December 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:07 PM, prasannatsmkumar <prasannatsmkumar(a)gmail.com> wrote: > Hi ALSA people, > > I am having a Realtek ALC 883 chip. The sound volume is very low > compared to what I hear in windows - I maintain same speaker volume > and play same audio. If I keep the volume as 10% in windows I have to > keep a volume of at least 25+% in Linux. Is there any way to improve > it? I have been seeing such a situation for past 4 or 5 years (from > Fedora 9 period till today). I have tried several distributions > (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the > same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. > > $cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.25. > > Should I file a bug? If so in which component (kernel or pulseaudio)? > > Thanks in advance, > PrasannaKumar Resending this as this mail was bounced by alsa devel list. Please feel free to ask for information necessary to solve this issue. I have copied output of some commands that may be useful for understanding or digging this issue. dmesg | grep hda output: [ 30.648663] snd_hda_intel 0000:00:1b.0: >irq 43 for MSI/MSI-X [ 30.716741] hda_codec: ALC883: SKU not ready 0x411111f0 lspci -vvv output (only info related audio device is in mail all others are stripped): 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) Subsystem: Intel Corporation Device d605 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 43 Region 0: Memory at 501c0000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4189 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=80 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c000 Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel amixer output: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 13 [42%] [-27.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 3 [100%] [30.00dB] Front Right: 3 [100%] [30.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 8 [26%] [0.00dB] [on] Front Right: Capture 8 [26%] [0.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [off] Front Right: Capture 0 [0%] [-12.00dB] [off] Simple mixer control '2ch-Upmix Master',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 255 Front Left: 255 [100%] Front Right: 255 [100%] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Rear Mic' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Thanks and Regards, PrasannaKumar

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[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

Hi ALSA / pulseaudio people, Resending this as this mail was bounced by alsa devel list. I am having a Realtek ALC 883 chip. The sound volume is very low compared to what I hear in windows - I maintain same speaker volume and play same audio. If I keep the volume as 10% in windows I have to keep a volume of at least 25+% in Linux. Is there any way to improve it? I have been seeing such a situation for past 4 or 5 years (from Fedora 9 period till today). I have tried several distributions (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. $cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.25. Should I file a bug? If so in which component (kernel or pulseaudio)? Thanks in advance, PrasannaKumar

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[alsa-devel] [PATCH RFT] ASoC: sta529: Fix update register bits in sta529_set_dai_fmt
by Axel Lin 20 Dec '12

20 Dec '12

Both the mask and mode settings are wrong in current code. According to the datasheet: S2PCFG0 (0x0A) BIT[3:1] DATA_FORMAT serial interface protocol format: 000: left Justified 001: I2S (default) 010: right justified 100: PCM no delay 101: PCM delay 111: DSP Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and RIGHT_J_DATA_FORMAT. Also adds define for DATA_FORMAT_MSK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Rajeev, I don't have this hardware, can you test this patch? Thanks, Axel sound/soc/codecs/sta529.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index ab355c4..40c07be 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -74,9 +74,10 @@ SNDRV_PCM_FMTBIT_S32_LE) #define S2PC_VALUE 0x98 #define CLOCK_OUT 0x60 -#define LEFT_J_DATA_FORMAT 0x10 -#define I2S_DATA_FORMAT 0x12 -#define RIGHT_J_DATA_FORMAT 0x14 +#define DATA_FORMAT_MSK 0x0E +#define LEFT_J_DATA_FORMAT 0x00 +#define I2S_DATA_FORMAT 0x02 +#define RIGHT_J_DATA_FORMAT 0x04 #define CODEC_MUTE_VAL 0x80 #define POWER_CNTLMSAK 0x40 @@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return -EINVAL; } - snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 20 Dec '12

20 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,6 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- 1.7.8.3 Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] [PATCH RFT] ASoC: arizona: Do proper shift for setting aif1 rate
by Axel Lin 20 Dec '12

20 Dec '12

ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */ Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/arizona.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b..38248a7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -714,7 +714,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 06:37:26AM +0000, Liu, Chuansheng wrote: > Meanwhile, is it useful to add one warning there for that case? > After all, in probing, set the bias to _STANDBY even idle_bias_off == 1, and calling get_runtime_sync(), it > will let the code more obscure. So giving a warning there to indicate the driver: > it is not suggested that in probing, set the bias to _STANDBY even idle_bias_off == 1. Probably, send a patch please. Like I say it is possible to start off in _STANDBY providing the driver grabs the runtime PM reference too but I can't think of any reason for doing that so the warning seems sensible.

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Re: [alsa-devel] [PATCH 2/7] ASoC: allow wolfson wm9712 codec to be instantiated using device tree
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 12:17:31AM +0100, Lucas Stach wrote: > static struct platform_driver wm9712_codec_driver = { > .driver = { > .name = "wm9712-codec", > .owner = THIS_MODULE, > + .of_match_table = wm9712_of_match, > }, No, this is really not a sane way of handling AC'97 at all. They're not platform devices, there is no sane reason why they should appear in the device tree as such. Really we ought to be just hotplugging them, AC'97 is perfectly enumerable.

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[alsa-devel] Adding wm8962 support to mx6qsabresd
by Fabio Estevam 20 Dec '12

20 Dec '12

Hi, I am working on adding audio support to mx6qsabresd board, which contains a wm8962 codec. Here is the log with debug information turned on that I am getting so far: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 200 => ffffffff wm8962 0-001a: read 201 => 0 wm8962 0-001a: read 202 => 0 wm8962 0-001a: read 203 => ffffffff wm8962 0-001a: read 204 => 8100 wm8962 0-001a: read 205 => 8100 wm8962 0-001a: DMIC not in use, disabling input: WM8962 Beep Generator as /devices/soc.0/2100000.aips-bus/21a0000.i2c/i2c-0/0-001a/input/input0 wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 300 => 1c00 wm8962 0-001a: read 17 => 160 wm8962 0-001a: read 48 => 0 wm8962 0-001a: read 8 => ffffffff wm8962 0-001a: read 52 => 4 wm8962 0-001a: read 19 => 0 wm8962 0-001a: write a0 = 149 wm8962 0-001a: write a1 = 271 wm8962 0-001a: write a2 = 7 wm8962 0-001a: FLL configured for 24000000Hz->11289600Hz wm8962 0-001a: No audio clocks configured ... imx-wm8962 sound.14: wm8962 <-> 2028000.ssi mapping ok root@freescale /home$ ./aplay Front_Left.wav wm8962 0-001a: hw_params set BCLK 1411200Hz LRCLK 44100Hz wm8962 0-001a: wm8962->sysclk_rate = 11289600 wm8962 0-001a: wm8962->lrclk = 44100 wm8962 0-001a: Selected sysclk ratio 256 wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 3 => 100 wm8962 0-001a: write 3 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTR PGA event failed: -5 wm8962 0-001a: read 29 => 100 wm8962 0-001a: write 29 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTR PGA event failed: -5 wm8962 0-001a: read 2 => 100 wm8962 0-001a: write 2 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTL PGA event failed: -5 wm8962 0-001a: read 28 => 100 wm8962 0-001a: write 28 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTL PGA event failed: -5 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 wm8962 0-001a: DC servo complete after 1ms Playing WAVE 'Front_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Any idea of why register 4 and 42 read fails? wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 Any suggestion is appreciated. Thanks, Fabio Estevam

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[alsa-devel] how to check the CODEC registers vaule
by Zhang wei 20 Dec '12

20 Dec '12

Hello everyone,I am confused again : ( I read the wm8985.pdf and found the description: If the address is not recognised or the R/W bit is '1' when operating in write only mode,the WM8985 returns to the idle conditon and waits for a new start condition and valid address. Does it mean the wm8985 registers write only, can not be read? I found other kinds of codec datasheet same as it. Is it necessary to read back the value from the register ? Thanks a lot. Best Regards zw

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[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

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