diff options
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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, sizeof(reg), 0); + + reg = 0; + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, sizeof(reg), 0); + reg = cpu_to_be32(0x00000000); + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, 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, + ®, sizeof(reg), 0); + + /* Unregister the address. */ + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI, + ®, sizeof(reg), 0); + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO, + ®, 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; |