixer_controls)),
+
+ /* Global Mixer Enable */
+ SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0),
+
+ /* Pre-Amplifier */
+ SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL,
+ SUN4I_CODEC_ADC_ACTL_PA_EN, 0,
+ sun4i_codec_pa_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_pa_mixer_controls)),
+ SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0,
+ &sun4i_codec_pa_mute),
+
+ SND_SOC_DAPM_OUTPUT("HP Right"),
+ SND_SOC_DAPM_OUTPUT("HP Left"),
+};
+
+static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = {
+ /* Left DAC Routes */
+ { "Left DAC", NULL, "DAC" },
+
+ /* Right DAC Routes */
+ { "Right DAC", NULL, "DAC" },
+
+ /* Right Mixer Routes */
+ { "Right Mixer", NULL, "Mixer Enable" },
+ { "Right Mixer", "Left DAC Playback Switch", "Left DAC" },
+ { "Right Mixer", "Right DAC Playback Switch", "Right DAC" },
+
+ /* Left Mixer Routes */
+ { "Left Mixer", NULL, "Mixer Enable" },
+ { "Left Mixer", "Left DAC Playback Switch", "Left DAC" },
+
+ /* Pre-Amplifier Mixer Routes */
+ { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" },
+ { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" },
+ { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" },
+ { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" },
+
+ /* PA -> HP path */
+ { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" },
+ { "HP Right", NULL, "Pre-Amplifier Mute" },
+ { "HP Left", NULL, "Pre-Amplifier Mute" },
+};
+
+static struct snd_soc_codec_driver sun4i_codec_codec = {
+ .controls = sun4i_codec_widgets,
+ .num_controls = ARRAY_SIZE(sun4i_codec_widgets),
+ .dapm_widgets = sun4i_codec_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets),
+ .dapm_routes = sun4i_codec_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes),
+};
+
+static const struct snd_soc_component_driver sun4i_codec_component = {
+ .name = "sun4i-codec",
+};
+
+#define SUN4I_CODEC_RATES SNDRV_PCM_RATE_8000_192000
+#define SUN4I_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static int sun4i_codec_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai);
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+
+ snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data,
+ NULL);
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver dummy_cpu_dai = {
+ .name = "sun4i-codec-cpu-dai",
+ .probe = sun4i_codec_dai_probe,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SUN4I_CODEC_RATES,
+ .formats = SUN4I_CODEC_FORMATS,
+ .sig_bits = 24,
+ },
+};
+
+static const struct regmap_config sun4i_codec_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL,
+};
+
+static const struct of_device_id sun4i_codec_of_match[] = {
+ { .compatible = "allwinner,sun4i-a10-codec" },
+ { .compatible = "allwinner,sun7i-a20-codec" },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sun4i_codec_of_match);
+
+static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev,
+ int *num_links)
+{
+ struct snd_soc_dai_link *link = devm_kzalloc(dev, sizeof(*link),
+ GFP_KERNEL);
+ if (!link)
+ return NULL;
+
+ link->name = "cdc";
+ link->stream_name = "CDC PCM";
+ link->codec_dai_name = "Codec";
+ link->cpu_dai_name = dev_name(dev);
+ link->codec_name = dev_name(dev);
+ link->platform_name = dev_name(dev);
+ link->dai_fmt = SND_SOC_DAIFMT_I2S;
+
+ *num_links = 1;
+
+ return link;
+};
+
+static struct snd_soc_card *sun4i_codec_create_card(struct device *dev)
+{
+ struct snd_soc_card *card;
+ int ret;
+
+ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return NULL;
+
+ card->dai_link = sun4i_codec_create_link(dev, &card->num_links);
+ if (!card->dai_link)
+ return NULL;
+
+ card->dev = dev;
+ card->name = "sun4i-codec";
+
+ ret = snd_soc_of_parse_audio_routing(card, "routing");
+ if (ret) {
+ dev_err(dev, "Failed to create our audio routing\n");
+ return NULL;
+ }
+
+ return card;
+};
+
+static int sun4i_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card;
+ struct sun4i_codec *scodec;
+ struct resource *res;
+ void __iomem *base;
+ int ret;
+
+ scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL);
+ if (!scodec)
+ return -ENOMEM;
+
+ scodec->dev = &pdev->dev;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(base)) {
+ dev_err(&pdev->dev, "Failed to map the registers\n");
+ return PTR_ERR(base);
+ }
+
+ scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sun4i_codec_regmap_config);
+ if (IS_ERR(scodec->regmap)) {
+ dev_err(&pdev->dev, "Failed to create our regmap\n");
+ return PTR_ERR(scodec->regmap);
+ }
+
+ /* Get the clocks from the DT */
+ scodec->clk_apb = devm_clk_get(&pdev->dev, "apb");
+ if (IS_ERR(scodec->clk_apb)) {
+ dev_err(&pdev->dev, "Failed to get the APB clock\n");
+ return PTR_ERR(scodec->clk_apb);
+ }
+
+ scodec->clk_module = devm_clk_get(&pdev->dev, "codec");
+ if (IS_ERR(scodec->clk_module)) {
+ dev_err(&pdev->dev, "Failed to get the module clock\n");
+ return PTR_ERR(scodec->clk_module);
+ }
+
+ /* Enable the bus clock */
+ if (clk_prepare_enable(scodec->clk_apb)) {
+ dev_err(&pdev->dev, "Failed to enable the APB clock\n");
+ return -EINVAL;
+ }
+
+ /* DMA configuration for TX FIFO */
+ scodec->playback_dma_data.addr = res->start + SUN4I_CODEC_DAC_TXDATA;
+ scodec->playback_dma_data.maxburst = 4;
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+
+ ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec,
+ &sun4i_codec_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our codec\n");
+ goto err_clk_disable;
+ }
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &sun4i_codec_component,
+ &dummy_cpu_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our DAI\n");
+ goto err_unregister_codec;
+ }
+
+ ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register against DMAEngine\n");
+ goto err_unregister_codec;
+ }
+
+ card = sun4i_codec_create_card(&pdev->dev);
+ if (!card) {
+ dev_err(&pdev->dev, "Failed to create our card\n");
+ goto err_unregister_codec;
+ }
+
+ platform_set_drvdata(pdev, card);
+ snd_soc_card_set_drvdata(card, scodec);
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our card\n");
+ goto err_unregister_codec;
+ }
+
+ return 0;
+
+err_unregister_codec:
+ snd_soc_unregister_codec(&pdev->dev);
+err_clk_disable:
+ clk_disable_unprepare(scodec->clk_apb);
+ return ret;
+}
+
+static int sun4i_codec_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+
+ snd_soc_unregister_card(card);
+ snd_soc_unregister_codec(&pdev->dev);
+ clk_disable_unprepare(scodec->clk_apb);
+
+ return 0;
+}
+
+static struct platform_driver sun4i_codec_driver = {
+ .driver = {
+ .name = "sun4i-codec",
+ .of_match_table = sun4i_codec_of_match,
+ },
+ .probe = sun4i_codec_probe,
+ .remove = sun4i_codec_remove,
+};
+module_platform_driver(sun4i_codec_driver);
+
+MODULE_DESCRIPTION("Allwinner A10 codec driver");
+MODULE_AUTHOR("Emilio López <emilio@elopez.com.ar>");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_AUTHOR("Maxime Ripard <maxime.ripard@free-electrons.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 4e0c0e502ade..ba9fc099cf67 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -152,6 +152,7 @@ static const struct of_device_id snd_soc_mop500_match[] = {
{ .compatible = "stericsson,snd-soc-mop500", },
{},
};
+MODULE_DEVICE_TABLE(of, snd_soc_mop500_match);
static struct platform_driver snd_soc_mop500_driver = {
.driver = {
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index f5df08ded770..6ba8ae9ecc7a 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -843,6 +843,7 @@ static const struct of_device_id ux500_msp_i2s_match[] = {
{ .compatible = "stericsson,ux500-msp-i2s", },
{},
};
+MODULE_DEVICE_TABLE(of, ux500_msp_i2s_match);
static struct platform_driver msp_i2s_driver = {
.driver = {
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 417ebb11cf48..7661616f3636 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1903,11 +1903,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi *umidi)
hostif = &intf->altsetting[1];
intfd = get_iface_desc(hostif);
+ /* If either or both of the endpoints support interrupt transfer,
+ * then use the alternate setting
+ */
if (intfd->bNumEndpoints != 2 ||
- (get_endpoint(hostif, 0)->bmAttributes &
- USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK ||
- (get_endpoint(hostif, 1)->bmAttributes &
- USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+ !((get_endpoint(hostif, 0)->bmAttributes &
+ USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT ||
+ (get_endpoint(hostif, 1)->bmAttributes &
+ USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT))
return;
dev_dbg(&umidi->dev->dev, "switching to altsetting %d with int ep\n",
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index d3608c0a29f3..fe91184ce832 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -338,7 +338,7 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol,
struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
struct usb_mixer_interface *mixer = list->mixer;
int index = kcontrol->private_value & 0xff;
- int value = ucontrol->value.integer.value[0];
+ unsigned int value = ucontrol->value.integer.value[0];
int old_value = kcontrol->private_value >> 8;
int err;