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

[alsa-devel] [PATCH] ASoC: wm8962: Add device tree support
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add device tree support. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8962.txt | 16 ++++++++++++++++ sound/soc/codecs/wm8962.c | 7 +++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt new file mode 100644 index 0000000..dceb3b1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -0,0 +1,16 @@ +WM8962 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8962" + + - reg : the I2C address of the device. + +Example: + +codec: wm8962@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8fd38cb..1ebeff4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3758,10 +3758,17 @@ static const struct i2c_device_id wm8962_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id); +static const struct of_device_id wm8962_of_match[] = { + { .compatible = "wlf,wm8962", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8962_of_match); + static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", .owner = THIS_MODULE, + .of_match_table = wm8962_of_match, .pm = &wm8962_pm, }, .probe = wm8962_i2c_probe, -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off
by Mark Brown 24 Dec '12

24 Dec '12

On Fri, Dec 21, 2012 at 06:17:12PM +0800, Chuansheng Liu wrote: > > Just found some cases that some codec drivers set the bias to _STANDBY and > set idle_bias_off to 1 during probing. > It will cause unpaired runtime_get_sync/put() issue. Also as Mark suggested, > there is no reason to start from _STANDBY bias with idle_bias_off == 1. Applied, thanks.

1 0

[alsa-devel] [PATCH RFT] ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()
by Axel Lin 24 Dec '12

24 Dec '12

The mode variable is either 0 or 1. To update mode setting, the mask should be BIT(0) rather than BIT(1). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi M R Swami, I don't have the datasheet. Can you check if this patch is correct? Thanks, Axel sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index d75257d..c0d203b 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1218,7 +1218,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, - LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5), (aif_val | mode | clk_phase)); snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); -- 1.7.9.5

3 2

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Add sgtl5000 into the Subject sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

2 1

[alsa-devel] Does it will effect the precision of DMA position?
by Zhang wei 24 Dec '12

24 Dec '12

Hello guys. In the function snd_pcm_update_hw_ptr0(): pos -= pos % runtime->min_align; after running it, the 'pos' maybe smaller than before.because there is need to aligned.so that means lost or drop some Audio data for ever ,is that true? The data playbacked is a bit of different from we send to? Best Regards zw

1 0

[alsa-devel] [alsa-lib, dmix] Support for S24_3BE
by Vincent Vinel 24 Dec '12

24 Dec '12

Hello, The bugtracker seems to be down, so I'm posting here instead. I was trying to use software mixing with my Native Instruments USB soundcard (snd-usb-caiaq), but all attemps to use the dmix pcm ended up with the rather unhelpful error : ALSA lib pcm_dmix.c:1311:(_snd_pcm_dmix_open) Unsupported format When using the hw pcm, however, aplay was kind enough to display the list of available formats, which in my case was limited to S24_3BE. A quick check confirmed that dmix doesn't support that format, which happens to be the only format available for NI USB soundcards (as defined in snd-usb-caiaq). Is it worth writing a patch for 24bit big-endian support in dmix ? External soundcards like these ones are usually used for realtime effects processing and low-latency audio, things for which dmix proably isn't suited anyway. -- Vincent Vinel

2 1

Re: [alsa-devel] [PATCH V5 2/2] ASoC: SAMSUNG: Add DT support for i2s
by Grant Likely 24 Dec '12

24 Dec '12

On Thu, 13 Dec 2012 16:12:53 +0530, Padmavathi Venna <padma.v(a)samsung.com> wrote: > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- > .../devicetree/bindings/sound/samsung-i2s.txt | 75 +++++++ > sound/soc/samsung/dma.c | 1 + > sound/soc/samsung/dma.h | 1 + > sound/soc/samsung/i2s.c | 233 ++++++++++++++++---- > 4 files changed, 269 insertions(+), 41 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt > > diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > new file mode 100644 > index 0000000..bff47b4 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > @@ -0,0 +1,75 @@ > +* Samsung I2S controller > + > +Required SoC Specific Properties: > + > +- compatible : "samsung,samsung-i2s" Isn't that kind of redundant? :-) The format of the compatible strings should be "<vendor>,<part-number>-i2s". Please be specific about the part number that you're doing the binding for. For example; use "samsung,exynos4210-i2s" instead of "samsung,exynos-i2s". > +- reg: physical base address of the controller and length of memory mapped > + region. > + > +[PRELIMINARY: the dma channel allocation will change once there are > +official DMA bindings] > + > +- tx-dma-channel-secondary: The dma channel specifier for secondary tx > + operations. The format of the dma specifier depends on the dma > + controller. > +- tx-dma-channel: The dma channel specifier for tx operations. The format of > + the dma specifier depends on the dma controller. > +- rx-dma-channel: The dma channel specifier for rx operations. The format of > + the dma specifier depends on the dma controller. > + > +Optional SoC Specific Properties: > + > +- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel > + support, this flag is enabled. > +- samsung,supports-rstclr: This flag should be set if I2S software reset bit > + control is required. When this flag is set I2S software reset bit will be > + enabled or disabled based on need. > +- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, > + then this flag is enabled. > +- samsung,idma-addr: Internal DMA register base address of the audio > + sub system(used in secondary sound source). > + > +Required Board Specific Properties: > + > +- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK > + interface lines. The format of the gpio specifier depends on the gpio > + controller. > + The syntax of samsung gpio specifier is > + <[phandle of the gpio controller node] > + [pin number within the gpio controller] > + [mux function] > + [flags and pull up/down] > + [drive strength]> > + > +Aliases: > + > +- All the I2S controller nodes are represented in the aliases node using > + the following format 'i2s{n}' where n is a unique number for the alias. Are the aliases actually being used now? If not, then drop this section. I don't see any references in this patch. Otherwise the binding looks good to me. g.

4 3

[alsa-devel] sound: usb-audio: full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 23 Dec '12

23 Dec '12

Hi folks, I have put together full support for Digidesign Mbox 2, but it needs someone to help me merge it into alsa. I haven't seen the latest snd-usb-audio code since last year, it seems to have branched into multiple files. I have worked out how the Digidesign Mbox 2 needs to be configured, it supports SPDIF i/o simultaneously with analogue but can be toggled on and off by using different altsettings. Also the sample clock rate must be configured via a special method. Therefore I have provided for this in a device_setup parameter with 4 settings: 48Khz Spdif+analog = 0 44.1Khz Spdif+analog = 1 48Khz Analog only = 2 44.1Khz Analog only = 3 ###### audio /* Digidesign Mbox 2 workaround: * supports S24_3BE BIG ENDIAN in an unusual way */ if (chip->usb_id == USB_ID(0x0dba, 0x3000)) pcm_format = SNDRV_PCM_FORMAT_S24_3BE; static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { ..... ..... /* Digidesign Mbox 2: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0dba, 0x3000) && mbox2_skip_setting_quirk(chip, iface_no, altno)) continue; ..... } #define MBOX2_SET_48K_SPDIF 0 #define MBOX2_SET_44K_SPDIF 1 #define MBOX2_SET_48K_ANALOG 2 #define MBOX2_SET_44K_ANALOG 3 static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) { u8 srate[3]; /* Reset ifaces 2-5 to 0 altsetting. */ usb_set_interface(chip->dev, iface, 0); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF || device_setup[chip->index] == MBOX2_SET_48K_ANALOG) { //setup 48k srate[0]=0x80; srate[1]=0xbb; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 48000Hz"); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } else { //setup 44k srate[0]=0x44; srate[1]=0xac; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 44100Hz"); if (device_setup[chip->index] == MBOX2_SET_44K_SPDIF || device_setup[chip->index] == MBOX2_SET_44K_ANALOG) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } return 0; } #define MBOX2_FIRMWARE_SIZE 646 #define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ #define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) { struct usb_host_config *config = dev->actconfig; int err; u8 enablemagic[3]; u8 bootresponse; int fwsize; if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); /* From USB Snoop, * * SetupPacket = RT RQ VHVL INDX SIZE * RT = 0xRT Request Type * RQ = 0xRQ Request * VHVL = 0xVLVH Value in reverse byte order * INDX = 0xDXIN Index in reverse byte order * SIZE = 0xZESI Size in reverse byte order * * Magic boot code setup packet: c0 85 01 00 00 00 12 00 * RQ RT VLVH DXIN ZESI * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT * for snd_usb_ctl_msg() */ mbox2_reboot: snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); if (bootresponse == MBOX2_BOOT_LOADING) { snd_printdd("device not ready, resending boot sequence...\n"); goto mbox2_reboot; } if (bootresponse == MBOX2_BOOT_READY) { snd_printdd("device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); snd_printdd("mbox2_boot: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) * but now we will try sending capture/playback enable magic * * 80 bb 00 = 24bit mode - S24_3BE * 44 ac 00 = 16bit mode? * */ /* We want 48khz mode for now */ enablemagic[0]=0x80; enablemagic[1]=0xbb; enablemagic[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); return 0; /* Succesful boot */ } snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); return -ENODEV; } snd_printk("Invalid firmware size: %d\n",fwsize); return -ENODEV; } ####### quirk: { USB_DEVICE(0x0dba, 0x3000), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Digidesign", .product_name = "Mbox 2", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 6, .type = QUIRK_MIDI_MBOX2, .data = & (const struct snd_usb_midi_endpoint_info) { .out_ep = 0x02, .out_cables = 0x0001, .in_ep = 0x81, .in_interval = 0x01, .in_cables = 0x0001 } }, { .ifnum = -1 } } } }, ##### midi case QUIRK_MIDI_MBOX2: /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; /* * We have to make sure that the USB core looks * again at interface 6 by calling usb_set_interface() on it. */ usb_set_interface(umidi->chip->dev, 6, 0); memcpy(&endpoints[0], quirk->data, sizeof(struct snd_usb_midi_endpoint_info)); err = 0; break; Damien

4 26

[alsa-devel] [PATCH] ALSA: usb-audio: support delay calculation on capture streams
by Pierre-Louis Bossart 21 Dec '12

21 Dec '12

Enable delay report on capture path. The delay is reset when an URB is retired and increment at each call to .pointer based on frame counter changes. The precision of the delay information is limited to 1ms as in the playback case. This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..3a38447 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -59,7 +59,12 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, /* Approximation based on number of samples per USB frame (ms), some truncation for 44.1 but the estimate is good enough */ - est_delay = subs->last_delay - (frame_diff * rate / 1000); + est_delay = frame_diff * rate / 1000; + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + est_delay = subs->last_delay - est_delay; + else + est_delay = subs->last_delay + est_delay; + if (est_delay < 0) est_delay = 0; return est_delay; @@ -78,8 +83,7 @@ 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); @@ -1147,6 +1151,10 @@ static void retire_capture_urb(struct snd_usb_substream *subs, int i, period_elapsed = 0; unsigned long flags; unsigned char *cp; + int current_frame_number; + + /* read frame number here, update pointer in critical section */ + current_frame_number = usb_get_current_frame_number(subs->dev); stride = runtime->frame_bits >> 3; @@ -1180,6 +1188,15 @@ static void retire_capture_urb(struct snd_usb_substream *subs, subs->transfer_done -= runtime->period_size; period_elapsed = 1; } + /* capture delay is by construction limited to one URB, + * reset delays here + */ + runtime->delay = subs->last_delay = 0; + + /* realign last_frame_number */ + subs->last_frame_number = current_frame_number; + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ if (oldptr + bytes > runtime->buffer_size * stride) { -- 1.7.9.5

2 1