February 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] ucm: miscellaneous fixes
by Tanu Kaskinen 11 Feb '15

11 Feb '15

While working on PulseAudio's UCM code, I noticed strange use of const in snd_use_case_get(). The third patch fixes that. While working on the third patch, I also noticed strange code in execute_sequence(). The first patch fixes one obvious bug, but there was another issue which I didn't know how to fix, so I just added a FIXME comment instead (that's just as an RFC, I hope the discussion will reveal what the proper fix should be). Tanu Kaskinen (3): ucm: fix variable mixup ucm: add a FIXME comment ucm: fix inappropriate use of const include/use-case.h | 2 +- src/ucm/main.c | 34 +++++++++++++++++++++------------- 2 files changed, 22 insertions(+), 14 deletions(-) -- 1.9.3

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[alsa-devel] [patch] ALSA: seq: potential out of bounds in do_control()
by Dan Carpenter 11 Feb '15

11 Feb '15

Smatch complains that "control" is user specifigy and needs to be capped. The call tree to understand this warning is quite long. snd_seq_write() <-- get the event from the user snd_seq_client_enqueue_event() snd_seq_deliver_event() deliver_to_subscribers() snd_seq_deliver_single_event() snd_opl3_oss_event_input() snd_midi_process_event() do_control() Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- I have spent some time reviewing this code, but I may have missed something where we verify that control is in bounds. I'm not very familiar with this code and the call tree is fairly long. diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 9b6470c..7ba9373 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -269,6 +269,9 @@ do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chse { int i; + if (control >= ARRAY_SIZE(chan->control)) + return; + /* Switches */ if ((control >=64 && control <=69) || (control >= 80 && control <= 83)) { /* These are all switches; either off or on so set to 0 or 127 */

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[alsa-devel] [PATCH v2] ALSA: line6: Improve line6_read/write_data() interfaces
by Chris Rorvick 11 Feb '15

11 Feb '15

The address cannot be negative so make it unsigned. Also, an unsigned int is always sufficient for the length, so no need to overdo it with a size_t. Finally, add in range checks to see if the values passed in actually fit where they are used. Signed-off-by: Chris Rorvick <chris(a)rorvick.com> --- This was dropped from an earlier series ("Cleanup reads/writes to Line 6 device memory".) This version leaves the arguements wide (within reason) and does range checking. Regards, Chris sound/usb/line6/driver.c | 14 ++++++++++---- sound/usb/line6/driver.h | 8 ++++---- 2 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 99b63a7..4f82b57 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -302,14 +302,17 @@ static void line6_data_received(struct urb *urb) /* Read data from device. */ -int line6_read_data(struct usb_line6 *line6, int address, void *data, - size_t datalen) +int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, + unsigned datalen) { struct usb_device *usbdev = line6->usbdev; int ret; unsigned char len; unsigned count; + if (address > 0xffff || datalen > 0xff) + return EINVAL; + /* query the serial number: */ ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, @@ -370,14 +373,17 @@ EXPORT_SYMBOL_GPL(line6_read_data); /* Write data to device. */ -int line6_write_data(struct usb_line6 *line6, int address, void *data, - size_t datalen) +int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, + unsigned datalen) { struct usb_device *usbdev = line6->usbdev; int ret; unsigned char status; int count; + if (address > 0xffff || datalen > 0xffff) + return EINVAL; + ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x0022, address, data, datalen, diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 5d20294..7da643e 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -147,8 +147,8 @@ struct usb_line6 { extern char *line6_alloc_sysex_buffer(struct usb_line6 *line6, int code1, int code2, int size); -extern int line6_read_data(struct usb_line6 *line6, int address, void *data, - size_t datalen); +extern int line6_read_data(struct usb_line6 *line6, unsigned address, + void *data, unsigned datalen); extern int line6_read_serial_number(struct usb_line6 *line6, u32 *serial_number); extern int line6_send_raw_message_async(struct usb_line6 *line6, @@ -161,8 +161,8 @@ extern void line6_start_timer(struct timer_list *timer, unsigned long msecs, void (*function)(unsigned long), unsigned long data); extern int line6_version_request_async(struct usb_line6 *line6); -extern int line6_write_data(struct usb_line6 *line6, int address, void *data, - size_t datalen); +extern int line6_write_data(struct usb_line6 *line6, unsigned address, + void *data, unsigned datalen); int line6_probe(struct usb_interface *interface, const struct usb_device_id *id, -- 2.1.0

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[alsa-devel] Memory and Cycles overhead in <snd_soc_init>
by Bhaskara rao Budiredla 11 Feb '15

11 Feb '15

Hi - May I know the purpose of the call to snd_soc_init( ) from kernel_init thread? snd_soc_init( ) is trying to register a dummy platform device "snd-soc-dummy" containing (dummy_codec/dummy_platform) and a sound card device. Neither of those devices are added to platform bus before they are making a call to platform_driver_register( ). Due to that platform_match( ) will fail and they will not be added to platform bus . Thus, snd_soc_init( ) is not doing any useful task during kernel boot time. Could someone help me if my understanding is correct? If I am correct, shall we remove the call to snd_soc_init( ) in kernel_init thread to save boot time and memory? Otherwise please suggest me where those dummy devices and sound card gets added to platform bus and the purpose of creating them? Thanks, Bhaskara

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[alsa-devel] [PATCH 0/6] Cleanup reads/writes to Line 6 device memory
by Chris Rorvick 11 Feb '15

11 Feb '15

The primary fix here is to throttle the requests for status after a read or write operation. In some cases I was seeing more than 1000 before the device was ready with a response. The remainder are smaller improvements related to the read/write logic. All of this seems pretty straightforward with the possible exception of changing the returned error from EINVAL to EIO. The EINVAL seems wrong to me but maybe not? Regards, Chris Chris Rorvick (6): ALSA: line6: Improve line6_read/write_data() interfaces ALSA: line6: Add delay before reading status ALSA: line6: Return error if device not responding ALSA: line6: Return EIO if read/write not successful ALSA: line6: Use explicit type for serial number ALSA: line6: toneport: Use explicit type for firmware version sound/usb/line6/driver.c | 51 +++++++++++++++++++++++++++++++++------------- sound/usb/line6/driver.h | 10 ++++----- sound/usb/line6/pod.c | 4 ++-- sound/usb/line6/toneport.c | 4 ++-- 4 files changed, 46 insertions(+), 23 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH v3 kernel 1/3] snd_pcm_start_at and friends.
by Tim Cussins 11 Feb '15

11 Feb '15

We introduce the kernel-side of the START_AT ioctl. struct runtime is updated to hold information about the currently active start_at timer, if any. This facilitates cancellation via snd_pcm_start_at_abort(), and querying via snd_pcm_status(). struct snd_start_at holds a startat operation and its arguments. Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 0e88e7a..2943e1a 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -421,7 +421,10 @@ struct snd_pcm_status { snd_pcm_state_t suspended_state; /* suspended stream state */ __u32 reserved_alignment; /* must be filled with zero */ struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + int startat_pending; /* 1 if a start_at timer is pending, 0 otherwise */ + int startat_clock_type; /* start_at clock type, if pending */ + struct timespec startat_start_time; /* start_at start time, if pending */ + unsigned char reserved[48-(2*sizeof(struct timespec))]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -473,6 +476,34 @@ enum { SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; +enum { + SNDRV_PCM_STARTAT_OP_SET = 0, + SNDRV_PCM_STARTAT_OP_CANCEL, + SNDRV_PCM_STARTAT_OP_STATUS, + SNDRV_PCM_STARTAT_OP_LAST = SNDRV_PCM_STARTAT_OP_STATUS, +}; + +enum { + SNDRV_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY = 0, + SNDRV_PCM_STARTAT_CLOCK_TYPE_MONOTONIC, + SNDRV_PCM_STARTAT_CLOCK_TYPE_LINK, + SNDRV_PCM_STARTAT_CLOCK_TYPE_LAST = SNDRV_PCM_STARTAT_CLOCK_TYPE_LINK, +}; + +struct snd_start_at { + int op; /* startat operation to be performed */ + union { /* fields for setting a startat timer */ + struct { + int clock_type; /* clock type e.g. SNDRV_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY */ + struct timespec start_time; /* start time */ + } set; + struct { + int clock_type; + struct timespec current_time; + } status; + } args; +}; + /* channel positions */ enum { SNDRV_CHMAP_UNKNOWN = 0, @@ -551,6 +582,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_start_at) + /***************************************************************************** * * -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Clean data after SST fw fetch
by libin.yang＠intel.com 11 Feb '15

11 Feb '15

From: Libin Yang <libin.yang(a)intel.com> The BDW audio firmware DSP manages the DMA and the DMA cannot be stopped exactly at the end of the playback stream. This means stale samples may be played at PCM stop unless the driver copies silence to the subsequent periods. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 28 +++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 9 ++++++ sound/soc/intel/sst-haswell-pcm.c | 65 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 102 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 0ab1309..394af56 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -31,6 +31,7 @@ #include <linux/dma-mapping.h> #include <linux/debugfs.h> #include <linux/pm_runtime.h> +#include <sound/asound.h> #include "sst-haswell-ipc.h" #include "sst-dsp.h" @@ -242,6 +243,9 @@ struct sst_hsw_stream { u32 (*notify_position)(struct sst_hsw_stream *stream, void *data); void *pdata; + /* record the fw read position when playback */ + snd_pcm_uframes_t old_position; + bool play_silence; struct list_head node; }; @@ -1347,6 +1351,30 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return 0; } +snd_pcm_uframes_t sst_hsw_stream_get_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->old_position; +} + +void sst_hsw_stream_set_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, snd_pcm_uframes_t val) +{ + stream->old_position = val; +} + +bool sst_hsw_stream_get_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->play_silence; +} + +void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, bool val) +{ + stream->play_silence = val; +} + /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ int sst_hsw_mixer_get_info(struct sst_hsw *hsw) diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index c1ad901..8580960 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -20,6 +20,7 @@ #include <linux/types.h> #include <linux/kernel.h> #include <linux/platform_device.h> +#include <sound/asound.h> #define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 @@ -425,6 +426,14 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); +snd_pcm_uframes_t sst_hsw_stream_get_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +void sst_hsw_stream_set_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, snd_pcm_uframes_t val); +bool sst_hsw_stream_get_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, bool val); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 78fa01b..d6fa9d5 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -36,6 +36,11 @@ #define HSW_PCM_COUNT 6 #define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */ +#define SST_OLD_POSITION(d, r, o) ((d) + \ + frames_to_bytes(r, o)) +#define SST_SAMPLES(r, x) (bytes_to_samples(r, \ + frames_to_bytes(r, (x)))) + /* simple volume table */ static const u32 volume_map[] = { HSW_VOLUME_MAX >> 30, @@ -577,23 +582,34 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); struct hsw_pcm_data *pcm_data; + struct sst_hsw_stream *sst_stream; struct sst_hsw *hsw = pdata->hsw; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; int dai; dai = mod_map[rtd->cpu_dai->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; + sst_stream = pcm_data->stream; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_hsw_stream_set_silence_start(hsw, sst_stream, false); sst_hsw_stream_resume(hsw, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_hsw_stream_set_silence_start(hsw, sst_stream, false); sst_hsw_stream_pause(hsw, pcm_data->stream, 0); break; + case SNDRV_PCM_TRIGGER_DRAIN: + pos = runtime->control->appl_ptr % runtime->buffer_size; + sst_hsw_stream_set_old_position(hsw, pcm_data->stream, pos); + sst_hsw_stream_set_silence_start(hsw, sst_stream, true); + break; default: break; } @@ -607,13 +623,62 @@ static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) struct snd_pcm_substream *substream = pcm_data->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_hsw *hsw = pdata->hsw; u32 pos; + snd_pcm_uframes_t position = bytes_to_frames(runtime, + sst_hsw_get_dsp_position(hsw, pcm_data->stream)); + unsigned char *dma_area = runtime->dma_area; + snd_pcm_uframes_t dma_frames = + bytes_to_frames(runtime, runtime->dma_bytes); + snd_pcm_uframes_t old_position; + ssize_t samples; pos = frames_to_bytes(runtime, (runtime->control->appl_ptr % runtime->buffer_size)); dev_vdbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + /* SST fw don't know where to stop dma + * So, SST driver need to clean the data which has been consumed + */ + if (dma_area == NULL || dma_frames <= 0 + || (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + || !sst_hsw_stream_get_silence_start(hsw, stream)) { + snd_pcm_period_elapsed(substream); + return pos; + } + + old_position = sst_hsw_stream_get_old_position(hsw, stream); + if (position > old_position) { + if (position < dma_frames) { + samples = SST_SAMPLES(runtime, position - old_position); + snd_pcm_format_set_silence(runtime->format, + SST_OLD_POSITION(dma_area, + runtime, old_position), + samples); + } else + dev_err(rtd->dev, "PCM: position is wrong\n"); + } else { + if (old_position < dma_frames) { + samples = SST_SAMPLES(runtime, + dma_frames - old_position); + snd_pcm_format_set_silence(runtime->format, + SST_OLD_POSITION(dma_area, + runtime, old_position), + samples); + } else + dev_err(rtd->dev, "PCM: dma_bytes is wrong\n"); + if (position < dma_frames) { + samples = SST_SAMPLES(runtime, position); + snd_pcm_format_set_silence(runtime->format, + dma_area, samples); + } else + dev_err(rtd->dev, "PCM: position is wrong\n"); + } + sst_hsw_stream_set_old_position(hsw, stream, position); + /* let alsa know we have play a period */ snd_pcm_period_elapsed(substream); return pos; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix the incorrect limitation of the bit clock rate
by Nicolin Chen 11 Feb '15

11 Feb '15

According to i.MX Reference Manual, the bit-clock frequency generated by SSI must be never greater than 1/5 of the peripheral clock frequency. This peripheral clock, however, is not baudclk but the IPG clock (i.e. ssi_private->clk in the fsl_ssi driver). So this patch just simply fixes the incorrect limitation applied to the bit clock (baudclk) rate. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 059496e..d7365c5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - /* The bclk rate must be smaller than 1/5 sysclk rate */ - if (factor * (i + 1) < 5) - continue; - tmprate = freq * factor * (i + 2); if (baudclk_is_used) @@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (clkrate * 5 > clk_get_rate(ssi_private->clk)) + continue; + clkrate /= factor; afreq = clkrate / (i + 1); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5677: Add the chip type to distinguish the setting of the clock source
by Oder Chiou 11 Feb '15

11 Feb '15

There is only one clock source in the rt5676, so the chip type is added to distinguish the setting of the clock source in the VAD function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 23 +++++++++++++++++------ sound/soc/codecs/rt5677.h | 6 ++++++ 2 files changed, 23 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb87..3aae8a4 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -718,11 +718,19 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); + if (rt5677->type == RT5677) { + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | + RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | + RT5677_DSP_CLK_SRC_BYPASS); + } else { + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_MASK, + RT5677_DSP_CLK_SRC_BYPASS); + } regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); rt5677_set_dsp_mode(codec, true); @@ -4733,7 +4741,8 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", 0 }, + { "rt5677", RT5677 }, + { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -4850,6 +4859,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); + rt5677->type = id->driver_data; + if (pdata) rt5677->pdata = *pdata; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index c0a625f..07df96b 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1665,6 +1665,11 @@ enum { RT5677_IRQ_JD3, }; +enum rt5677_type { + RT5677, + RT5676, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1681,6 +1686,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [GIT PULL]: Add sst audio firmware for Braswell platforms
by Vinod Koul 11 Feb '15

11 Feb '15

Hello Linux firmware maintainers, Please PULL to receive the Intel audio firmware for audio DSP in Braswell platforms. This firmware is using the same license as the Baytrail firmware already available in linux-firmware git. The following changes since commit b794c5039dcf0b7ebfeb58929d035f7a1d4c80dd: Merge branch 'marvell-mwifiex' (2015-01-27 10:42:47 -0500) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git BSW for you to fetch changes up to f692aba54c8c80cb27a3eb5388594a55a30504ea: linux-firmware: add sst audio firmware for braswell platforms (2015-02-02 21:12:22 -0800) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: add sst audio firmware for braswell platforms WHENCE | 8 ++++++++ intel/fw_sst_22a8.bin | Bin 0 -> 647390 bytes 2 files changed, 8 insertions(+) create mode 100644 intel/fw_sst_22a8.bin Thanks -- ~Vinod

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