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

[alsa-devel] [PATCH] ASoC: Intel: Clean data after SST fw fetch
by libin.yang＠intel.com 05 Jan '15

05 Jan '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 | 29 ++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 9 ++++++ sound/soc/intel/sst-haswell-pcm.c | 64 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 102 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c482..57d6e89 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" @@ -240,6 +241,10 @@ 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; }; @@ -1476,6 +1481,30 @@ u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, return stream->reply.volume_register_address[channel]; } +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; +} + int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 138e894..5c72e49 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 @@ -452,6 +453,14 @@ u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 channel); u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 channel); +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 6195252..4b70bdf 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, @@ -553,19 +558,29 @@ 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 = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw_stream *sst_stream = pcm_data->stream; struct sst_hsw *hsw = pdata->hsw; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; 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; } @@ -579,13 +594,62 @@ 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] ALSA: hda - Fix wrong gpio_dir & gpio_mask hint setups for IDT/STAC codecs
by Takashi Iwai 05 Jan '15

05 Jan '15

stac_store_hints() does utterly wrong for masking the values for gpio_dir and gpio_data, likely due to copy&paste errors. Fortunately, this feature is used very rarely, so the impact must be really small. Reported-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4f6413e01c13..605d14003d25 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -568,9 +568,9 @@ static void stac_store_hints(struct hda_codec *codec) spec->gpio_mask; } if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) - spec->gpio_mask &= spec->gpio_mask; - if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_data &= spec->gpio_mask; if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) spec->eapd_mask &= spec->gpio_mask; if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) -- 2.2.1

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[alsa-devel] Odd no-op in sound/pci/hda/patch_sigmatel.c
by Rasmus Villemoes 05 Jan '15

05 Jan '15

Hi Takashi The "spec->gpio_mask &= spec->gpio_mask" on line 571 of sound/pci/hda/patch_sigmatel.c is obviously a no-op. Is that on purpose, or was it supposed to be something else? Rasmus

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[alsa-devel] [patch] ALSA: asihpi: fix an information leak in asihpi_hpi_ioctl()
by Dan Carpenter 05 Jan '15

05 Jan '15

So far as I can see "hr->h.size" is set to "res_max_size" which is a user controlled value between 12 and USHRT_MAX. If it's larger than sizeof(*hr), then that leads to an information leak. I am not very familiar with this code, my other question here is that on lines before we set "hr->h.size = sizeof(hr->h)". It think this is a bug. I also think this particular code is never executed and I added a comment to that effect. But we do it in earlier in the function as well: copy_to_user(puhr, hr, sizeof(hr->h)); It doesn't make sense to me. Anyway, I think my patch is safe and it seems to fix a real information leak. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 6aa677e..f88109a 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -283,6 +283,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } + /* FIXME: isn't this a no-op? */ if (hr->h.size > res_max_size) { HPI_DEBUG_LOG(ERROR, "response too big %d %d\n", hr->h.size, res_max_size); @@ -290,6 +291,9 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) hr->h.specific_error = hr->h.size; hr->h.size = sizeof(hr->h); } + /* prevent an information leak */ + if (hr->h.size > sizeof(*hr)) + hr->h.size = sizeof(*hr); uncopied_bytes = copy_to_user(puhr, hr, hr->h.size); if (uncopied_bytes) {

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[alsa-devel] [PATCH] ALSA: snd-usb-caiaq: fix stream count check
by Daniel Mack 05 Jan '15

05 Jan '15

Commit 897c329bc ("ALSA: usb: caiaq: check for cdev->n_streams > 1") introduced a safety check to protect against bogus data provided by devices. However, the n_streams variable is already divided by CHANNELS_PER_STREAM, so the correct check is 'n_streams > 0'. Fix this to un-break support for stereo devices. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Cc: stable(a)kernel.org [v3.18+] --- sound/usb/caiaq/audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 2728447..327f864 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -816,7 +816,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) return -EINVAL; } - if (cdev->n_streams < 2) { + if (cdev->n_streams < 1) { dev_err(dev, "bogus number of streams: %d\n", cdev->n_streams); return -EINVAL; } -- 2.2.0

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[alsa-devel] DSD over HDMI: audio data packetization
by Xuebing Wang 05 Jan '15

05 Jan '15

Hi Community, I am trying to implement DSD (one bit audio) over HDMI, for Intel integrated graphics. I am referencing HDMI specification 1.3a. Regarding audio data packetization, I am referencing spec: 5.3.4 Audio Sample Packet 7.6 Audio Data Packetization Q1) Would you please point to me where is the source code to handle L-PCM audio sample packetization (to implement 5.3.4 Audio Sample Packet)? Q2) I admit that I did not carefully read HD-Audio specification. Is this possible that this L-PCM audio data packetization is done by *hardware* (e.g HDMI related widgets/pins)? Q3) Did we implement "7.2 Audio Sample Clock Capture and Regeneration"? Thanks. Xuebing Wang

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[alsa-devel] [PATCH - ALSA: snd-virtuoso 1/1] Xonar DSX support
by sgiurgiu11＠gmail.com 04 Jan '15

04 Jan '15

From: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Thank you, Sergiu Giurgiu. Signed-off-by: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> diff --git a/mirror/sound/pci/oxygen/virtuoso.c b/mirror/sound/pci/oxygen/virtuoso.c index 3d71423..8104eab 100644 --- a/mirror/sound/pci/oxygen/virtuoso.c +++ b/mirror/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/mirror/sound/pci/oxygen/xonar_wm87x6.c b/mirror/sound/pci/oxygen/xonar_wm87x6.c index 63cff90..b555b62 100644 --- a/mirror/sound/pci/oxygen/xonar_wm87x6.c +++ b/mirror/sound/pci/oxygen/xonar_wm87x6.c @@ -1255,7 +1255,6 @@ static void dump_wm87x6_registers(struct oxygen *chip, } static const struct oxygen_model model_xonar_ds = { - .shortname = "Xonar DS", .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, @@ -1327,6 +1326,11 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, switch (id->subdevice) { case 0x838e: chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DS"; + break; + case 0x8522: + chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DSX"; break; case 0x835e: chip->model = model_xonar_hdav_slim; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: Intel: Don't change offset of block allocator during fixed allocate
by Jie Yang 04 Jan '15

04 Jan '15

The offset of block allocator, ba->offset, should not be changed during fixed address allocating, for the caller may treat it as the offset of allocated memory and use it. In the case that we allocate more than 1 blocks, we should make sure this offset is correct. Here introduces a temp allocator for the later continuous allocating. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 50d6925..a2ae2c5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -707,6 +707,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba struct list_head *block_list) { struct sst_mem_block *block, *tmp; + struct sst_block_allocator ba_tmp = *ba; u32 end = ba->offset + ba->size, block_end; int err; @@ -731,9 +732,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba if (ba->offset >= block->offset && ba->offset < block_end) { /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; @@ -768,10 +769,10 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba list_move(&block->list, &dsp->used_block_list); list_add(&block->module_list, block_list); /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; -- 1.9.1

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[alsa-devel] [PATCH] ALSA: pcm: Remove unused functions declaration
by Lars-Peter Clausen 03 Jan '15

03 Jan '15

Remove function declarations for functions that don't have a matching implementation. For snd_pcm_build_linear_format the implementation was removed in 64d27f96cb719cf8 ("[ALSA] Support 3-bytes 24bit format in PCM OSS emulation"). All the others never had one (as far as git history goes). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm.h | 7 ------- 1 file changed, 7 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index bf32cea..cd09c1b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -518,7 +518,6 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream); int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, struct file *file, struct snd_pcm_substream **rsubstream); void snd_pcm_detach_substream(struct snd_pcm_substream *substream); -void snd_pcm_vma_notify_data(void *client, void *data); int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area); @@ -984,21 +983,15 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples); const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int frames); -snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsigned, int big_endian); void snd_pcm_set_ops(struct snd_pcm * pcm, int direction, const struct snd_pcm_ops *ops); void snd_pcm_set_sync(struct snd_pcm_substream *substream); -int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg); int snd_pcm_update_state(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); -int snd_pcm_playback_xrun_check(struct snd_pcm_substream *substream); -int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream); -int snd_pcm_playback_xrun_asap(struct snd_pcm_substream *substream); -int snd_pcm_capture_xrun_asap(struct snd_pcm_substream *substream); void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr); void snd_pcm_period_elapsed(struct snd_pcm_substream *substream); snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream, -- 1.8.0

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[alsa-devel] [PATCH] sound: soc: soc-pcm: Remove useless varialbe 'i'
by Chen Gang 03 Jan '15

03 Jan '15

'i' is useless in soc_pcm_set_msb(), so remove it, the related warning (with allmodconfig under s390): CC [M] sound/soc/soc-pcm.o sound/soc/soc-pcm.c: In function 'soc_pcm_set_msb': sound/soc/soc-pcm.c:307:11: warning: unused variable 'i' [-Wunused-variable] int ret, i; ^ Signed-off-by: Chen Gang <gang.chen.5i5j(a)gmail.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d6e8d6b..6b0136e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -304,7 +304,7 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - int ret, i; + int ret; if (!bits) return; -- 1.7.9.5

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