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-rw-r--r--sound/Kconfig2
-rw-r--r--sound/Makefile2
-rw-r--r--sound/arm/aaci.c56
-rw-r--r--sound/atmel/ac97c.c5
-rw-r--r--sound/core/hrtimer.c7
-rw-r--r--sound/core/jack.c1
-rw-r--r--sound/drivers/mtpav.c3
-rw-r--r--sound/firewire/Kconfig25
-rw-r--r--sound/firewire/Makefile6
-rw-r--r--sound/firewire/amdtp.c562
-rw-r--r--sound/firewire/amdtp.h169
-rw-r--r--sound/firewire/cmp.c308
-rw-r--r--sound/firewire/cmp.h41
-rw-r--r--sound/firewire/fcp.c224
-rw-r--r--sound/firewire/fcp.h12
-rw-r--r--sound/firewire/iso-resources.c232
-rw-r--r--sound/firewire/iso-resources.h39
-rw-r--r--sound/firewire/lib.c85
-rw-r--r--sound/firewire/lib.h19
-rw-r--r--sound/firewire/packets-buffer.c74
-rw-r--r--sound/firewire/packets-buffer.h26
-rw-r--r--sound/firewire/speakers.c858
-rw-r--r--sound/oss/Makefile4
-rw-r--r--sound/pci/au88x0/au88x0_core.c14
-rw-r--r--sound/pci/azt3328.c38
-rw-r--r--sound/pci/hda/hda_eld.c2
-rw-r--r--sound/pci/hda/hda_intel.c3
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_conexant.c217
-rw-r--r--sound/pci/hda/patch_hdmi.c7
-rw-r--r--sound/pci/hda/patch_realtek.c66
-rw-r--r--sound/pci/hda/patch_sigmatel.c15
-rw-r--r--sound/pci/hda/patch_via.c2
-rw-r--r--sound/pci/ice1712/delta.c7
-rw-r--r--sound/pci/oxygen/oxygen.h2
-rw-r--r--sound/pci/oxygen/oxygen_mixer.c2
-rw-r--r--sound/pci/oxygen/xonar_cs43xx.c2
-rw-r--r--sound/pci/oxygen/xonar_dg.c36
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf.h2
-rw-r--r--sound/pcmcia/vx/vxp_ops.c2
-rw-r--r--sound/usb/caiaq/audio.c2
-rw-r--r--sound/usb/caiaq/midi.c2
-rw-r--r--sound/usb/card.c4
-rw-r--r--sound/usb/mixer.c4
-rw-r--r--sound/usb/pcm.c7
-rw-r--r--sound/usb/quirks-table.h7
-rw-r--r--sound/usb/quirks.c3
-rw-r--r--sound/usb/usbaudio.h1
48 files changed, 3041 insertions, 168 deletions
diff --git a/sound/Kconfig b/sound/Kconfig
index fcad760f5691..1fef141ef8e7 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -97,6 +97,8 @@ source "sound/sh/Kconfig"
# here assuming USB is defined before ALSA
source "sound/usb/Kconfig"
+source "sound/firewire/Kconfig"
+
# the following will depend on the order of config.
# here assuming PCMCIA is defined before ALSA
source "sound/pcmcia/Kconfig"
diff --git a/sound/Makefile b/sound/Makefile
index ec467decfa79..ce9132b1c395 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -6,7 +6,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o
obj-$(CONFIG_SOUND_PRIME) += oss/
obj-$(CONFIG_DMASOUND) += oss/
obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
- sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/
+ firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/
obj-$(CONFIG_SND_AOA) += aoa/
# This one must be compilable even if sound is configured out
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 91acc9a243ec..7c1fc64cb53d 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -30,6 +30,8 @@
#define DRIVER_NAME "aaci-pl041"
+#define FRAME_PERIOD_US 21
+
/*
* PM support is not complete. Turn it off.
*/
@@ -48,7 +50,11 @@ static void aaci_ac97_select_codec(struct aaci *aaci, struct snd_ac97 *ac97)
if (v & SLFR_1RXV)
readl(aaci->base + AACI_SL1RX);
- writel(maincr, aaci->base + AACI_MAINCR);
+ if (maincr != readl(aaci->base + AACI_MAINCR)) {
+ writel(maincr, aaci->base + AACI_MAINCR);
+ readl(aaci->base + AACI_MAINCR);
+ udelay(1);
+ }
}
/*
@@ -64,8 +70,8 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
unsigned short val)
{
struct aaci *aaci = ac97->private_data;
+ int timeout;
u32 v;
- int timeout = 5000;
if (ac97->num >= 4)
return;
@@ -81,14 +87,17 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
writel(val << 4, aaci->base + AACI_SL2TX);
writel(reg << 12, aaci->base + AACI_SL1TX);
- /*
- * Wait for the transmission of both slots to complete.
- */
+ /* Initially, wait one frame period */
+ udelay(FRAME_PERIOD_US);
+
+ /* And then wait an additional eight frame periods for it to be sent */
+ timeout = FRAME_PERIOD_US * 8;
do {
+ udelay(1);
v = readl(aaci->base + AACI_SLFR);
} while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
- if (!timeout)
+ if (v & (SLFR_1TXB|SLFR_2TXB))
dev_err(&aaci->dev->dev,
"timeout waiting for write to complete\n");
@@ -101,9 +110,8 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
{
struct aaci *aaci = ac97->private_data;
+ int timeout, retries = 10;
u32 v;
- int timeout = 5000;
- int retries = 10;
if (ac97->num >= 4)
return ~0;
@@ -117,35 +125,34 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
*/
writel((reg << 12) | (1 << 19), aaci->base + AACI_SL1TX);
- /*
- * Wait for the transmission to complete.
- */
+ /* Initially, wait one frame period */
+ udelay(FRAME_PERIOD_US);
+
+ /* And then wait an additional eight frame periods for it to be sent */
+ timeout = FRAME_PERIOD_US * 8;
do {
+ udelay(1);
v = readl(aaci->base + AACI_SLFR);
} while ((v & SLFR_1TXB) && --timeout);
- if (!timeout) {
+ if (v & SLFR_1TXB) {
dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
v = ~0;
goto out;
}
- /*
- * Give the AC'97 codec more than enough time
- * to respond. (42us = ~2 frames at 48kHz.)
- */
- udelay(42);
+ /* Now wait for the response frame */
+ udelay(FRAME_PERIOD_US);
- /*
- * Wait for slot 2 to indicate data.
- */
- timeout = 5000;
+ /* And then wait an additional eight frame periods for data */
+ timeout = FRAME_PERIOD_US * 8;
do {
+ udelay(1);
cond_resched();
v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
} while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
- if (!timeout) {
+ if (v != (SLFR_1RXV|SLFR_2RXV)) {
dev_err(&aaci->dev->dev, "timeout on RX valid\n");
v = ~0;
goto out;
@@ -179,6 +186,7 @@ aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask)
int timeout = 5000;
do {
+ udelay(1);
val = readl(aacirun->base + AACI_SR);
} while (val & mask && timeout--);
}
@@ -874,7 +882,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci)
* Give the AC'97 codec more than enough time
* to wake up. (42us = ~2 frames at 48kHz.)
*/
- udelay(42);
+ udelay(FRAME_PERIOD_US * 2);
ret = snd_ac97_bus(aaci->card, 0, &aaci_bus_ops, aaci, &ac97_bus);
if (ret)
@@ -989,6 +997,8 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
* disabling the channel doesn't clear the FIFO.
*/
writel(aaci->maincr & ~MAINCR_IE, aaci->base + AACI_MAINCR);
+ readl(aaci->base + AACI_MAINCR);
+ udelay(1);
writel(aaci->maincr, aaci->base + AACI_MAINCR);
/*
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 10c3a871a12d..b310702c646e 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -33,9 +33,12 @@
#include <linux/dw_dmac.h>
#include <mach/cpu.h>
-#include <mach/hardware.h>
#include <mach/gpio.h>
+#ifdef CONFIG_ARCH_AT91
+#include <mach/hardware.h>
+#endif
+
#include "ac97c.h"
enum {
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 7730575bfadd..b8b31c433d64 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
{
struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
struct snd_timer *t = stime->timer;
+ unsigned long oruns;
if (!atomic_read(&stime->running))
return HRTIMER_NORESTART;
- hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
- snd_timer_interrupt(stime->timer, t->sticks);
+ oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
+ snd_timer_interrupt(stime->timer, t->sticks * oruns);
if (!atomic_read(&stime->running))
return HRTIMER_NORESTART;
@@ -104,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
}
static struct snd_timer_hardware hrtimer_hw = {
- .flags = SNDRV_TIMER_HW_AUTO,
+ .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
.open = snd_hrtimer_open,
.close = snd_hrtimer_close,
.start = snd_hrtimer_start,
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 4902ae568730..53b53e97c896 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -141,6 +141,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
fail_input:
input_free_device(jack->input_dev);
+ kfree(jack->id);
kfree(jack);
return err;
}
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index da03597fc893..5c426df87678 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -55,14 +55,13 @@
#include <linux/err.h>
#include <linux/platform_device.h>
#include <linux/ioport.h>
+#include <linux/io.h>
#include <linux/moduleparam.h>
#include <sound/core.h>
#include <sound/initval.h>
#include <sound/rawmidi.h>
#include <linux/delay.h>
-#include <asm/io.h>
-
/*
* globals
*/
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
new file mode 100644
index 000000000000..e486f48660fb
--- /dev/null
+++ b/sound/firewire/Kconfig
@@ -0,0 +1,25 @@
+menuconfig SND_FIREWIRE
+ bool "FireWire sound devices"
+ depends on FIREWIRE
+ default y
+ help
+ Support for IEEE-1394/FireWire/iLink sound devices.
+
+if SND_FIREWIRE && FIREWIRE
+
+config SND_FIREWIRE_LIB
+ tristate
+ depends on SND_PCM
+
+config SND_FIREWIRE_SPEAKERS
+ tristate "FireWire speakers"
+ select SND_PCM
+ select SND_FIREWIRE_LIB
+ help
+ Say Y here to include support for the Griffin FireWave Surround
+ and the LaCie FireWire Speakers.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-firewire-speakers.
+
+endif # SND_FIREWIRE
diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile
new file mode 100644
index 000000000000..e5b1634d9ad4
--- /dev/null
+++ b/sound/firewire/Makefile
@@ -0,0 +1,6 @@
+snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \
+ fcp.o cmp.o amdtp.o
+snd-firewire-speakers-objs := speakers.o
+
+obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o
+obj-$(CONFIG_SND_FIREWIRE_SPEAKERS) += snd-firewire-speakers.o
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
new file mode 100644
index 000000000000..b18140ff2b93
--- /dev/null
+++ b/sound/firewire/amdtp.c
@@ -0,0 +1,562 @@
+/*
+ * Audio and Music Data Transmission Protocol (IEC 61883-6) streams
+ * with Common Isochronous Packet (IEC 61883-1) headers
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/device.h>
+#include <linux/err.h>
+#include <linux/firewire.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include "amdtp.h"
+
+#define TICKS_PER_CYCLE 3072
+#define CYCLES_PER_SECOND 8000
+#define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND)
+
+#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 µs */
+
+#define TAG_CIP 1
+
+#define CIP_EOH (1u << 31)
+#define CIP_FMT_AM (0x10 << 24)
+#define AMDTP_FDF_AM824 (0 << 19)
+#define AMDTP_FDF_SFC_SHIFT 16
+
+/* TODO: make these configurable */
+#define INTERRUPT_INTERVAL 16
+#define QUEUE_LENGTH 48
+
+/**
+ * amdtp_out_stream_init - initialize an AMDTP output stream structure
+ * @s: the AMDTP output stream to initialize
+ * @unit: the target of the stream
+ * @flags: the packet transmission method to use
+ */
+int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
+ enum cip_out_flags flags)
+{
+ if (flags != CIP_NONBLOCKING)
+ return -EINVAL;
+
+ s->unit = fw_unit_get(unit);
+ s->flags = flags;
+ s->context = ERR_PTR(-1);
+ mutex_init(&s->mutex);
+ s->packet_index = 0;
+
+ return 0;
+}
+EXPORT_SYMBOL(amdtp_out_stream_init);
+
+/**
+ * amdtp_out_stream_destroy - free stream resources
+ * @s: the AMDTP output stream to destroy
+ */
+void amdtp_out_stream_destroy(struct amdtp_out_stream *s)
+{
+ WARN_ON(!IS_ERR(s->context));
+ mutex_destroy(&s->mutex);
+ fw_unit_put(s->unit);
+}
+EXPORT_SYMBOL(amdtp_out_stream_destroy);
+
+/**
+ * amdtp_out_stream_set_rate - set the sample rate
+ * @s: the AMDTP output stream to configure
+ * @rate: the sample rate
+ *
+ * The sample rate must be set before the stream is started, and must not be
+ * changed while the stream is running.
+ */
+void amdtp_out_stream_set_rate(struct amdtp_out_stream *s, unsigned int rate)
+{
+ static const struct {
+ unsigned int rate;
+ unsigned int syt_interval;
+ } rate_info[] = {
+ [CIP_SFC_32000] = { 32000, 8, },
+ [CIP_SFC_44100] = { 44100, 8, },
+ [CIP_SFC_48000] = { 48000, 8, },
+ [CIP_SFC_88200] = { 88200, 16, },
+ [CIP_SFC_96000] = { 96000, 16, },
+ [CIP_SFC_176400] = { 176400, 32, },
+ [CIP_SFC_192000] = { 192000, 32, },
+ };
+ unsigned int sfc;
+
+ if (WARN_ON(!IS_ERR(s->context)))
+ return;
+
+ for (sfc = 0; sfc < ARRAY_SIZE(rate_info); ++sfc)
+ if (rate_info[sfc].rate == rate) {
+ s->sfc = sfc;
+ s->syt_interval = rate_info[sfc].syt_interval;
+ return;
+ }
+ WARN_ON(1);
+}
+EXPORT_SYMBOL(amdtp_out_stream_set_rate);
+
+/**
+ * amdtp_out_stream_get_max_payload - get the stream's packet size
+ * @s: the AMDTP output stream
+ *
+ * This function must not be called before the stream has been configured
+ * with amdtp_out_stream_set_hw_params(), amdtp_out_stream_set_pcm(), and
+ * amdtp_out_stream_set_midi().
+ */
+unsigned int amdtp_out_stream_get_max_payload(struct amdtp_out_stream *s)
+{
+ static const unsigned int max_data_blocks[] = {
+ [CIP_SFC_32000] = 4,
+ [CIP_SFC_44100] = 6,
+ [CIP_SFC_48000] = 6,
+ [CIP_SFC_88200] = 12,
+ [CIP_SFC_96000] = 12,
+ [CIP_SFC_176400] = 23,
+ [CIP_SFC_192000] = 24,
+ };
+
+ s->data_block_quadlets = s->pcm_channels;
+ s->data_block_quadlets += DIV_ROUND_UP(s->midi_ports, 8);
+
+ return 8 + max_data_blocks[s->sfc] * 4 * s->data_block_quadlets;
+}
+EXPORT_SYMBOL(amdtp_out_stream_get_max_payload);
+
+static void amdtp_write_s16(struct amdtp_out_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+static void amdtp_write_s32(struct amdtp_out_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+
+/**
+ * amdtp_out_stream_set_pcm_format - set the PCM format
+ * @s: the AMDTP output stream to configure
+ * @format: the format of the ALSA PCM device
+ *
+ * The sample format must be set before the stream is started, and must not be
+ * changed while the stream is running.
+ */
+void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
+ snd_pcm_format_t format)
+{
+ if (WARN_ON(!IS_ERR(s->context)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ s->transfer_samples = amdtp_write_s16;
+ break;
+ case SNDRV_PCM_FORMAT_S32:
+ s->transfer_samples = amdtp_write_s32;
+ break;
+ }
+}
+EXPORT_SYMBOL(amdtp_out_stream_set_pcm_format);
+
+static unsigned int calculate_data_blocks(struct amdtp_out_stream *s)
+{
+ unsigned int phase, data_blocks;
+
+ if (!cip_sfc_is_base_44100(s->sfc)) {
+ /* Sample_rate / 8000 is an integer, and precomputed. */
+ data_blocks = s->data_block_state;
+ } else {
+ phase = s->data_block_state;
+
+ /*
+ * This calculates the number of data blocks per packet so that
+ * 1) the overall rate is correct and exactly synchronized to
+ * the bus clock, and
+ * 2) packets with a rounded-up number of blocks occur as early
+ * as possible in the sequence (to prevent underruns of the
+ * device's buffer).
+ */
+ if (s->sfc == CIP_SFC_44100)
+ /* 6 6 5 6 5 6 5 ... */
+ data_blocks = 5 + ((phase & 1) ^
+ (phase == 0 || phase >= 40));
+ else
+ /* 12 11 11 11 11 ... or 23 22 22 22 22 ... */
+ data_blocks = 11 * (s->sfc >> 1) + (phase == 0);
+ if (++phase >= (80 >> (s->sfc >> 1)))
+ phase = 0;
+ s->data_block_state = phase;
+ }
+
+ return data_blocks;
+}
+
+static unsigned int calculate_syt(struct amdtp_out_stream *s,
+ unsigned int cycle)
+{
+ unsigned int syt_offset, phase, index, syt;
+
+ if (s->last_syt_offset < TICKS_PER_CYCLE) {
+ if (!cip_sfc_is_base_44100(s->sfc))
+ syt_offset = s->last_syt_offset + s->syt_offset_state;
+ else {
+ /*
+ * The time, in ticks, of the n'th SYT_INTERVAL sample is:
+ * n * SYT_INTERVAL * 24576000 / sample_rate
+ * Modulo TICKS_PER_CYCLE, the difference between successive
+ * elements is about 1386.23. Rounding the results of this
+ * formula to the SYT precision results in a sequence of
+ * differences that begins with:
+ * 1386 1386 1387 1386 1386 1386 1387 1386 1386 1386 1387 ...
+ * This code generates _exactly_ the same sequence.
+ */
+ phase = s->syt_offset_state;
+ index = phase % 13;
+ syt_offset = s->last_syt_offset;
+ syt_offset += 1386 + ((index && !(index & 3)) ||
+ phase == 146);
+ if (++phase >= 147)
+ phase = 0;
+ s->syt_offset_state = phase;
+ }
+ } else
+ syt_offset = s->last_syt_offset - TICKS_PER_CYCLE;
+ s->last_syt_offset = syt_offset;
+
+ if (syt_offset < TICKS_PER_CYCLE) {
+ syt_offset += TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE;
+ syt = (cycle + syt_offset / TICKS_PER_CYCLE) << 12;
+ syt += syt_offset % TICKS_PER_CYCLE;
+
+ return syt & 0xffff;
+ } else {
+ return 0xffff; /* no info */
+ }
+}
+
+static void amdtp_write_s32(struct amdtp_out_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, frame_step, i, c;
+ const u32 *src;
+
+ channels = s->pcm_channels;
+ src = (void *)runtime->dma_area +
+ s->pcm_buffer_pointer * (runtime->frame_bits / 8);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+ frame_step = s->data_block_quadlets - channels;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *buffer = cpu_to_be32((*src >> 8) | 0x40000000);
+ src++;
+ buffer++;
+ }
+ buffer += frame_step;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void amdtp_write_s16(struct amdtp_out_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, frame_step, i, c;
+ const u16 *src;
+
+ channels = s->pcm_channels;
+ src = (void *)runtime->dma_area +
+ s->pcm_buffer_pointer * (runtime->frame_bits / 8);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+ frame_step = s->data_block_quadlets - channels;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *buffer = cpu_to_be32((*src << 8) | 0x40000000);
+ src++;
+ buffer++;
+ }
+ buffer += frame_step;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void amdtp_fill_pcm_silence(struct amdtp_out_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[c] = cpu_to_be32(0x40000000);
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static void amdtp_fill_midi(struct amdtp_out_stream *s,
+ __be32 *buffer, unsigned int frames)
+{
+ unsigned int i;
+
+ for (i = 0; i < frames; ++i)
+ buffer[s->pcm_channels + i * s->data_block_quadlets] =
+ cpu_to_be32(0x80000000);
+}
+
+static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle)
+{
+ __be32 *buffer;
+ unsigned int index, data_blocks, syt, ptr;
+ struct snd_pcm_substream *pcm;
+ struct fw_iso_packet packet;
+ int err;
+
+ if (s->packet_index < 0)
+ return;
+ index = s->packet_index;
+
+ data_blocks = calculate_data_blocks(s);
+ syt = calculate_syt(s, cycle);
+
+ buffer = s->buffer.packets[index].buffer;
+ buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
+ (s->data_block_quadlets << 16) |
+ s->data_block_counter);
+ buffer[1] = cpu_to_be32(CIP_EOH | CIP_FMT_AM | AMDTP_FDF_AM824 |
+ (s->sfc << AMDTP_FDF_SFC_SHIFT) | syt);
+ buffer += 2;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm)
+ s->transfer_samples(s, pcm, buffer, data_blocks);
+ else
+ amdtp_fill_pcm_silence(s, buffer, data_blocks);
+ if (s->midi_ports)
+ amdtp_fill_midi(s, buffer, data_blocks);
+
+ s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff;
+
+ packet.payload_length = 8 + data_blocks * 4 * s->data_block_quadlets;
+ packet.interrupt = IS_ALIGNED(index + 1, INTERRUPT_INTERVAL);
+ packet.skip = 0;
+ packet.tag = TAG_CIP;
+ packet.sy = 0;
+ packet.header_length = 0;
+
+ err = fw_iso_context_queue(s->context, &packet, &s->buffer.iso_buffer,
+ s->buffer.packets[index].offset);
+ if (err < 0) {
+ dev_err(&s->unit->device, "queueing error: %d\n", err);
+ s->packet_index = -1;
+ amdtp_out_stream_pcm_abort(s);
+ return;
+ }
+
+ if (++index >= QUEUE_LENGTH)
+ index = 0;
+ s->packet_index = index;
+
+ if (pcm) {
+ ptr = s->pcm_buffer_pointer + data_blocks;
+ if (ptr >= pcm->runtime->buffer_size)
+ ptr -= pcm->runtime->buffer_size;
+ ACCESS_ONCE(s->pcm_buffer_pointer) = ptr;
+
+ s->pcm_period_pointer += data_blocks;
+ if (s->pcm_period_pointer >= pcm->runtime->period_size) {
+ s->pcm_period_pointer -= pcm->runtime->period_size;
+ snd_pcm_period_elapsed(pcm);
+ }
+ }
+}
+
+static void out_packet_callback(struct fw_iso_context *context, u32 cycle,
+ size_t header_length, void *header, void *data)
+{
+ struct amdtp_out_stream *s = data;
+ unsigned int i, packets = header_length / 4;
+
+ /*
+ * Compute the cycle of the last queued packet.
+ * (We need only the four lowest bits for the SYT, so we can ignore
+ * that bits 0-11 must wrap around at 3072.)
+ */
+ cycle += QUEUE_LENGTH - packets;
+
+ for (i = 0; i < packets; ++i)
+ queue_out_packet(s, ++cycle);
+}
+
+static int queue_initial_skip_packets(struct amdtp_out_stream *s)
+{
+ struct fw_iso_packet skip_packet = {
+ .skip = 1,
+ };
+ unsigned int i;
+ int err;
+
+ for (i = 0; i < QUEUE_LENGTH; ++i) {
+ skip_packet.interrupt = IS_ALIGNED(s->packet_index + 1,
+ INTERRUPT_INTERVAL);
+ err = fw_iso_context_queue(s->context, &skip_packet, NULL, 0);
+ if (err < 0)
+ return err;
+ if (++s->packet_index >= QUEUE_LENGTH)
+ s->packet_index = 0;
+ }
+
+ return 0;
+}
+
+/**
+ * amdtp_out_stream_start - start sending packets
+ * @s: the AMDTP output stream to start
+ * @channel: the isochronous channel on the bus
+ * @speed: firewire speed code
+ *
+ * The stream cannot be started until it has been configured with
+ * amdtp_out_stream_set_hw_params(), amdtp_out_stream_set_pcm(), and
+ * amdtp_out_stream_set_midi(); and it must be started before any
+ * PCM or MIDI device can be started.
+ */
+int amdtp_out_stream_start(struct amdtp_out_stream *s, int channel, int speed)
+{
+ static const struct {
+ unsigned int data_block;
+ unsigned int syt_offset;
+ } initial_state[] = {
+ [CIP_SFC_32000] = { 4, 3072 },
+ [CIP_SFC_48000] = { 6, 1024 },
+ [CIP_SFC_96000] = { 12, 1024 },
+ [CIP_SFC_192000] = { 24, 1024 },
+ [CIP_SFC_44100] = { 0, 67 },
+ [CIP_SFC_88200] = { 0, 67 },
+ [CIP_SFC_176400] = { 0, 67 },
+ };
+ int err;
+
+ mutex_lock(&s->mutex);
+
+ if (WARN_ON(!IS_ERR(s->context) ||
+ (!s->pcm_channels && !s->midi_ports))) {
+ err = -EBADFD;
+ goto err_unlock;
+ }
+
+ s->data_block_state = initial_state[s->sfc].data_block;
+ s->syt_offset_state = initial_state[s->sfc].syt_offset;
+ s->last_syt_offset = TICKS_PER_CYCLE;
+
+ err = iso_packets_buffer_init(&s->buffer, s->unit, QUEUE_LENGTH,
+ amdtp_out_stream_get_max_payload(s),
+ DMA_TO_DEVICE);
+ if (err < 0)
+ goto err_unlock;
+
+ s->context = fw_iso_context_create(fw_parent_device(s->unit)->card,
+ FW_ISO_CONTEXT_TRANSMIT,
+ channel, speed, 0,
+ out_packet_callback, s);
+ if (IS_ERR(s->context)) {
+ err = PTR_ERR(s->context);
+ if (err == -EBUSY)
+ dev_err(&s->unit->device,
+ "no free output stream on this controller\n");
+ goto err_buffer;
+ }
+
+ amdtp_out_stream_update(s);
+
+ s->packet_index = 0;
+ s->data_block_counter = 0;
+ err = queue_initial_skip_packets(s);
+ if (err < 0)
+ goto err_context;
+
+ err = fw_iso_context_start(s->context, -1, 0, 0);
+ if (err < 0)
+ goto err_context;
+
+ mutex_unlock(&s->mutex);
+
+ return 0;
+
+err_context:
+ fw_iso_context_destroy(s->context);
+ s->context = ERR_PTR(-1);
+err_buffer:
+ iso_packets_buffer_destroy(&s->buffer, s->unit);
+err_unlock:
+ mutex_unlock(&s->mutex);
+
+ return err;
+}
+EXPORT_SYMBOL(amdtp_out_stream_start);
+
+/**
+ * amdtp_out_stream_update - update the stream after a bus reset
+ * @s: the AMDTP output stream
+ */
+void amdtp_out_stream_update(struct amdtp_out_stream *s)
+{
+ ACCESS_ONCE(s->source_node_id_field) =
+ (fw_parent_device(s->unit)->card->node_id & 0x3f) << 24;
+}
+EXPORT_SYMBOL(amdtp_out_stream_update);
+
+/**
+ * amdtp_out_stream_stop - stop sending packets
+ * @s: the AMDTP output stream to stop
+ *
+ * All PCM and MIDI devices of the stream must be stopped before the stream
+ * itself can be stopped.
+ */
+void amdtp_out_stream_stop(struct amdtp_out_stream *s)
+{
+ mutex_lock(&s->mutex);
+
+ if (IS_ERR(s->context)) {
+ mutex_unlock(&s->mutex);
+ return;
+ }
+
+ fw_iso_context_stop(s->context);
+ fw_iso_context_destroy(s->context);
+ s->context = ERR_PTR(-1);
+ iso_packets_buffer_destroy(&s->buffer, s->unit);
+
+ mutex_unlock(&s->mutex);
+}
+EXPORT_SYMBOL(amdtp_out_stream_stop);
+
+/**
+ * amdtp_out_stream_pcm_abort - abort the running PCM device
+ * @s: the AMDTP stream about to be stopped
+ *
+ * If the isochronous stream needs to be stopped asynchronously, call this
+ * function first to stop the PCM device.
+ */
+void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s)
+{
+ struct snd_pcm_substream *pcm;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm) {
+ snd_pcm_stream_lock_irq(pcm);
+ if (snd_pcm_running(pcm))
+ snd_pcm_stop(pcm, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stream_unlock_irq(pcm);
+ }
+}
+EXPORT_SYMBOL(amdtp_out_stream_pcm_abort);
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
new file mode 100644
index 000000000000..537a9cb83581
--- /dev/null
+++ b/sound/firewire/amdtp.h
@@ -0,0 +1,169 @@
+#ifndef SOUND_FIREWIRE_AMDTP_H_INCLUDED
+#define SOUND_FIREWIRE_AMDTP_H_INCLUDED
+
+#include <linux/mutex.h>
+#include <linux/spinlock.h>
+#include "packets-buffer.h"
+
+/**
+ * enum cip_out_flags - describes details of the streaming protocol
+ * @CIP_NONBLOCKING: In non-blocking mode, each packet contains
+ * sample_rate/8000 samples, with rounding up or down to adjust
+ * for clock skew and left-over fractional samples. This should
+ * be used if supported by the device.
+ */
+enum cip_out_flags {
+ CIP_NONBLOCKING = 0,
+};
+
+/**
+ * enum cip_sfc - a stream's sample rate
+ */
+enum cip_sfc {
+ CIP_SFC_32000 = 0,
+ CIP_SFC_44100 = 1,
+ CIP_SFC_48000 = 2,
+ CIP_SFC_88200 = 3,
+ CIP_SFC_96000 = 4,
+ CIP_SFC_176400 = 5,
+ CIP_SFC_192000 = 6,
+};
+
+#define AMDTP_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \
+ SNDRV_PCM_FMTBIT_S32)
+
+struct fw_unit;
+struct fw_iso_context;
+struct snd_pcm_substream;
+
+struct amdtp_out_stream {
+ struct fw_unit *unit;
+ enum cip_out_flags flags;
+ 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_out_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+
+ unsigned int syt_interval;
+ unsigned int source_node_id_field;
+ struct iso_packets_buffer buffer;
+
+ struct snd_pcm_substream *pcm;
+
+ int packet_index;
+ unsigned int data_block_counter;
+
+ unsigned int data_block_state;
+
+ unsigned int last_syt_offset;
+ unsigned int syt_offset_state;
+
+ unsigned int pcm_buffer_pointer;
+ unsigned int pcm_period_pointer;
+};
+
+int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
+ enum cip_out_flags flags);
+void amdtp_out_stream_destroy(struct amdtp_out_stream *s);
+
+void amdtp_out_stream_set_rate(struct amdtp_out_stream *s, unsigned int rate);
+unsigned int amdtp_out_stream_get_max_payload(struct amdtp_out_stream *s);
+
+int amdtp_out_stream_start(struct amdtp_out_stream *s, int channel, int speed);
+void amdtp_out_stream_update(struct amdtp_out_stream *s);
+void amdtp_out_stream_stop(struct amdtp_out_stream *s);
+
+void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
+ snd_pcm_format_t format);
+void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s);
+
+/**
+ * amdtp_out_stream_set_pcm - configure format of PCM samples
+ * @s: the AMDTP output stream to be configured
+ * @pcm_channels: the number of PCM samples in each data block, to be encoded
+ * as AM824 multi-bit linear audio
+ *
+ * This function must not be called while the stream is running.
+ */
+static inline void amdtp_out_stream_set_pcm(struct amdtp_out_stream *s,
+ unsigned int pcm_channels)
+{
+ s->pcm_channels = pcm_channels;
+}
+
+/**
+ * amdtp_out_stream_set_midi - configure format of MIDI data
+ * @s: the AMDTP output stream to be configured
+ * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ *
+ * This function must not be called while the stream is running.
+ */
+static inline void amdtp_out_stream_set_midi(struct amdtp_out_stream *s,
+ unsigned int midi_ports)
+{
+ s->midi_ports = midi_ports;
+}
+
+/**
+ * amdtp_out_streaming_error - check for streaming error
+ * @s: the AMDTP output stream
+ *
+ * If this function returns true, the stream's packet queue has stopped due to
+ * an asynchronous error.
+ */
+static inline bool amdtp_out_streaming_error(struct amdtp_out_stream *s)
+{
+ return s->packet_index < 0;
+}
+
+/**
+ * amdtp_out_stream_pcm_prepare - prepare PCM device for running
+ * @s: the AMDTP output stream
+ *
+ * This function should be called from the PCM device's .prepare callback.
+ */
+static inline void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s)
+{
+ s->pcm_buffer_pointer = 0;
+ s->pcm_period_pointer = 0;
+}
+
+/**
+ * amdtp_out_stream_pcm_trigger - start/stop playback from a PCM device
+ * @s: the AMDTP output stream
+ * @pcm: the PCM 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 PCM data. This function should be called from the PCM
+ * device's .trigger callback.
+ */
+static inline void amdtp_out_stream_pcm_trigger(struct amdtp_out_stream *s,
+ struct snd_pcm_substream *pcm)
+{
+ ACCESS_ONCE(s->pcm) = pcm;
+}
+
+/**
+ * amdtp_out_stream_pcm_pointer - get the PCM buffer position
+ * @s: the AMDTP output stream that transports the PCM data
+ *
+ * Returns the current buffer position, in frames.
+ */
+static inline unsigned long
+amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s)
+{
+ return ACCESS_ONCE(s->pcm_buffer_pointer);
+}
+
+static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
+{
+ return sfc & 1;
+}
+
+#endif
diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c
new file mode 100644
index 000000000000..4a37f3a6fab9
--- /dev/null
+++ b/sound/firewire/cmp.c
@@ -0,0 +1,308 @@
+/*
+ * Connection Management Procedures (IEC 61883-1) helper functions
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/module.h>
+#include <linux/sched.h>
+#include "lib.h"
+#include "iso-resources.h"
+#include "cmp.h"
+
+#define IMPR_SPEED_MASK 0xc0000000
+#define IMPR_SPEED_SHIFT 30
+#define IMPR_XSPEED_MASK 0x00000060
+#define IMPR_XSPEED_SHIFT 5
+#define IMPR_PLUGS_MASK 0x0000001f
+
+#define IPCR_ONLINE 0x80000000
+#define IPCR_BCAST_CONN 0x40000000
+#define IPCR_P2P_CONN_MASK 0x3f000000
+#define IPCR_P2P_CONN_SHIFT 24
+#define IPCR_CHANNEL_MASK 0x003f0000
+#define IPCR_CHANNEL_SHIFT 16
+
+enum bus_reset_handling {
+ ABORT_ON_BUS_RESET,
+ SUCCEED_ON_BUS_RESET,
+};
+
+static __attribute__((format(printf, 2, 3)))
+void cmp_error(struct cmp_connection *c, const char *fmt, ...)
+{
+ va_list va;
+
+ va_start(va, fmt);
+ dev_err(&c->resources.unit->device, "%cPCR%u: %pV",
+ 'i', c->pcr_index, &(struct va_format){ fmt, &va });
+ va_end(va);
+}
+
+static int pcr_modify(struct cmp_connection *c,
+ __be32 (*modify)(struct cmp_connection *c, __be32 old),
+ int (*check)(struct cmp_connection *c, __be32 pcr),
+ enum bus_reset_handling bus_reset_handling)
+{
+ struct fw_device *device = fw_parent_device(c->resources.unit);
+ __be32 *buffer = c->resources.buffer;
+ int generation = c->resources.generation;
+ int rcode, errors = 0;
+ __be32 old_arg;
+ int err;
+
+ buffer[0] = c->last_pcr_value;
+ for (;;) {
+ old_arg = buffer[0];
+ buffer[1] = modify(c, buffer[0]);
+
+ rcode = fw_run_transaction(
+ device->card, TCODE_LOCK_COMPARE_SWAP,
+ device->node_id, generation, device->max_speed,
+ CSR_REGISTER_BASE + CSR_IPCR(c->pcr_index),
+ buffer, 8);
+
+ if (rcode == RCODE_COMPLETE) {
+ if (buffer[0] == old_arg) /* success? */
+ break;
+
+ if (check) {
+ err = check(c, buffer[0]);
+ if (err < 0)
+ return err;
+ }
+ } else if (rcode == RCODE_GENERATION)
+ goto bus_reset;
+ else if (rcode_is_permanent_error(rcode) || ++errors >= 3)
+ goto io_error;
+ }
+ c->last_pcr_value = buffer[1];
+
+ return 0;
+
+io_error:
+ cmp_error(c, "transaction failed: %s\n", rcode_string(rcode));
+ return -EIO;
+
+bus_reset:
+ return bus_reset_handling == ABORT_ON_BUS_RESET ? -EAGAIN : 0;
+}
+
+
+/**
+ * cmp_connection_init - initializes a connection manager
+ * @c: the connection manager to initialize
+ * @unit: a unit of the target device
+ * @ipcr_index: the index of the iPCR on the target device
+ */
+int cmp_connection_init(struct cmp_connection *c,
+ struct fw_unit *unit,
+ unsigned int ipcr_index)
+{
+ __be32 impr_be;
+ u32 impr;
+ int err;
+
+ err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST,
+ CSR_REGISTER_BASE + CSR_IMPR,
+ &impr_be, 4);
+ if (err < 0)
+ return err;
+ impr = be32_to_cpu(impr_be);
+
+ if (ipcr_index >= (impr & IMPR_PLUGS_MASK))
+ return -EINVAL;
+
+ err = fw_iso_resources_init(&c->resources, unit);
+ if (err < 0)
+ return err;
+
+ c->connected = false;
+ mutex_init(&c->mutex);
+ c->last_pcr_value = cpu_to_be32(0x80000000);
+ c->pcr_index = ipcr_index;
+ c->max_speed = (impr & IMPR_SPEED_MASK) >> IMPR_SPEED_SHIFT;
+ if (c->max_speed == SCODE_BETA)
+ c->max_speed += (impr & IMPR_XSPEED_MASK) >> IMPR_XSPEED_SHIFT;
+
+ return 0;
+}
+EXPORT_SYMBOL(cmp_connection_init);
+
+/**
+ * cmp_connection_destroy - free connection manager resources
+ * @c: the connection manager
+ */
+void cmp_connection_destroy(struct cmp_connection *c)
+{
+ WARN_ON(c->connected);
+ mutex_destroy(&c->mutex);
+ fw_iso_resources_destroy(&c->resources);
+}
+EXPORT_SYMBOL(cmp_connection_destroy);
+
+
+static __be32 ipcr_set_modify(struct cmp_connection *c, __be32 ipcr)
+{
+ ipcr &= ~cpu_to_be32(IPCR_BCAST_CONN |
+ IPCR_P2P_CONN_MASK |
+ IPCR_CHANNEL_MASK);
+ ipcr |= cpu_to_be32(1 << IPCR_P2P_CONN_SHIFT);
+ ipcr |= cpu_to_be32(c->resources.channel << IPCR_CHANNEL_SHIFT);
+
+ return ipcr;
+}
+
+static int ipcr_set_check(struct cmp_connection *c, __be32 ipcr)
+{
+ if (ipcr & cpu_to_be32(IPCR_BCAST_CONN |
+ IPCR_P2P_CONN_MASK)) {
+ cmp_error(c, "plug is already in use\n");
+ return -EBUSY;
+ }
+ if (!(ipcr & cpu_to_be32(IPCR_ONLINE))) {
+ cmp_error(c, "plug is not on-line\n");
+ return -ECONNREFUSED;
+ }
+
+ return 0;
+}
+
+/**
+ * cmp_connection_establish - establish a connection to the target
+ * @c: the connection manager
+ * @max_payload_bytes: the amount of data (including CIP headers) per packet
+ *
+ * This function establishes a point-to-point connection from the local
+ * computer to the target by allocating isochronous resources (channel and
+ * bandwidth) and setting the target's input plug control register. When this
+ * function succeeds, the caller is responsible for starting transmitting
+ * packets.
+ */
+int cmp_connection_establish(struct cmp_connection *c,
+ unsigned int max_payload_bytes)
+{
+ int err;
+
+ if (WARN_ON(c->connected))
+ return -EISCONN;
+
+ c->speed = min(c->max_speed,
+ fw_parent_device(c->resources.unit)->max_speed);
+
+ mutex_lock(&c->mutex);
+
+retry_after_bus_reset:
+ err = fw_iso_resources_allocate(&c->resources,
+ max_payload_bytes, c->speed);
+ if (err < 0)
+ goto err_mutex;
+
+ err = pcr_modify(c, ipcr_set_modify, ipcr_set_check,
+ ABORT_ON_BUS_RESET);
+ if (err == -EAGAIN) {
+ fw_iso_resources_free(&c->resources);
+ goto retry_after_bus_reset;
+ }
+ if (err < 0)
+ goto err_resources;
+
+ c->connected = true;
+
+ mutex_unlock(&c->mutex);
+
+ return 0;
+
+err_resources:
+ fw_iso_resources_free(&c->resources);
+err_mutex:
+ mutex_unlock(&c->mutex);
+
+ return err;
+}
+EXPORT_SYMBOL(cmp_connection_establish);
+
+/**
+ * cmp_connection_update - update the connection after a bus reset
+ * @c: the connection manager
+ *
+ * This function must be called from the driver's .update handler to reestablish
+ * any connection that might have been active.
+ *
+ * Returns zero on success, or a negative error code. On an error, the
+ * connection is broken and the caller must stop transmitting iso packets.
+ */
+int cmp_connection_update(struct cmp_connection *c)
+{
+ int err;
+
+ mutex_lock(&c->mutex);
+
+ if (!c->connected) {
+ mutex_unlock(&c->mutex);
+ return 0;
+ }
+
+ err = fw_iso_resources_update(&c->resources);
+ if (err < 0)
+ goto err_unconnect;
+
+ err = pcr_modify(c, ipcr_set_modify, ipcr_set_check,
+ SUCCEED_ON_BUS_RESET);
+ if (err < 0)
+ goto err_resources;
+
+ mutex_unlock(&c->mutex);
+
+ return 0;
+
+err_resources:
+ fw_iso_resources_free(&c->resources);
+err_unconnect:
+ c->connected = false;
+ mutex_unlock(&c->mutex);
+
+ return err;
+}
+EXPORT_SYMBOL(cmp_connection_update);
+
+
+static __be32 ipcr_break_modify(struct cmp_connection *c, __be32 ipcr)
+{
+ return ipcr & ~cpu_to_be32(IPCR_BCAST_CONN | IPCR_P2P_CONN_MASK);
+}
+
+/**
+ * cmp_connection_break - break the connection to the target
+ * @c: the connection manager
+ *
+ * This function deactives the connection in the target's input plug control
+ * register, and frees the isochronous resources of the connection. Before
+ * calling this function, the caller should cease transmitting packets.
+ */
+void cmp_connection_break(struct cmp_connection *c)
+{
+ int err;
+
+ mutex_lock(&c->mutex);
+
+ if (!c->connected) {
+ mutex_unlock(&c->mutex);
+ return;
+ }
+
+ err = pcr_modify(c, ipcr_break_modify, NULL, SUCCEED_ON_BUS_RESET);
+ if (err < 0)
+ cmp_error(c, "plug is still connected\n");
+
+ fw_iso_resources_free(&c->resources);
+
+ c->connected = false;
+
+ mutex_unlock(&c->mutex);
+}
+EXPORT_SYMBOL(cmp_connection_break);
diff --git a/sound/firewire/cmp.h b/sound/firewire/cmp.h
new file mode 100644
index 000000000000..f47de08feb12
--- /dev/null
+++ b/sound/firewire/cmp.h
@@ -0,0 +1,41 @@
+#ifndef SOUND_FIREWIRE_CMP_H_INCLUDED
+#define SOUND_FIREWIRE_CMP_H_INCLUDED
+
+#include <linux/mutex.h>
+#include <linux/types.h>
+#include "iso-resources.h"
+
+struct fw_unit;
+
+/**
+ * struct cmp_connection - manages an isochronous connection to a device
+ * @speed: the connection's actual speed
+ *
+ * This structure manages (using CMP) an isochronous stream from the local
+ * computer to a device's input plug (iPCR).
+ *
+ * There is no corresponding oPCR created on the local computer, so it is not
+ * possible to overlay connections on top of this one.
+ */
+struct cmp_connection {
+ int speed;
+ /* private: */
+ bool connected;
+ struct mutex mutex;
+ struct fw_iso_resources resources;
+ __be32 last_pcr_value;
+ unsigned int pcr_index;
+ unsigned int max_speed;
+};
+
+int cmp_connection_init(struct cmp_connection *connection,
+ struct fw_unit *unit,
+ unsigned int ipcr_index);
+void cmp_connection_destroy(struct cmp_connection *connection);
+
+int cmp_connection_establish(struct cmp_connection *connection,
+ unsigned int max_payload);
+int cmp_connection_update(struct cmp_connection *connection);
+void cmp_connection_break(struct cmp_connection *connection);
+
+#endif
diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c
new file mode 100644
index 000000000000..ec578b5ad8da
--- /dev/null
+++ b/sound/firewire/fcp.c
@@ -0,0 +1,224 @@
+/*
+ * Function Control Protocol (IEC 61883-1) helper functions
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/list.h>
+#include <linux/module.h>
+#include <linux/sched.h>
+#include <linux/spinlock.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+#include "fcp.h"
+#include "lib.h"
+
+#define CTS_AVC 0x00
+
+#define ERROR_RETRIES 3
+#define ERROR_DELAY_MS 5
+#define FCP_TIMEOUT_MS 125
+
+static DEFINE_SPINLOCK(transactions_lock);
+static LIST_HEAD(transactions);
+
+enum fcp_state {
+ STATE_PENDING,
+ STATE_BUS_RESET,
+ STATE_COMPLETE,
+};
+
+struct fcp_transaction {
+ struct list_head list;
+ struct fw_unit *unit;
+ void *response_buffer;
+ unsigned int response_size;
+ unsigned int response_match_bytes;
+ enum fcp_state state;
+ wait_queue_head_t wait;
+};
+
+/**
+ * fcp_avc_transaction - send an AV/C command and wait for its response
+ * @unit: a unit on the target device
+ * @command: a buffer containing the command frame; must be DMA-able
+ * @command_size: the size of @command
+ * @response: a buffer for the response frame
+ * @response_size: the maximum size of @response
+ * @response_match_bytes: a bitmap specifying the bytes used to detect the
+ * correct response frame
+ *
+ * This function sends a FCP command frame to the target and waits for the
+ * corresponding response frame to be returned.
+ *
+ * Because it is possible for multiple FCP transactions to be active at the
+ * same time, the correct response frame is detected by the value of certain
+ * bytes. These bytes must be set in @response before calling this function,
+ * and the corresponding bits must be set in @response_match_bytes.
+ *
+ * @command and @response can point to the same buffer.
+ *
+ * Asynchronous operation (INTERIM, NOTIFY) is not supported at the moment.
+ *
+ * Returns the actual size of the response frame, or a negative error code.
+ */
+int fcp_avc_transaction(struct fw_unit *unit,
+ const void *command, unsigned int command_size,
+ void *response, unsigned int response_size,
+ unsigned int response_match_bytes)
+{
+ struct fcp_transaction t;
+ int tcode, ret, tries = 0;
+
+ t.unit = unit;
+ t.response_buffer = response;
+ t.response_size = response_size;
+ t.response_match_bytes = response_match_bytes;
+ t.state = STATE_PENDING;
+ init_waitqueue_head(&t.wait);
+
+ spin_lock_irq(&transactions_lock);
+ list_add_tail(&t.list, &transactions);
+ spin_unlock_irq(&transactions_lock);
+
+ for (;;) {
+ tcode = command_size == 4 ? TCODE_WRITE_QUADLET_REQUEST
+ : TCODE_WRITE_BLOCK_REQUEST;
+ ret = snd_fw_transaction(t.unit, tcode,
+ CSR_REGISTER_BASE + CSR_FCP_COMMAND,
+ (void *)command, command_size);
+ if (ret < 0)
+ break;
+
+ wait_event_timeout(t.wait, t.state != STATE_PENDING,
+ msecs_to_jiffies(FCP_TIMEOUT_MS));
+
+ if (t.state == STATE_COMPLETE) {
+ ret = t.response_size;
+ break;
+ } else if (t.state == STATE_BUS_RESET) {
+ msleep(ERROR_DELAY_MS);
+ } else if (++tries >= ERROR_RETRIES) {
+ dev_err(&t.unit->device, "FCP command timed out\n");
+ ret = -EIO;
+ break;
+ }
+ }
+
+ spin_lock_irq(&transactions_lock);
+ list_del(&t.list);
+ spin_unlock_irq(&transactions_lock);
+
+ return ret;
+}
+EXPORT_SYMBOL(fcp_avc_transaction);
+
+/**
+ * fcp_bus_reset - inform the target handler about a bus reset
+ * @unit: the unit that might be used by fcp_avc_transaction()
+ *
+ * This function must be called from the driver's .update handler to inform
+ * the FCP transaction handler that a bus reset has happened. Any pending FCP
+ * transactions are retried.
+ */
+void fcp_bus_reset(struct fw_unit *unit)
+{
+ struct fcp_transaction *t;
+
+ spin_lock_irq(&transactions_lock);
+ list_for_each_entry(t, &transactions, list) {
+ if (t->unit == unit &&
+ t->state == STATE_PENDING) {
+ t->state = STATE_BUS_RESET;
+ wake_up(&t->wait);
+ }
+ }
+ spin_unlock_irq(&transactions_lock);
+}
+EXPORT_SYMBOL(fcp_bus_reset);
+
+/* checks whether the response matches the masked bytes in response_buffer */
+static bool is_matching_response(struct fcp_transaction *transaction,
+ const void *response, size_t length)
+{
+ const u8 *p1, *p2;
+ unsigned int mask, i;
+
+ p1 = response;
+ p2 = transaction->response_buffer;
+ mask = transaction->response_match_bytes;
+
+ for (i = 0; ; ++i) {
+ if ((mask & 1) && p1[i] != p2[i])
+ return false;
+ mask >>= 1;
+ if (!mask)
+ return true;
+ if (--length == 0)
+ return false;
+ }
+}
+
+static void fcp_response(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 fcp_transaction *t;
+ unsigned long flags;
+
+ if (length < 1 || (*(const u8 *)data & 0xf0) != CTS_AVC)
+ return;
+
+ spin_lock_irqsave(&transactions_lock, flags);
+ list_for_each_entry(t, &transactions, list) {
+ struct fw_device *device = fw_parent_device(t->unit);
+ if (device->card != card ||
+ device->generation != generation)
+ continue;
+ smp_rmb(); /* node_id vs. generation */
+ if (device->node_id != source)
+ continue;
+
+ if (t->state == STATE_PENDING &&
+ is_matching_response(t, data, length)) {
+ t->state = STATE_COMPLETE;
+ t->response_size = min((unsigned int)length,
+ t->response_size);
+ memcpy(t->response_buffer, data, t->response_size);
+ wake_up(&t->wait);
+ }
+ }
+ spin_unlock_irqrestore(&transactions_lock, flags);
+}
+
+static struct fw_address_handler response_register_handler = {
+ .length = 0x200,
+ .address_callback = fcp_response,
+};
+
+static int __init fcp_module_init(void)
+{
+ static const struct fw_address_region response_register_region = {
+ .start = CSR_REGISTER_BASE + CSR_FCP_RESPONSE,
+ .end = CSR_REGISTER_BASE + CSR_FCP_END,
+ };
+
+ fw_core_add_address_handler(&response_register_handler,
+ &response_register_region);
+
+ return 0;
+}
+
+static void __exit fcp_module_exit(void)
+{
+ WARN_ON(!list_empty(&transactions));
+ fw_core_remove_address_handler(&response_register_handler);
+}
+
+module_init(fcp_module_init);
+module_exit(fcp_module_exit);
diff --git a/sound/firewire/fcp.h b/sound/firewire/fcp.h
new file mode 100644
index 000000000000..86595688bd91
--- /dev/null
+++ b/sound/firewire/fcp.h
@@ -0,0 +1,12 @@
+#ifndef SOUND_FIREWIRE_FCP_H_INCLUDED
+#define SOUND_FIREWIRE_FCP_H_INCLUDED
+
+struct fw_unit;
+
+int fcp_avc_transaction(struct fw_unit *unit,
+ const void *command, unsigned int command_size,
+ void *response, unsigned int response_size,
+ unsigned int response_match_bytes);
+void fcp_bus_reset(struct fw_unit *unit);
+
+#endif
diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c
new file mode 100644
index 000000000000..775dbd5f3445
--- /dev/null
+++ b/sound/firewire/iso-resources.c
@@ -0,0 +1,232 @@
+/*
+ * isochronous resources helper functions
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/jiffies.h>
+#include <linux/mutex.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/spinlock.h>
+#include "iso-resources.h"
+
+/**
+ * fw_iso_resources_init - initializes a &struct fw_iso_resources
+ * @r: the resource manager to initialize
+ * @unit: the device unit for which the resources will be needed
+ *
+ * If the device does not support all channel numbers, change @r->channels_mask
+ * after calling this function.
+ */
+int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
+{
+ r->buffer = kmalloc(2 * 4, GFP_KERNEL);
+ if (!r->buffer)
+ return -ENOMEM;
+
+ r->channels_mask = ~0uLL;
+ r->unit = fw_unit_get(unit);
+ mutex_init(&r->mutex);
+ r->allocated = false;
+
+ return 0;
+}
+
+/**
+ * fw_iso_resources_destroy - destroy a resource manager
+ * @r: the resource manager that is no longer needed
+ */
+void fw_iso_resources_destroy(struct fw_iso_resources *r)
+{
+ WARN_ON(r->allocated);
+ kfree(r->buffer);
+ mutex_destroy(&r->mutex);
+ fw_unit_put(r->unit);
+}
+
+static unsigned int packet_bandwidth(unsigned int max_payload_bytes, int speed)
+{
+ unsigned int bytes, s400_bytes;
+
+ /* iso packets have three header quadlets and quadlet-aligned payload */
+ bytes = 3 * 4 + ALIGN(max_payload_bytes, 4);
+
+ /* convert to bandwidth units (quadlets at S1600 = bytes at S400) */
+ if (speed <= SCODE_400)
+ s400_bytes = bytes * (1 << (SCODE_400 - speed));
+ else
+ s400_bytes = DIV_ROUND_UP(bytes, 1 << (speed - SCODE_400));
+
+ return s400_bytes;
+}
+
+static int current_bandwidth_overhead(struct fw_card *card)
+{
+ /*
+ * Under the usual pessimistic assumption (cable length 4.5 m), the
+ * isochronous overhead for N cables is 1.797 µs + N * 0.494 µs, or
+ * 88.3 + N * 24.3 in bandwidth units.
+ *
+ * The calculation below tries to deduce N from the current gap count.
+ * If the gap count has been optimized by measuring the actual packet
+ * transmission time, this derived overhead should be near the actual
+ * overhead as well.
+ */
+ return card->gap_count < 63 ? card->gap_count * 97 / 10 + 89 : 512;
+}
+
+static int wait_isoch_resource_delay_after_bus_reset(struct fw_card *card)
+{
+ for (;;) {
+ s64 delay = (card->reset_jiffies + HZ) - get_jiffies_64();
+ if (delay <= 0)
+ return 0;
+ if (schedule_timeout_interruptible(delay) > 0)
+ return -ERESTARTSYS;
+ }
+}
+
+/**
+ * fw_iso_resources_allocate - allocate isochronous channel and bandwidth
+ * @r: the resource manager
+ * @max_payload_bytes: the amount of data (including CIP headers) per packet
+ * @speed: the speed (e.g., SCODE_400) at which the packets will be sent
+ *
+ * This function allocates one isochronous channel and enough bandwidth for the
+ * specified packet size.
+ *
+ * Returns the channel number that the caller must use for streaming, or
+ * a negative error code. Due to potentionally long delays, this function is
+ * interruptible and can return -ERESTARTSYS. On success, the caller is
+ * responsible for calling fw_iso_resources_update() on bus resets, and
+ * fw_iso_resources_free() when the resources are not longer needed.
+ */
+int fw_iso_resources_allocate(struct fw_iso_resources *r,
+ unsigned int max_payload_bytes, int speed)
+{
+ struct fw_card *card = fw_parent_device(r->unit)->card;
+ int bandwidth, channel, err;
+
+ if (WARN_ON(r->allocated))
+ return -EBADFD;
+
+ r->bandwidth = packet_bandwidth(max_payload_bytes, speed);
+
+retry_after_bus_reset:
+ spin_lock_irq(&card->lock);
+ r->generation = card->generation;
+ r->bandwidth_overhead = current_bandwidth_overhead(card);
+ spin_unlock_irq(&card->lock);
+
+ err = wait_isoch_resource_delay_after_bus_reset(card);
+ if (err < 0)
+ return err;
+
+ mutex_lock(&r->mutex);
+
+ bandwidth = r->bandwidth + r->bandwidth_overhead;
+ fw_iso_resource_manage(card, r->generation, r->channels_mask,
+ &channel, &bandwidth, true, r->buffer);
+ if (channel == -EAGAIN) {
+ mutex_unlock(&r->mutex);
+ goto retry_after_bus_reset;
+ }
+ if (channel >= 0) {
+ r->channel = channel;
+ r->allocated = true;
+ } else {
+ if (channel == -EBUSY)
+ dev_err(&r->unit->device,
+ "isochronous resources exhausted\n");
+ else
+ dev_err(&r->unit->device,
+ "isochronous resource allocation failed\n");
+ }
+
+ mutex_unlock(&r->mutex);
+
+ return channel;
+}
+
+/**
+ * fw_iso_resources_update - update resource allocations after a bus reset
+ * @r: the resource manager
+ *
+ * This function must be called from the driver's .update handler to reallocate
+ * any resources that were allocated before the bus reset. It is safe to call
+ * this function if no resources are currently allocated.
+ *
+ * Returns a negative error code on failure. If this happens, the caller must
+ * stop streaming.
+ */
+int fw_iso_resources_update(struct fw_iso_resources *r)
+{
+ struct fw_card *card = fw_parent_device(r->unit)->card;
+ int bandwidth, channel;
+
+ mutex_lock(&r->mutex);
+
+ if (!r->allocated) {
+ mutex_unlock(&r->mutex);
+ return 0;
+ }
+
+ spin_lock_irq(&card->lock);
+ r->generation = card->generation;
+ r->bandwidth_overhead = current_bandwidth_overhead(card);
+ spin_unlock_irq(&card->lock);
+
+ bandwidth = r->bandwidth + r->bandwidth_overhead;
+
+ fw_iso_resource_manage(card, r->generation, 1uLL << r->channel,
+ &channel, &bandwidth, true, r->buffer);
+ /*
+ * When another bus reset happens, pretend that the allocation
+ * succeeded; we will try again for the new generation later.
+ */
+ if (channel < 0 && channel != -EAGAIN) {
+ r->allocated = false;
+ if (channel == -EBUSY)
+ dev_err(&r->unit->device,
+ "isochronous resources exhausted\n");
+ else
+ dev_err(&r->unit->device,
+ "isochronous resource allocation failed\n");
+ }
+
+ mutex_unlock(&r->mutex);
+
+ return channel;
+}
+
+/**
+ * fw_iso_resources_free - frees allocated resources
+ * @r: the resource manager
+ *
+ * This function deallocates the channel and bandwidth, if allocated.
+ */
+void fw_iso_resources_free(struct fw_iso_resources *r)
+{
+ struct fw_card *card = fw_parent_device(r->unit)->card;
+ int bandwidth, channel;
+
+ mutex_lock(&r->mutex);
+
+ if (r->allocated) {
+ bandwidth = r->bandwidth + r->bandwidth_overhead;
+ fw_iso_resource_manage(card, r->generation, 1uLL << r->channel,
+ &channel, &bandwidth, false, r->buffer);
+ if (channel < 0)
+ dev_err(&r->unit->device,
+ "isochronous resource deallocation failed\n");
+
+ r->allocated = false;
+ }
+
+ mutex_unlock(&r->mutex);
+}
diff --git a/sound/firewire/iso-resources.h b/sound/firewire/iso-resources.h
new file mode 100644
index 000000000000..3f0730e4d841
--- /dev/null
+++ b/sound/firewire/iso-resources.h
@@ -0,0 +1,39 @@
+#ifndef SOUND_FIREWIRE_ISO_RESOURCES_H_INCLUDED
+#define SOUND_FIREWIRE_ISO_RESOURCES_H_INCLUDED
+
+#include <linux/mutex.h>
+#include <linux/types.h>
+
+struct fw_unit;
+
+/**
+ * struct fw_iso_resources - manages channel/bandwidth allocation
+ * @channels_mask: if the device does not support all channel numbers, set this
+ * bit mask to something else than the default (all ones)
+ *
+ * This structure manages (de)allocation of isochronous resources (channel and
+ * bandwidth) for one isochronous stream.
+ */
+struct fw_iso_resources {
+ u64 channels_mask;
+ /* private: */
+ struct fw_unit *unit;
+ struct mutex mutex;
+ unsigned int channel;
+ unsigned int bandwidth; /* in bandwidth units, without overhead */
+ unsigned int bandwidth_overhead;
+ int generation; /* in which allocation is valid */
+ bool allocated;
+ __be32 *buffer;
+};
+
+int fw_iso_resources_init(struct fw_iso_resources *r,
+ struct fw_unit *unit);
+void fw_iso_resources_destroy(struct fw_iso_resources *r);
+
+int fw_iso_resources_allocate(struct fw_iso_resources *r,
+ unsigned int max_payload_bytes, int speed);
+int fw_iso_resources_update(struct fw_iso_resources *r);
+void fw_iso_resources_free(struct fw_iso_resources *r);
+
+#endif
diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c
new file mode 100644
index 000000000000..4750cea2210e
--- /dev/null
+++ b/sound/firewire/lib.c
@@ -0,0 +1,85 @@
+/*
+ * miscellaneous helper functions
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/module.h>
+#include "lib.h"
+
+#define ERROR_RETRY_DELAY_MS 5
+
+/**
+ * rcode_string - convert a firewire result code to a string
+ * @rcode: the result
+ */
+const char *rcode_string(unsigned int rcode)
+{
+ static const char *const names[] = {
+ [RCODE_COMPLETE] = "complete",
+ [RCODE_CONFLICT_ERROR] = "conflict error",
+ [RCODE_DATA_ERROR] = "data error",
+ [RCODE_TYPE_ERROR] = "type error",
+ [RCODE_ADDRESS_ERROR] = "address error",
+ [RCODE_SEND_ERROR] = "send error",
+ [RCODE_CANCELLED] = "cancelled",
+ [RCODE_BUSY] = "busy",
+ [RCODE_GENERATION] = "generation",
+ [RCODE_NO_ACK] = "no ack",
+ };
+
+ if (rcode < ARRAY_SIZE(names) && names[rcode])
+ return names[rcode];
+ else
+ return "unknown";
+}
+EXPORT_SYMBOL(rcode_string);
+
+/**
+ * snd_fw_transaction - send a request and wait for its completion
+ * @unit: the driver's unit on the target device
+ * @tcode: the transaction code
+ * @offset: the address in the target's address space
+ * @buffer: input/output data
+ * @length: length of @buffer
+ *
+ * Submits an asynchronous request to the target device, and waits for the
+ * response. The node ID and the current generation are derived from @unit.
+ * On a bus reset or an error, the transaction is retried a few times.
+ * Returns zero on success, or a negative error code.
+ */
+int snd_fw_transaction(struct fw_unit *unit, int tcode,
+ u64 offset, void *buffer, size_t length)
+{
+ struct fw_device *device = fw_parent_device(unit);
+ int generation, rcode, tries = 0;
+
+ for (;;) {
+ generation = device->generation;
+ smp_rmb(); /* node_id vs. generation */
+ rcode = fw_run_transaction(device->card, tcode,
+ device->node_id, generation,
+ device->max_speed, offset,
+ buffer, length);
+
+ if (rcode == RCODE_COMPLETE)
+ return 0;
+
+ if (rcode_is_permanent_error(rcode) || ++tries >= 3) {
+ dev_err(&unit->device, "transaction failed: %s\n",
+ rcode_string(rcode));
+ return -EIO;
+ }
+
+ msleep(ERROR_RETRY_DELAY_MS);
+ }
+}
+EXPORT_SYMBOL(snd_fw_transaction);
+
+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
new file mode 100644
index 000000000000..064f3fd9ab06
--- /dev/null
+++ b/sound/firewire/lib.h
@@ -0,0 +1,19 @@
+#ifndef SOUND_FIREWIRE_LIB_H_INCLUDED
+#define SOUND_FIREWIRE_LIB_H_INCLUDED
+
+#include <linux/firewire-constants.h>
+#include <linux/types.h>
+
+struct fw_unit;
+
+int snd_fw_transaction(struct fw_unit *unit, int tcode,
+ u64 offset, void *buffer, size_t length);
+const char *rcode_string(unsigned int rcode);
+
+/* returns true if retrying the transaction would not make sense */
+static inline bool rcode_is_permanent_error(int rcode)
+{
+ return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR;
+}
+
+#endif
diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c
new file mode 100644
index 000000000000..1e20e60ba6a6
--- /dev/null
+++ b/sound/firewire/packets-buffer.c
@@ -0,0 +1,74 @@
+/*
+ * helpers for managing a buffer for many packets
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/firewire.h>
+#include <linux/slab.h>
+#include "packets-buffer.h"
+
+/**
+ * iso_packets_buffer_init - allocates the memory for packets
+ * @b: the buffer structure to initialize
+ * @unit: the device at the other end of the stream
+ * @count: the number of packets
+ * @packet_size: the (maximum) size of a packet, in bytes
+ * @direction: %DMA_TO_DEVICE or %DMA_FROM_DEVICE
+ */
+int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit,
+ unsigned int count, unsigned int packet_size,
+ enum dma_data_direction direction)
+{
+ unsigned int packets_per_page, pages;
+ unsigned int i, page_index, offset_in_page;
+ void *p;
+ int err;
+
+ b->packets = kmalloc(count * sizeof(*b->packets), GFP_KERNEL);
+ if (!b->packets) {
+ err = -ENOMEM;
+ goto error;
+ }
+
+ packet_size = L1_CACHE_ALIGN(packet_size);
+ packets_per_page = PAGE_SIZE / packet_size;
+ if (WARN_ON(!packets_per_page)) {
+ err = -EINVAL;
+ goto error;
+ }
+ pages = DIV_ROUND_UP(count, packets_per_page);
+
+ err = fw_iso_buffer_init(&b->iso_buffer, fw_parent_device(unit)->card,
+ pages, direction);
+ if (err < 0)
+ goto err_packets;
+
+ for (i = 0; i < count; ++i) {
+ page_index = i / packets_per_page;
+ p = page_address(b->iso_buffer.pages[page_index]);
+ offset_in_page = (i % packets_per_page) * packet_size;
+ b->packets[i].buffer = p + offset_in_page;
+ b->packets[i].offset = page_index * PAGE_SIZE + offset_in_page;
+ }
+
+ return 0;
+
+err_packets:
+ kfree(b->packets);
+error:
+ return err;
+}
+
+/**
+ * iso_packets_buffer_destroy - frees packet buffer resources
+ * @b: the buffer structure to free
+ * @unit: the device at the other end of the stream
+ */
+void iso_packets_buffer_destroy(struct iso_packets_buffer *b,
+ struct fw_unit *unit)
+{
+ fw_iso_buffer_destroy(&b->iso_buffer, fw_parent_device(unit)->card);
+ kfree(b->packets);
+}
diff --git a/sound/firewire/packets-buffer.h b/sound/firewire/packets-buffer.h
new file mode 100644
index 000000000000..6513c5cb6ea9
--- /dev/null
+++ b/sound/firewire/packets-buffer.h
@@ -0,0 +1,26 @@
+#ifndef SOUND_FIREWIRE_PACKETS_BUFFER_H_INCLUDED
+#define SOUND_FIREWIRE_PACKETS_BUFFER_H_INCLUDED
+
+#include <linux/dma-mapping.h>
+#include <linux/firewire.h>
+
+/**
+ * struct iso_packets_buffer - manages a buffer for many packets
+ * @iso_buffer: the memory containing the packets
+ * @packets: an array, with each element pointing to one packet
+ */
+struct iso_packets_buffer {
+ struct fw_iso_buffer iso_buffer;
+ struct {
+ void *buffer;
+ unsigned int offset;
+ } *packets;
+};
+
+int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit,
+ unsigned int count, unsigned int packet_size,
+ enum dma_data_direction direction);
+void iso_packets_buffer_destroy(struct iso_packets_buffer *b,
+ struct fw_unit *unit);
+
+#endif
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
new file mode 100644
index 000000000000..0fce9218abb1
--- /dev/null
+++ b/sound/firewire/speakers.c
@@ -0,0 +1,858 @@
+/*
+ * OXFW970-based speakers driver
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#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 <sound/control.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "cmp.h"
+#include "fcp.h"
+#include "amdtp.h"
+#include "lib.h"
+
+#define OXFORD_FIRMWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x50000)
+/* 0x970?vvvv or 0x971?vvvv, where vvvv = firmware version */
+
+#define OXFORD_HARDWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x90020)
+#define OXFORD_HARDWARE_ID_OXFW970 0x39443841
+#define OXFORD_HARDWARE_ID_OXFW971 0x39373100
+
+#define VENDOR_GRIFFIN 0x001292
+#define VENDOR_LACIE 0x00d04b
+
+#define SPECIFIER_1394TA 0x00a02d
+#define VERSION_AVC 0x010001
+
+struct device_info {
+ const char *driver_name;
+ const char *short_name;
+ const char *long_name;
+ int (*pcm_constraints)(struct snd_pcm_runtime *runtime);
+ unsigned int mixer_channels;
+ u8 mute_fb_id;
+ u8 volume_fb_id;
+};
+
+struct fwspk {
+ struct snd_card *card;
+ struct fw_unit *unit;
+ const struct device_info *device_info;
+ struct snd_pcm_substream *pcm;
+ struct mutex mutex;
+ struct cmp_connection connection;
+ struct amdtp_out_stream stream;
+ bool stream_running;
+ bool mute;
+ s16 volume[6];
+ s16 volume_min;
+ s16 volume_max;
+};
+
+MODULE_DESCRIPTION("FireWire speakers driver");
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_LICENSE("GPL v2");
+
+static int firewave_rate_constraint(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ static unsigned int stereo_rates[] = { 48000, 96000 };
+ struct snd_interval *channels =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *rate =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+
+ /* two channels work only at 48/96 kHz */
+ if (snd_interval_max(channels) < 6)
+ return snd_interval_list(rate, 2, stereo_rates, 0);
+ return 0;
+}
+
+static int firewave_channels_constraint(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ static const struct snd_interval all_channels = { .min = 6, .max = 6 };
+ struct snd_interval *rate =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* 32/44.1 kHz work only with all six channels */
+ if (snd_interval_max(rate) < 48000)
+ return snd_interval_refine(channels, &all_channels);
+ return 0;
+}
+
+static int firewave_constraints(struct snd_pcm_runtime *runtime)
+{
+ static unsigned int channels_list[] = { 2, 6 };
+ static struct snd_pcm_hw_constraint_list channels_list_constraint = {
+ .count = 2,
+ .list = channels_list,
+ };
+ int err;
+
+ runtime->hw.rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.channels_max = 6;
+
+ err = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &channels_list_constraint);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ firewave_rate_constraint, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ firewave_channels_constraint, NULL,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int lacie_speakers_constraints(struct snd_pcm_runtime *runtime)
+{
+ runtime->hw.rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+
+ return 0;
+}
+
+static int fwspk_open(struct snd_pcm_substream *substream)
+{
+ static const struct snd_pcm_hardware hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .formats = AMDTP_OUT_PCM_FORMAT_BITS,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 4 * 1024 * 1024,
+ .period_bytes_min = 1,
+ .period_bytes_max = UINT_MAX,
+ .periods_min = 1,
+ .periods_max = UINT_MAX,
+ };
+ struct fwspk *fwspk = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ runtime->hw = hardware;
+
+ err = fwspk->device_info->pcm_constraints(runtime);
+ if (err < 0)
+ return err;
+ err = snd_pcm_limit_hw_rates(runtime);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+ 5000, 8192000);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int fwspk_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static void fwspk_stop_stream(struct fwspk *fwspk)
+{
+ if (fwspk->stream_running) {
+ amdtp_out_stream_stop(&fwspk->stream);
+ cmp_connection_break(&fwspk->connection);
+ fwspk->stream_running = false;
+ }
+}
+
+static int fwspk_set_rate(struct fwspk *fwspk, unsigned int sfc)
+{
+ u8 *buf;
+ int err;
+
+ buf = kmalloc(8, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ buf[0] = 0x00; /* AV/C, CONTROL */
+ buf[1] = 0xff; /* unit */
+ buf[2] = 0x19; /* INPUT PLUG SIGNAL FORMAT */
+ buf[3] = 0x00; /* plug 0 */
+ buf[4] = 0x90; /* format: audio */
+ buf[5] = 0x00 | sfc; /* AM824, frequency */
+ buf[6] = 0xff; /* SYT (not used) */
+ buf[7] = 0xff;
+
+ err = fcp_avc_transaction(fwspk->unit, buf, 8, buf, 8,
+ BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5));
+ if (err < 0)
+ goto error;
+ if (err < 6 || buf[0] != 0x09 /* ACCEPTED */) {
+ dev_err(&fwspk->unit->device, "failed to set sample rate\n");
+ err = -EIO;
+ goto error;
+ }
+
+ err = 0;
+
+error:
+ kfree(buf);
+
+ return err;
+}
+
+static int fwspk_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct fwspk *fwspk = substream->private_data;
+ int err;
+
+ mutex_lock(&fwspk->mutex);
+ fwspk_stop_stream(fwspk);
+ mutex_unlock(&fwspk->mutex);
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ goto error;
+
+ amdtp_out_stream_set_rate(&fwspk->stream, params_rate(hw_params));
+ amdtp_out_stream_set_pcm(&fwspk->stream, params_channels(hw_params));
+
+ amdtp_out_stream_set_pcm_format(&fwspk->stream,
+ params_format(hw_params));
+
+ err = fwspk_set_rate(fwspk, fwspk->stream.sfc);
+ if (err < 0)
+ goto err_buffer;
+
+ return 0;
+
+err_buffer:
+ snd_pcm_lib_free_vmalloc_buffer(substream);
+error:
+ return err;
+}
+
+static int fwspk_hw_free(struct snd_pcm_substream *substream)
+{
+ struct fwspk *fwspk = substream->private_data;
+
+ mutex_lock(&fwspk->mutex);
+ fwspk_stop_stream(fwspk);
+ mutex_unlock(&fwspk->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int fwspk_prepare(struct snd_pcm_substream *substream)
+{
+ struct fwspk *fwspk = substream->private_data;
+ int err;
+
+ mutex_lock(&fwspk->mutex);
+
+ if (amdtp_out_streaming_error(&fwspk->stream))
+ fwspk_stop_stream(fwspk);
+
+ if (!fwspk->stream_running) {
+ err = cmp_connection_establish(&fwspk->connection,
+ amdtp_out_stream_get_max_payload(&fwspk->stream));
+ if (err < 0)
+ goto err_mutex;
+
+ err = amdtp_out_stream_start(&fwspk->stream,
+ fwspk->connection.resources.channel,
+ fwspk->connection.speed);
+ if (err < 0)
+ goto err_connection;
+
+ fwspk->stream_running = true;
+ }
+
+ mutex_unlock(&fwspk->mutex);
+
+ amdtp_out_stream_pcm_prepare(&fwspk->stream);
+
+ return 0;
+
+err_connection:
+ cmp_connection_break(&fwspk->connection);
+err_mutex:
+ mutex_unlock(&fwspk->mutex);
+
+ return err;
+}
+
+static int fwspk_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct fwspk *fwspk = substream->private_data;
+ struct snd_pcm_substream *pcm;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ pcm = substream;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pcm = NULL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ amdtp_out_stream_pcm_trigger(&fwspk->stream, pcm);
+ return 0;
+}
+
+static snd_pcm_uframes_t fwspk_pointer(struct snd_pcm_substream *substream)
+{
+ struct fwspk *fwspk = substream->private_data;
+
+ return amdtp_out_stream_pcm_pointer(&fwspk->stream);
+}
+
+static int fwspk_create_pcm(struct fwspk *fwspk)
+{
+ static struct snd_pcm_ops ops = {
+ .open = fwspk_open,
+ .close = fwspk_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = fwspk_hw_params,
+ .hw_free = fwspk_hw_free,
+ .prepare = fwspk_prepare,
+ .trigger = fwspk_trigger,
+ .pointer = fwspk_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+ };
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(fwspk->card, "OXFW970", 0, 1, 0, &pcm);
+ if (err < 0)
+ return err;
+ pcm->private_data = fwspk;
+ strcpy(pcm->name, fwspk->device_info->short_name);
+ fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ fwspk->pcm->ops = &ops;
+ return 0;
+}
+
+enum control_action { CTL_READ, CTL_WRITE };
+enum control_attribute {
+ CTL_MIN = 0x02,
+ CTL_MAX = 0x03,
+ CTL_CURRENT = 0x10,
+};
+
+static int fwspk_mute_command(struct fwspk *fwspk, bool *value,
+ enum control_action action)
+{
+ u8 *buf;
+ u8 response_ok;
+ int err;
+
+ buf = kmalloc(11, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ if (action == CTL_READ) {
+ buf[0] = 0x01; /* AV/C, STATUS */
+ response_ok = 0x0c; /* STABLE */
+ } else {
+ buf[0] = 0x00; /* AV/C, CONTROL */
+ response_ok = 0x09; /* ACCEPTED */
+ }
+ buf[1] = 0x08; /* audio unit 0 */
+ buf[2] = 0xb8; /* FUNCTION BLOCK */
+ buf[3] = 0x81; /* function block type: feature */
+ buf[4] = fwspk->device_info->mute_fb_id; /* function block ID */
+ buf[5] = 0x10; /* control attribute: current */
+ buf[6] = 0x02; /* selector length */
+ buf[7] = 0x00; /* audio channel number */
+ buf[8] = 0x01; /* control selector: mute */
+ buf[9] = 0x01; /* control data length */
+ if (action == CTL_READ)
+ buf[10] = 0xff;
+ else
+ buf[10] = *value ? 0x70 : 0x60;
+
+ err = fcp_avc_transaction(fwspk->unit, buf, 11, buf, 11, 0x3fe);
+ if (err < 0)
+ goto error;
+ if (err < 11) {
+ dev_err(&fwspk->unit->device, "short FCP response\n");
+ err = -EIO;
+ goto error;
+ }
+ if (buf[0] != response_ok) {
+ dev_err(&fwspk->unit->device, "mute command failed\n");
+ err = -EIO;
+ goto error;
+ }
+ if (action == CTL_READ)
+ *value = buf[10] == 0x70;
+
+ err = 0;
+
+error:
+ kfree(buf);
+
+ return err;
+}
+
+static int fwspk_volume_command(struct fwspk *fwspk, s16 *value,
+ unsigned int channel,
+ enum control_attribute attribute,
+ enum control_action action)
+{
+ u8 *buf;
+ u8 response_ok;
+ int err;
+
+ buf = kmalloc(12, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ if (action == CTL_READ) {
+ buf[0] = 0x01; /* AV/C, STATUS */
+ response_ok = 0x0c; /* STABLE */
+ } else {
+ buf[0] = 0x00; /* AV/C, CONTROL */
+ response_ok = 0x09; /* ACCEPTED */
+ }
+ buf[1] = 0x08; /* audio unit 0 */
+ buf[2] = 0xb8; /* FUNCTION BLOCK */
+ buf[3] = 0x81; /* function block type: feature */
+ buf[4] = fwspk->device_info->volume_fb_id; /* function block ID */
+ buf[5] = attribute; /* control attribute */
+ buf[6] = 0x02; /* selector length */
+ buf[7] = channel; /* audio channel number */
+ buf[8] = 0x02; /* control selector: volume */
+ buf[9] = 0x02; /* control data length */
+ if (action == CTL_READ) {
+ buf[10] = 0xff;
+ buf[11] = 0xff;
+ } else {
+ buf[10] = *value >> 8;
+ buf[11] = *value;
+ }
+
+ err = fcp_avc_transaction(fwspk->unit, buf, 12, buf, 12, 0x3fe);
+ if (err < 0)
+ goto error;
+ if (err < 12) {
+ dev_err(&fwspk->unit->device, "short FCP response\n");
+ err = -EIO;
+ goto error;
+ }
+ if (buf[0] != response_ok) {
+ dev_err(&fwspk->unit->device, "volume command failed\n");
+ err = -EIO;
+ goto error;
+ }
+ if (action == CTL_READ)
+ *value = (buf[10] << 8) | buf[11];
+
+ err = 0;
+
+error:
+ kfree(buf);
+
+ return err;
+}
+
+static int fwspk_mute_get(struct snd_kcontrol *control,
+ struct snd_ctl_elem_value *value)
+{
+ struct fwspk *fwspk = control->private_data;
+
+ value->value.integer.value[0] = !fwspk->mute;
+
+ return 0;
+}
+
+static int fwspk_mute_put(struct snd_kcontrol *control,
+ struct snd_ctl_elem_value *value)
+{
+ struct fwspk *fwspk = control->private_data;
+ bool mute;
+ int err;
+
+ mute = !value->value.integer.value[0];
+
+ if (mute == fwspk->mute)
+ return 0;
+
+ err = fwspk_mute_command(fwspk, &mute, CTL_WRITE);
+ if (err < 0)
+ return err;
+ fwspk->mute = mute;
+
+ return 1;
+}
+
+static int fwspk_volume_info(struct snd_kcontrol *control,
+ struct snd_ctl_elem_info *info)
+{
+ struct fwspk *fwspk = control->private_data;
+
+ info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ info->count = fwspk->device_info->mixer_channels;
+ info->value.integer.min = fwspk->volume_min;
+ info->value.integer.max = fwspk->volume_max;
+
+ return 0;
+}
+
+static const u8 channel_map[6] = { 0, 1, 4, 5, 2, 3 };
+
+static int fwspk_volume_get(struct snd_kcontrol *control,
+ struct snd_ctl_elem_value *value)
+{
+ struct fwspk *fwspk = control->private_data;
+ unsigned int i;
+
+ for (i = 0; i < fwspk->device_info->mixer_channels; ++i)
+ value->value.integer.value[channel_map[i]] = fwspk->volume[i];
+
+ return 0;
+}
+
+static int fwspk_volume_put(struct snd_kcontrol *control,
+ struct snd_ctl_elem_value *value)
+{
+ struct fwspk *fwspk = control->private_data;
+ unsigned int i, changed_channels;
+ bool equal_values = true;
+ s16 volume;
+ int err;
+
+ for (i = 0; i < fwspk->device_info->mixer_channels; ++i) {
+ if (value->value.integer.value[i] < fwspk->volume_min ||
+ value->value.integer.value[i] > fwspk->volume_max)
+ return -EINVAL;
+ if (value->value.integer.value[i] !=
+ value->value.integer.value[0])
+ equal_values = false;
+ }
+
+ changed_channels = 0;
+ for (i = 0; i < fwspk->device_info->mixer_channels; ++i)
+ if (value->value.integer.value[channel_map[i]] !=
+ fwspk->volume[i])
+ changed_channels |= 1 << (i + 1);
+
+ if (equal_values && changed_channels != 0)
+ changed_channels = 1 << 0;
+
+ for (i = 0; i <= fwspk->device_info->mixer_channels; ++i) {
+ volume = value->value.integer.value[channel_map[i ? i - 1 : 0]];
+ if (changed_channels & (1 << i)) {
+ err = fwspk_volume_command(fwspk, &volume, i,
+ CTL_CURRENT, CTL_WRITE);
+ if (err < 0)
+ return err;
+ }
+ if (i > 0)
+ fwspk->volume[i - 1] = volume;
+ }
+
+ return changed_channels != 0;
+}
+
+static int fwspk_create_mixer(struct fwspk *fwspk)
+{
+ static const struct snd_kcontrol_new controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = fwspk_mute_get,
+ .put = fwspk_mute_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Volume",
+ .info = fwspk_volume_info,
+ .get = fwspk_volume_get,
+ .put = fwspk_volume_put,
+ },
+ };
+ unsigned int i, first_ch;
+ int err;
+
+ err = fwspk_volume_command(fwspk, &fwspk->volume_min,
+ 0, CTL_MIN, CTL_READ);
+ if (err < 0)
+ return err;
+ err = fwspk_volume_command(fwspk, &fwspk->volume_max,
+ 0, CTL_MAX, CTL_READ);
+ if (err < 0)
+ return err;
+
+ err = fwspk_mute_command(fwspk, &fwspk->mute, CTL_READ);
+ if (err < 0)
+ return err;
+
+ first_ch = fwspk->device_info->mixer_channels == 1 ? 0 : 1;
+ for (i = 0; i < fwspk->device_info->mixer_channels; ++i) {
+ err = fwspk_volume_command(fwspk, &fwspk->volume[i],
+ first_ch + i, CTL_CURRENT, CTL_READ);
+ if (err < 0)
+ return err;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(controls); ++i) {
+ err = snd_ctl_add(fwspk->card,
+ snd_ctl_new1(&controls[i], fwspk));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static u32 fwspk_read_firmware_version(struct fw_unit *unit)
+{
+ __be32 data;
+ int err;
+
+ err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST,
+ OXFORD_FIRMWARE_ID_ADDRESS, &data, 4);
+ return err >= 0 ? be32_to_cpu(data) : 0;
+}
+
+static void fwspk_card_free(struct snd_card *card)
+{
+ struct fwspk *fwspk = card->private_data;
+ struct fw_device *dev = fw_parent_device(fwspk->unit);
+
+ amdtp_out_stream_destroy(&fwspk->stream);
+ cmp_connection_destroy(&fwspk->connection);
+ fw_unit_put(fwspk->unit);
+ fw_device_put(dev);
+ mutex_destroy(&fwspk->mutex);
+}
+
+static const struct device_info *__devinit fwspk_detect(struct fw_device *dev)
+{
+ static const struct device_info griffin_firewave = {
+ .driver_name = "FireWave",
+ .short_name = "FireWave",
+ .long_name = "Griffin FireWave Surround",
+ .pcm_constraints = firewave_constraints,
+ .mixer_channels = 6,
+ .mute_fb_id = 0x01,
+ .volume_fb_id = 0x02,
+ };
+ static const struct device_info lacie_speakers = {
+ .driver_name = "FWSpeakers",
+ .short_name = "FireWire Speakers",
+ .long_name = "LaCie FireWire Speakers",
+ .pcm_constraints = lacie_speakers_constraints,
+ .mixer_channels = 1,
+ .mute_fb_id = 0x01,
+ .volume_fb_id = 0x01,
+ };
+ struct fw_csr_iterator i;
+ int key, value;
+
+ fw_csr_iterator_init(&i, dev->config_rom);
+ while (fw_csr_iterator_next(&i, &key, &value))
+ if (key == CSR_VENDOR)
+ switch (value) {
+ case VENDOR_GRIFFIN:
+ return &griffin_firewave;
+ case VENDOR_LACIE:
+ return &lacie_speakers;
+ }
+
+ return NULL;
+}
+
+static int __devinit fwspk_probe(struct device *unit_dev)
+{
+ struct fw_unit *unit = fw_unit(unit_dev);
+ struct fw_device *fw_dev = fw_parent_device(unit);
+ struct snd_card *card;
+ struct fwspk *fwspk;
+ u32 firmware;
+ int err;
+
+ err = snd_card_create(-1, NULL, THIS_MODULE, sizeof(*fwspk), &card);
+ if (err < 0)
+ return err;
+ snd_card_set_dev(card, unit_dev);
+
+ fwspk = card->private_data;
+ fwspk->card = card;
+ mutex_init(&fwspk->mutex);
+ fw_device_get(fw_dev);
+ fwspk->unit = fw_unit_get(unit);
+ fwspk->device_info = fwspk_detect(fw_dev);
+ if (!fwspk->device_info) {
+ err = -ENODEV;
+ goto err_unit;
+ }
+
+ err = cmp_connection_init(&fwspk->connection, unit, 0);
+ if (err < 0)
+ goto err_unit;
+
+ err = amdtp_out_stream_init(&fwspk->stream, unit, CIP_NONBLOCKING);
+ if (err < 0)
+ goto err_connection;
+
+ card->private_free = fwspk_card_free;
+
+ strcpy(card->driver, fwspk->device_info->driver_name);
+ strcpy(card->shortname, fwspk->device_info->short_name);
+ firmware = fwspk_read_firmware_version(unit);
+ snprintf(card->longname, sizeof(card->longname),
+ "%s (OXFW%x %04x), GUID %08x%08x at %s, S%d",
+ fwspk->device_info->long_name,
+ firmware >> 20, firmware & 0xffff,
+ fw_dev->config_rom[3], fw_dev->config_rom[4],
+ dev_name(&unit->device), 100 << fw_dev->max_speed);
+ strcpy(card->mixername, "OXFW970");
+
+ err = fwspk_create_pcm(fwspk);
+ if (err < 0)
+ goto error;
+
+ err = fwspk_create_mixer(fwspk);
+ if (err < 0)
+ goto error;
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ dev_set_drvdata(unit_dev, fwspk);
+
+ return 0;
+
+err_connection:
+ cmp_connection_destroy(&fwspk->connection);
+err_unit:
+ fw_unit_put(fwspk->unit);
+ fw_device_put(fw_dev);
+ mutex_destroy(&fwspk->mutex);
+error:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit fwspk_remove(struct device *dev)
+{
+ struct fwspk *fwspk = dev_get_drvdata(dev);
+
+ snd_card_disconnect(fwspk->card);
+
+ mutex_lock(&fwspk->mutex);
+ amdtp_out_stream_pcm_abort(&fwspk->stream);
+ fwspk_stop_stream(fwspk);
+ mutex_unlock(&fwspk->mutex);
+
+ snd_card_free_when_closed(fwspk->card);
+
+ return 0;
+}
+
+static void fwspk_bus_reset(struct fw_unit *unit)
+{
+ struct fwspk *fwspk = dev_get_drvdata(&unit->device);
+
+ fcp_bus_reset(fwspk->unit);
+
+ if (cmp_connection_update(&fwspk->connection) < 0) {
+ mutex_lock(&fwspk->mutex);
+ amdtp_out_stream_pcm_abort(&fwspk->stream);
+ fwspk_stop_stream(fwspk);
+ mutex_unlock(&fwspk->mutex);
+ return;
+ }
+
+ amdtp_out_stream_update(&fwspk->stream);
+}
+
+static const struct ieee1394_device_id fwspk_id_table[] = {
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = VENDOR_GRIFFIN,
+ .model_id = 0x00f970,
+ .specifier_id = SPECIFIER_1394TA,
+ .version = VERSION_AVC,
+ },
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = VENDOR_LACIE,
+ .model_id = 0x00f970,
+ .specifier_id = SPECIFIER_1394TA,
+ .version = VERSION_AVC,
+ },
+ { }
+};
+MODULE_DEVICE_TABLE(ieee1394, fwspk_id_table);
+
+static struct fw_driver fwspk_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = KBUILD_MODNAME,
+ .bus = &fw_bus_type,
+ .probe = fwspk_probe,
+ .remove = __devexit_p(fwspk_remove),
+ },
+ .update = fwspk_bus_reset,
+ .id_table = fwspk_id_table,
+};
+
+static int __init alsa_fwspk_init(void)
+{
+ return driver_register(&fwspk_driver.driver);
+}
+
+static void __exit alsa_fwspk_exit(void)
+{
+ driver_unregister(&fwspk_driver.driver);
+}
+
+module_init(alsa_fwspk_init);
+module_exit(alsa_fwspk_exit);
diff --git a/sound/oss/Makefile b/sound/oss/Makefile
index 96f14dcd0cd1..90ffb99c6b17 100644
--- a/sound/oss/Makefile
+++ b/sound/oss/Makefile
@@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y)
$(obj)/bin2hex pss_synth < $< > $@
else
$(obj)/pss_boot.h:
- ( \
+ $(Q)( \
echo 'static unsigned char * pss_synth = NULL;'; \
echo 'static int pss_synthLen = 0;'; \
) > $@
@@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y)
$(obj)/hex2hex -i trix_boot < $< > $@
else
$(obj)/trix_boot.h:
- ( \
+ $(Q)( \
echo 'static unsigned char * trix_boot = NULL;'; \
echo 'static int trix_boot_len = 0;'; \
) > $@
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 23f49f356e0f..16c0bdfbb164 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) {
static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma)
{
stream_t *dma = &vortex->dma_adb[adbdma];
- int temp;
+ int temp, page, delta;
temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
- temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
- return temp;
+ page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT;
+ if (dma->nr_periods >= 4)
+ delta = (page - dma->period_real) & 3;
+ else {
+ delta = (page - dma->period_real);
+ if (delta < 0)
+ delta += dma->nr_periods;
+ }
+ return (dma->period_virt + delta) * dma->period_bytes
+ + (temp & (dma->period_bytes - 1));
}
static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 6117595fc075..573594bf3225 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -979,31 +979,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
snd_azf3328_dbgcallenter();
switch (bitrate) {
-#define AZF_FMT_XLATE(in_freq, out_bits) \
- do { \
- case AZF_FREQ_ ## in_freq: \
- freq = SOUNDFORMAT_FREQ_ ## out_bits; \
- break; \
- } while (0);
- AZF_FMT_XLATE(4000, SUSPECTED_4000)
- AZF_FMT_XLATE(4800, SUSPECTED_4800)
- /* the AZF3328 names it "5510" for some strange reason: */
- AZF_FMT_XLATE(5512, 5510)
- AZF_FMT_XLATE(6620, 6620)
- AZF_FMT_XLATE(8000, 8000)
- AZF_FMT_XLATE(9600, 9600)
- AZF_FMT_XLATE(11025, 11025)
- AZF_FMT_XLATE(13240, SUSPECTED_13240)
- AZF_FMT_XLATE(16000, 16000)
- AZF_FMT_XLATE(22050, 22050)
- AZF_FMT_XLATE(32000, 32000)
+ case AZF_FREQ_4000: freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+ case AZF_FREQ_4800: freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+ case AZF_FREQ_5512:
+ /* the AZF3328 names it "5510" for some strange reason */
+ freq = SOUNDFORMAT_FREQ_5510; break;
+ case AZF_FREQ_6620: freq = SOUNDFORMAT_FREQ_6620; break;
+ case AZF_FREQ_8000: freq = SOUNDFORMAT_FREQ_8000; break;
+ case AZF_FREQ_9600: freq = SOUNDFORMAT_FREQ_9600; break;
+ case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break;
+ case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+ case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break;
+ case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break;
+ case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
default:
snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
/* fall-through */
- AZF_FMT_XLATE(44100, 44100)
- AZF_FMT_XLATE(48000, 48000)
- AZF_FMT_XLATE(66200, SUSPECTED_66200)
-#undef AZF_FMT_XLATE
+ case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
+ case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
+ case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
}
/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 4a663471dadc..74b0560289c0 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
snd_print_pcm_rates(a->rates, buf, sizeof(buf));
if (a->format == AUDIO_CODING_TYPE_LPCM)
- snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8));
+ snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8);
else if (a->max_bitrate)
snprintf(buf2, sizeof(buf2),
", max bitrate = %d", a->max_bitrate);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2e91a991eb15..fcedad9a5fef 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
@@ -2703,7 +2704,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
if (err < 0)
goto out_free;
#ifdef CONFIG_SND_HDA_PATCH_LOADER
- if (patch[dev]) {
+ if (patch[dev] && *patch[dev]) {
snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n",
patch[dev]);
err = snd_hda_load_patch(chip->bus, patch[dev]);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index a07b031090d8..067982f4f182 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1039,9 +1039,11 @@ static struct hda_verb cs_errata_init_verbs[] = {
{0x11, AC_VERB_SET_PROC_COEF, 0x0008},
{0x11, AC_VERB_SET_PROC_STATE, 0x00},
+#if 0 /* Don't to set to D3 as we are in power-up sequence */
{0x07, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Rx: D3 */
{0x08, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Tx: D3 */
/*{0x01, AC_VERB_SET_POWER_STATE, 0x03},*/ /* AFG: D3 This is already handled */
+#endif
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9bb030a469cd..4d5004e693f0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -85,6 +85,7 @@ struct conexant_spec {
unsigned int auto_mic;
int auto_mic_ext; /* autocfg.inputs[] index for ext mic */
unsigned int need_dac_fix;
+ hda_nid_t slave_dig_outs[2];
/* capture */
unsigned int num_adc_nids;
@@ -127,6 +128,7 @@ struct conexant_spec {
unsigned int ideapad:1;
unsigned int thinkpad:1;
unsigned int hp_laptop:1;
+ unsigned int asus:1;
unsigned int ext_mic_present;
unsigned int recording;
@@ -352,6 +354,8 @@ static int conexant_build_pcms(struct hda_codec *codec)
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
spec->dig_in_nid;
}
+ if (spec->slave_dig_outs[0])
+ codec->slave_dig_outs = spec->slave_dig_outs;
}
return 0;
@@ -403,10 +407,16 @@ static int conexant_add_jack(struct hda_codec *codec,
struct conexant_spec *spec;
struct conexant_jack *jack;
const char *name;
- int err;
+ int i, err;
spec = codec->spec;
snd_array_init(&spec->jacks, sizeof(*jack), 32);
+
+ jack = spec->jacks.list;
+ for (i = 0; i < spec->jacks.used; i++, jack++)
+ if (jack->nid == nid)
+ return 0 ; /* already present */
+
jack = snd_array_new(&spec->jacks);
name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ;
@@ -2100,7 +2110,7 @@ static int patch_cxt5051(struct hda_codec *codec)
static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
-#define CXT5066_SPDIF_OUT 0x21
+static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
/* OLPC's microphone port is DC coupled for use with external sensors,
* therefore we use a 50% mic bias in order to center the input signal with
@@ -2312,6 +2322,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
}
}
+
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_asus_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_jack_detect(codec, 0x1b);
+ snd_printdd("CXT5066: external microphone present=%d\n", present);
+ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 1 : 0);
+}
+
+
/* toggle input of built-in digital mic and mic jack appropriately */
static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
{
@@ -2387,79 +2410,55 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
cxt5066_update_speaker(codec);
}
-/* unsolicited event for jack sensing */
-static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
+/* Dispatch the right mic autoswitch function */
+static void cxt5066_automic(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
- snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
- /* ignore mic events in DC mode; we're always using the jack */
- if (!spec->dc_enable)
- cxt5066_olpc_automic(codec);
- break;
- }
-}
-/* unsolicited event for jack sensing */
-static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res)
-{
- snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
+ if (spec->dell_vostro)
cxt5066_vostro_automic(codec);
- break;
- }
-}
-
-/* unsolicited event for jack sensing */
-static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res)
-{
- snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
+ else if (spec->ideapad)
cxt5066_ideapad_automic(codec);
- break;
- }
+ else if (spec->thinkpad)
+ cxt5066_thinkpad_automic(codec);
+ else if (spec->hp_laptop)
+ cxt5066_hp_laptop_automic(codec);
+ else if (spec->asus)
+ cxt5066_asus_automic(codec);
}
/* unsolicited event for jack sensing */
-static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
{
- snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26);
+ struct conexant_spec *spec = codec->spec;
+ snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
switch (res >> 26) {
case CONEXANT_HP_EVENT:
cxt5066_hp_automute(codec);
break;
case CONEXANT_MIC_EVENT:
- cxt5066_hp_laptop_automic(codec);
+ /* ignore mic events in DC mode; we're always using the jack */
+ if (!spec->dc_enable)
+ cxt5066_olpc_automic(codec);
break;
}
}
/* unsolicited event for jack sensing */
-static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
{
- snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26);
+ snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
switch (res >> 26) {
case CONEXANT_HP_EVENT:
cxt5066_hp_automute(codec);
break;
case CONEXANT_MIC_EVENT:
- cxt5066_thinkpad_automic(codec);
+ cxt5066_automic(codec);
break;
}
}
+
static const struct hda_input_mux cxt5066_analog_mic_boost = {
.num_items = 5,
.items = {
@@ -2633,6 +2632,27 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
spec->recording = 0;
}
+static void conexant_check_dig_outs(struct hda_codec *codec,
+ hda_nid_t *dig_pins,
+ int num_pins)
+{
+ struct conexant_spec *spec = codec->spec;
+ hda_nid_t *nid_loc = &spec->multiout.dig_out_nid;
+ int i;
+
+ for (i = 0; i < num_pins; i++, dig_pins++) {
+ unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins);
+ if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE)
+ continue;
+ if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1)
+ continue;
+ if (spec->slave_dig_outs[0])
+ nid_loc++;
+ else
+ nid_loc = spec->slave_dig_outs;
+ }
+}
+
static struct hda_input_mux cxt5066_capture_source = {
.num_items = 4,
.items = {
@@ -3039,20 +3059,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
/* initialize jack-sensing, too */
static int cxt5066_init(struct hda_codec *codec)
{
- struct conexant_spec *spec = codec->spec;
-
snd_printdd("CXT5066: init\n");
conexant_init(codec);
if (codec->patch_ops.unsol_event) {
cxt5066_hp_automute(codec);
- if (spec->dell_vostro)
- cxt5066_vostro_automic(codec);
- else if (spec->ideapad)
- cxt5066_ideapad_automic(codec);
- else if (spec->thinkpad)
- cxt5066_thinkpad_automic(codec);
- else if (spec->hp_laptop)
- cxt5066_hp_laptop_automic(codec);
+ cxt5066_automic(codec);
}
cxt5066_set_mic_boost(codec);
return 0;
@@ -3080,6 +3091,7 @@ enum {
CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */
CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */
CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */
+ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */
CXT5066_HP_LAPTOP, /* HP Laptop */
CXT5066_MODELS
};
@@ -3091,6 +3103,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
[CXT5066_DELL_VOSTRO] = "dell-vostro",
[CXT5066_IDEAPAD] = "ideapad",
[CXT5066_THINKPAD] = "thinkpad",
+ [CXT5066_ASUS] = "asus",
[CXT5066_HP_LAPTOP] = "hp-laptop",
};
@@ -3101,8 +3114,12 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
+ SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
+ SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
@@ -3111,7 +3128,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
{}
};
@@ -3133,7 +3152,8 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->multiout.max_channels = 2;
spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
spec->multiout.dac_nids = cxt5066_dac_nids;
- spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT;
+ conexant_check_dig_outs(codec, cxt5066_digout_pin_nids,
+ ARRAY_SIZE(cxt5066_digout_pin_nids));
spec->num_adc_nids = 1;
spec->adc_nids = cxt5066_adc_nids;
spec->capsrc_nids = cxt5066_capsrc_nids;
@@ -3167,17 +3187,20 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->num_init_verbs++;
spec->dell_automute = 1;
break;
+ case CXT5066_ASUS:
case CXT5066_HP_LAPTOP:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_hp_laptop_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->init_verbs[spec->num_init_verbs] =
cxt5066_init_verbs_hp_laptop;
spec->num_init_verbs++;
- spec->hp_laptop = 1;
+ spec->hp_laptop = board_config == CXT5066_HP_LAPTOP;
+ spec->asus = board_config == CXT5066_ASUS;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
/* no S/PDIF out */
- spec->multiout.dig_out_nid = 0;
+ if (board_config == CXT5066_HP_LAPTOP)
+ spec->multiout.dig_out_nid = 0;
/* input source automatically selected */
spec->input_mux = NULL;
spec->port_d_mode = 0;
@@ -3207,7 +3230,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_DELL_VOSTRO:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_vostro_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->init_verbs[0] = cxt5066_init_verbs_vostro;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
@@ -3224,7 +3247,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_IDEAPAD:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_ideapad_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->init_verbs[0] = cxt5066_init_verbs_ideapad;
@@ -3240,7 +3263,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_THINKPAD:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_thinkpad_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->init_verbs[0] = cxt5066_init_verbs_thinkpad;
@@ -3389,7 +3412,7 @@ static void cx_auto_parse_output(struct hda_codec *codec)
}
}
spec->multiout.dac_nids = spec->private_dac_nids;
- spec->multiout.max_channels = nums * 2;
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (cfg->hp_outs > 0)
spec->auto_mute = 1;
@@ -3708,9 +3731,9 @@ static int cx_auto_init(struct hda_codec *codec)
return 0;
}
-static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
+static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
const char *dir, int cidx,
- hda_nid_t nid, int hda_dir)
+ hda_nid_t nid, int hda_dir, int amp_idx)
{
static char name[32];
static struct snd_kcontrol_new knew[] = {
@@ -3722,7 +3745,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
for (i = 0; i < 2; i++) {
struct snd_kcontrol *kctl;
- knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir);
+ knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+ hda_dir);
knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
knew[i].index = cidx;
snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]);
@@ -3738,6 +3762,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
return 0;
}
+#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \
+ cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+
#define cx_auto_add_pb_volume(codec, nid, str, idx) \
cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
@@ -3787,29 +3814,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
struct conexant_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
static const char *prev_label;
- int i, err, cidx;
+ int i, err, cidx, conn_len;
+ hda_nid_t conn[HDA_MAX_CONNECTIONS];
+
+ int multi_adc_volume = 0; /* If the ADC nid has several input volumes */
+ int adc_nid = spec->adc_nids[0];
+
+ conn_len = snd_hda_get_connections(codec, adc_nid, conn,
+ HDA_MAX_CONNECTIONS);
+ if (conn_len < 0)
+ return conn_len;
+
+ multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1;
+ if (!multi_adc_volume) {
+ err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid,
+ HDA_INPUT);
+ if (err < 0)
+ return err;
+ }
- err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0],
- HDA_INPUT);
- if (err < 0)
- return err;
prev_label = NULL;
cidx = 0;
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t nid = cfg->inputs[i].pin;
const char *label;
- if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+ int j;
+ int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP;
+ if (!pin_amp && !multi_adc_volume)
continue;
+
label = hda_get_autocfg_input_label(codec, cfg, i);
if (label == prev_label)
cidx++;
else
cidx = 0;
prev_label = label;
- err = cx_auto_add_volume(codec, label, " Capture", cidx,
- nid, HDA_INPUT);
- if (err < 0)
- return err;
+
+ if (pin_amp) {
+ err = cx_auto_add_volume(codec, label, " Boost", cidx,
+ nid, HDA_INPUT);
+ if (err < 0)
+ return err;
+ }
+
+ if (!multi_adc_volume)
+ continue;
+ for (j = 0; j < conn_len; j++) {
+ if (conn[j] == nid) {
+ err = cx_auto_add_volume_idx(codec, label,
+ " Capture", cidx, adc_nid, HDA_INPUT, j);
+ if (err < 0)
+ return err;
+ break;
+ }
+ }
}
return 0;
}
@@ -3881,6 +3939,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_cxt5066 },
{ .id = 0x14f15069, .name = "CX20585",
.patch = patch_cxt5066 },
+ { .id = 0x14f1506e, .name = "CX20590",
+ .patch = patch_cxt5066 },
{ .id = 0x14f15097, .name = "CX20631",
.patch = patch_conexant_auto },
{ .id = 0x14f15098, .name = "CX20632",
@@ -3907,6 +3967,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066");
MODULE_ALIAS("snd-hda-codec-id:14f15067");
MODULE_ALIAS("snd-hda-codec-id:14f15068");
MODULE_ALIAS("snd-hda-codec-id:14f15069");
+MODULE_ALIAS("snd-hda-codec-id:14f1506e");
MODULE_ALIAS("snd-hda-codec-id:14f15097");
MODULE_ALIAS("snd-hda-codec-id:14f15098");
MODULE_ALIAS("snd-hda-codec-id:14f150a1");
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2d5b83fa8d24..ec0fa2dd0a27 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
hdmi_ai->ver = 0x01;
hdmi_ai->len = 0x0a;
hdmi_ai->CC02_CT47 = channels - 1;
+ hdmi_ai->CA = ca;
hdmi_checksum_audio_infoframe(hdmi_ai);
} else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */
struct dp_audio_infoframe *dp_ai;
@@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
dp_ai->len = 0x1b;
dp_ai->ver = 0x11 << 2;
dp_ai->CC02_CT47 = channels - 1;
+ dp_ai->CA = ca;
} else {
snd_printd("HDMI: unknown connection type at pin %d\n",
pin_nid);
@@ -1632,6 +1634,9 @@ static struct hda_codec_preset snd_hda_preset_hdmi[] = {
{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
+{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
+{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
+/* 17 is known to be absent */
{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
@@ -1674,6 +1679,8 @@ MODULE_ALIAS("snd-hda-codec-id:10de0011");
MODULE_ALIAS("snd-hda-codec-id:10de0012");
MODULE_ALIAS("snd-hda-codec-id:10de0013");
MODULE_ALIAS("snd-hda-codec-id:10de0014");
+MODULE_ALIAS("snd-hda-codec-id:10de0015");
+MODULE_ALIAS("snd-hda-codec-id:10de0016");
MODULE_ALIAS("snd-hda-codec-id:10de0018");
MODULE_ALIAS("snd-hda-codec-id:10de0019");
MODULE_ALIAS("snd-hda-codec-id:10de001a");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 269dbff70b92..4261bb8eec1d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1133,11 +1133,8 @@ static void alc_automute_speaker(struct hda_codec *codec, int pinctl)
nid = spec->autocfg.hp_pins[i];
if (!nid)
break;
- if (snd_hda_jack_detect(codec, nid)) {
- spec->jack_present = 1;
- break;
- }
- alc_report_jack(codec, spec->autocfg.hp_pins[i]);
+ alc_report_jack(codec, nid);
+ spec->jack_present |= snd_hda_jack_detect(codec, nid);
}
mute = spec->jack_present ? HDA_AMP_MUTE : 0;
@@ -1721,7 +1718,9 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
{
struct alc_spec *spec = codec->spec;
int id = spec->fixup_id;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
const char *modelname = spec->fixup_name;
+#endif
int depth = 0;
if (!spec->fixup_list)
@@ -2288,6 +2287,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -10357,7 +10379,7 @@ static struct alc_config_preset alc882_presets[] = {
.init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_4930G] = {
- .mixers = { alc888_base_mixer,
+ .mixers = { alc888_acer_aspire_4930g_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
alc888_acer_aspire_4930g_verbs },
@@ -10930,9 +10952,6 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
return 0;
}
-static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg);
-
/* almost identical with ALC880 parser... */
static int alc882_parse_auto_config(struct hda_codec *codec)
{
@@ -10950,10 +10969,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
if (err < 0)
return err;
- if (codec->vendor_id == 0x10ec0887)
- err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
- else
- err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
@@ -12635,6 +12651,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
ALC262_HP_BPC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
+ ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -14956,8 +14974,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
- SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
{}
@@ -14991,7 +15012,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC),
@@ -17134,7 +17155,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
#define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c)
/* add playback controls from the parsed DAC table */
-/* Based on ALC880 version. But ALC861VD and ALC887 have separate,
+/* Based on ALC880 version. But ALC861VD has separate,
* different NIDs for mute/unmute switch and volume control */
static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
@@ -18801,6 +18822,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
{}
};
@@ -19461,6 +19483,7 @@ enum {
ALC662_FIXUP_ASPIRE,
ALC662_FIXUP_IDEAPAD,
ALC272_FIXUP_MARIO,
+ ALC662_FIXUP_CZC_P10T,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -19481,14 +19504,23 @@ static const struct alc_fixup alc662_fixups[] = {
[ALC272_FIXUP_MARIO] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc272_fixup_mario,
- }
+ },
+ [ALC662_FIXUP_CZC_P10T] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0},
+ {}
+ }
+ },
};
static struct snd_pci_quirk alc662_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
+ SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
{}
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9ea48b425d0b..bd7b123f6440 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -586,7 +586,12 @@ static hda_nid_t stac92hd83xxx_pin_nids[10] = {
0x0f, 0x10, 0x11, 0x1f, 0x20,
};
-static hda_nid_t stac92hd88xxx_pin_nids[10] = {
+static hda_nid_t stac92hd87xxx_pin_nids[6] = {
+ 0x0a, 0x0b, 0x0c, 0x0d,
+ 0x0f, 0x11,
+};
+
+static hda_nid_t stac92hd88xxx_pin_nids[8] = {
0x0a, 0x0b, 0x0c, 0x0d,
0x0f, 0x11, 0x1f, 0x20,
};
@@ -5430,12 +5435,13 @@ again:
switch (codec->vendor_id) {
case 0x111d76d1:
case 0x111d76d9:
+ case 0x111d76e5:
spec->dmic_nids = stac92hd87b_dmic_nids;
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd87b_dmic_nids,
STAC92HD87B_NUM_DMICS);
- spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids);
- spec->pin_nids = stac92hd88xxx_pin_nids;
+ spec->num_pins = ARRAY_SIZE(stac92hd87xxx_pin_nids);
+ spec->pin_nids = stac92hd87xxx_pin_nids;
spec->mono_nid = 0;
spec->num_pwrs = 0;
break;
@@ -5443,6 +5449,7 @@ again:
case 0x111d7667:
case 0x111d7668:
case 0x111d7669:
+ case 0x111d76e3:
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd88xxx_dmic_nids,
STAC92HD88XXX_NUM_DMICS);
@@ -6387,6 +6394,8 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx},
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index a76c3260d941..63b0054200a8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
hda_nid_t nid = cfg->inputs[i].pin;
if (spec->smart51_enabled && is_smart51_pins(spec, nid))
ctl = PIN_OUT;
- else if (i == AUTO_PIN_MIC)
+ else if (cfg->inputs[i].type == AUTO_PIN_MIC)
ctl = PIN_VREF50;
else
ctl = PIN_IN;
diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
index 7b62de089fee..20c6b079d0df 100644
--- a/sound/pci/ice1712/delta.c
+++ b/sound/pci/ice1712/delta.c
@@ -580,6 +580,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
{
int err;
struct snd_akm4xxx *ak;
+ unsigned char tmp;
if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA1010 &&
ice->eeprom.gpiodir == 0x7b)
@@ -622,6 +623,12 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
break;
}
+ /* initialize the SPI clock to high */
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+ tmp |= ICE1712_DELTA_AP_CCLK;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(5);
+
/* initialize spdif */
switch (ice->eeprom.subvendor) {
case ICE1712_SUBDEVICE_AUDIOPHILE:
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index c2ae63d17cd2..f53897a708b4 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -92,6 +92,8 @@ struct oxygen_model {
void (*update_dac_volume)(struct oxygen *chip);
void (*update_dac_mute)(struct oxygen *chip);
void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed);
+ unsigned int (*adjust_dac_routing)(struct oxygen *chip,
+ unsigned int play_routing);
void (*gpio_changed)(struct oxygen *chip);
void (*uart_input)(struct oxygen *chip);
void (*ac97_switch)(struct oxygen *chip,
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 9bff14d5895d..26c7e8bcb229 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip)
(1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
(2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
(3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+ if (chip->model.adjust_dac_routing)
+ reg_value = chip->model.adjust_dac_routing(chip, reg_value);
oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value,
OXYGEN_PLAY_DAC0_SOURCE_MASK |
OXYGEN_PLAY_DAC1_SOURCE_MASK |
diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c
index 9f72d424969c..252719101c42 100644
--- a/sound/pci/oxygen/xonar_cs43xx.c
+++ b/sound/pci/oxygen/xonar_cs43xx.c
@@ -392,7 +392,7 @@ static void dump_d1_registers(struct oxygen *chip,
unsigned int i;
snd_iprintf(buffer, "\nCS4398: 7?");
- for (i = 2; i <= 8; ++i)
+ for (i = 2; i < 8; ++i)
snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
snd_iprintf(buffer, "\n");
dump_cs4362a_registers(data, buffer);
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index e1fa602eba79..bc6eb58be380 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -24,6 +24,11 @@
*
* SPI 0 -> CS4245
*
+ * I²S 1 -> CS4245
+ * I²S 2 -> CS4361 (center/LFE)
+ * I²S 3 -> CS4361 (surround)
+ * I²S 4 -> CS4361 (front)
+ *
* GPIO 3 <- ?
* GPIO 4 <- headphone detect
* GPIO 5 -> route input jack to line-in (0) or mic-in (1)
@@ -36,6 +41,7 @@
* input 1 <- aux
* input 2 <- front mic
* input 4 <- line/mic
+ * DAC out -> headphones
* aux out -> front panel headphones
*/
@@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip,
cs4245_write_cached(chip, CS4245_ADC_CTRL, value);
}
+static inline unsigned int shift_bits(unsigned int value,
+ unsigned int shift_from,
+ unsigned int shift_to,
+ unsigned int mask)
+{
+ if (shift_from < shift_to)
+ return (value << (shift_to - shift_from)) & mask;
+ else
+ return (value >> (shift_from - shift_to)) & mask;
+}
+
+static unsigned int adjust_dg_dac_routing(struct oxygen *chip,
+ unsigned int play_routing)
+{
+ return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC1_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC2_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC0_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC3_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC3_SOURCE_MASK);
+}
+
static int output_switch_info(struct snd_kcontrol *ctl,
struct snd_ctl_elem_info *info)
{
@@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = {
.resume = dg_resume,
.set_dac_params = set_cs4245_dac_params,
.set_adc_params = set_cs4245_adc_params,
+ .adjust_dac_routing = adjust_dg_dac_routing,
.dump_registers = dump_cs4245_registers,
.model_data_size = sizeof(struct dg),
.device_config = PLAYBACK_0_TO_I2S |
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
index bd26e092aead..6ce9ad700290 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.h
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -22,7 +22,7 @@
#define __PDAUDIOCF_H
#include <sound/pcm.h>
-#include <asm/io.h>
+#include <linux/io.h>
#include <linux/interrupt.h>
#include <pcmcia/cistpl.h>
#include <pcmcia/ds.h>
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 989e04abb520..fe33e122e372 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -23,8 +23,8 @@
#include <linux/delay.h>
#include <linux/device.h>
#include <linux/firmware.h>
+#include <linux/io.h>
#include <sound/core.h>
-#include <asm/io.h>
#include "vxpocket.h"
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 68b97477577b..66eabafb1c24 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
}
dev->pcm->private_data = dev;
- strcpy(dev->pcm->name, dev->product_name);
+ strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name));
memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c
index 2f218c77fff2..a1a47088fd0c 100644
--- a/sound/usb/caiaq/midi.c
+++ b/sound/usb/caiaq/midi.c
@@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device)
if (ret < 0)
return ret;
- strcpy(rmidi->name, device->product_name);
+ strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name));
rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX;
rmidi->private_data = device;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 800f7cb4f251..c0f8270bc199 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -323,6 +323,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
return -ENOMEM;
}
+ mutex_init(&chip->shutdown_mutex);
chip->index = idx;
chip->dev = dev;
chip->card = card;
@@ -531,6 +532,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
chip = ptr;
card = chip->card;
mutex_lock(&register_mutex);
+ mutex_lock(&chip->shutdown_mutex);
chip->shutdown = 1;
chip->num_interfaces--;
if (chip->num_interfaces <= 0) {
@@ -548,9 +550,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
snd_usb_mixer_disconnect(p);
}
usb_chip[chip->index] = NULL;
+ mutex_unlock(&chip->shutdown_mutex);
mutex_unlock(&register_mutex);
snd_card_free_when_closed(card);
} else {
+ mutex_unlock(&chip->shutdown_mutex);
mutex_unlock(&register_mutex);
}
}
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 7df89b3d7ded..85af6051b52d 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -95,7 +95,7 @@ enum {
};
-/*E-mu 0202(0404) eXtension Unit(XU) control*/
+/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/
enum {
USB_XU_CLOCK_RATE = 0xe301,
USB_XU_CLOCK_SOURCE = 0xe302,
@@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
cval->initialized = 1;
} else {
if (type == USB_XU_CLOCK_RATE) {
- /* E-Mu USB 0404/0202/TrackerPre
+ /* E-Mu USB 0404/0202/TrackerPre/0204
* samplerate control quirk
*/
cval->min = 0;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 4132522ac90f..e3f680526cb5 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -361,6 +361,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
}
if (changed) {
+ mutex_lock(&subs->stream->chip->shutdown_mutex);
/* format changed */
snd_usb_release_substream_urbs(subs, 0);
/* influenced: period_bytes, channels, rate, format, */
@@ -368,6 +369,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
params_rate(hw_params),
snd_pcm_format_physical_width(params_format(hw_params)) *
params_channels(hw_params));
+ mutex_unlock(&subs->stream->chip->shutdown_mutex);
}
return ret;
@@ -385,8 +387,9 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream)
subs->cur_audiofmt = NULL;
subs->cur_rate = 0;
subs->period_bytes = 0;
- if (!subs->stream->chip->shutdown)
- snd_usb_release_substream_urbs(subs, 0);
+ mutex_lock(&subs->stream->chip->shutdown_mutex);
+ snd_usb_release_substream_urbs(subs, 0);
+ mutex_unlock(&subs->stream->chip->shutdown_mutex);
return snd_pcm_lib_free_vmalloc_buffer(substream);
}
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 35999874d301..921a86fd9884 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -79,6 +79,13 @@
.idProduct = 0x3f0a,
.bInterfaceClass = USB_CLASS_AUDIO,
},
+{
+ /* E-Mu 0204 USB */
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x041e,
+ .idProduct = 0x3f19,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
/*
* Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index cf8bf088394b..e314cdb85003 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat
}
/*
- * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device,
+ * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device,
* not for interface.
*/
@@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */
+ case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
set_format_emu_quirk(subs, fmt);
break;
}
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index db3eb21627ee..6e66fffe87f5 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -36,6 +36,7 @@ struct snd_usb_audio {
struct snd_card *card;
u32 usb_id;
int shutdown;
+ struct mutex shutdown_mutex;
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
int num_interfaces;
int num_suspended_intf;