May 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] aplay: main:722: audio open error: No such file or directory
by Christopher Head 27 May '14

27 May '14

Hello! Apologies if this is not the right place to report this. I tried to use the bug tracker at <https://bugtrack.alsa-project.org/alsa-bug>, but both times I went to it (a couple of weeks ago and again today) it was offline. Also, I am not subscribed to the list, so please include me in replies (or tell me if this is not acceptable and I will subscribe). I can't play audio on non-default devices. Running aplay with no options, or with -D default, works fine. Running aplay with -D hdmi does not work, failing with this message: aplay: main:722: audio open error: No such file or directory Running it in strace, I see that the last interesting thing it does before printing that message is it tries to open /dev/snd/pcmC0D3p, which does not exist; it should probably be addressing pcmC1D3p instead. I have no ALSA configuration files anywhere. kruskal asound # aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=MID HDA Intel MID, ALC663 Analog Default Audio Device sysdefault:CARD=MID HDA Intel MID, ALC663 Analog Default Audio Device front:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, ALC663 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=MID,DEV=0 HDA Intel MID, ALC663 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 HDMI Audio Output kruskal asound # ls -1 /dev/snd by-path controlC0 controlC1 hwC0D0 hwC1D0 hwC1D1 hwC1D2 hwC1D3 pcmC0D0c pcmC0D0p pcmC0D1p pcmC0D2c pcmC1D3p pcmC1D7p pcmC1D8p pcmC1D9p seq timer kruskal asound # cat cards 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xf4900000 irq 53 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xf3080000 irq 17 kruskal asound # cat devices 1: : sequencer 2: [ 0- 2]: digital audio capture 3: [ 0- 1]: digital audio playback 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: [ 1- 9]: digital audio playback 9: [ 1- 8]: digital audio playback 10: [ 1- 7]: digital audio playback 11: [ 1- 3]: digital audio playback 12: [ 1- 3]: hardware dependent 13: [ 1- 2]: hardware dependent 14: [ 1- 1]: hardware dependent 15: [ 1- 0]: hardware dependent 16: [ 1] : control 33: : timer kruskal asound # cat pcm 00-00: ALC663 Analog : ALC663 Analog : playback 1 : capture 1 00-01: ALC663 Digital : ALC663 Digital : playback 1 00-02: ALC663 Alt Analog : ALC663 Alt Analog : capture 1 01-03: HDMI 0 : HDMI 0 : playback 1 01-07: HDMI 0 : HDMI 0 : playback 1 01-08: HDMI 0 : HDMI 0 : playback 1 01-09: HDMI 0 : HDMI 0 : playback 1 -- Christopher Head

2 2

[alsa-devel] [RFC][PATCH 00/13 v1] Enhancement for Dice driver
by Takashi Sakamoto 27 May '14

27 May '14

This patchset is for enhancement of Dice driver. This patchset is based on this series. (This series is under reviewing.) [alsa-devel] [PATCH 00/49 v4] Enhancement for support of Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075841.html If you try this patchset, please apply the series in advance then apply this patchset. Or just use backport codes in my github repository: https://github.com/takaswie/snd-firewire-improve Or if you use Ubuntu 14.04, DKMS package is available from David's PPA. (Special thanks to David Henningsson.) https://launchpad.net/~diwic/+archive/snd-firewire-improve If you have any Dice based devices, please test this patchset because I can't test this driver with actual devices. == Background Dice is 'Digital Interface Communications Engine', produced by TC Applied Technologies. Several vendors produced devices with Dice chipsets. Currently, ALSA driver for Dice chipset has some restrictions: - Support playback of PCM samples only - Support SYT-Match mode for sampling clock source only This patchset adds more functionality into the driver: - Support both of capture/playback - Support both of PCM/MIDI - Support both of SYT-Match or non SYT-Match mode == Call for testing I don't have any Dice based devices. So I work for this patchset with my experiences for the other drivers and some stub codes. So it may not work fine, perhaps. If you have some Dice based devices, I hope you to test these patches and report the result. If it works badly, please get transaction logs and send it to me. The way to get transaction logs is: 1. Set debug parameter of 'firewire-ohci' driver. With root privillege: $ echo 1 > /sys/module/firewire_ohci/parameters/debug 2. Then transaction logs are output to /var/log/syslog $ tail -n 10 /var/log/syslog kernel: [ 2376.107790] snd_dice fw1.0: transaction failed: address error kernel: [ 2376.108015] firewire_core 0000:02:00.0: created device fw1: GUID 00a0de000002e247, S400 kernel: [ 2382.044469] firewire_ohci 0000:02:00.0: AR evt_bus_reset, generation 25 kernel: [ 2384.095868] snd_dice fw1.0: transaction failed: bus reset kernel: [ 2384.143898] snd_dice fw1.0: transaction failed: bus reset kernel: [ 2389.517686] firewire_ohci 0000:02:00.0: AR evt_bus_reset, generation 26 kernel: [ 2390.015821] firewire_ohci 0000:02:00.0: AT spd 0 tl 1f, ffc0 -> ffc1, evt_missing_ack, QR req, fffff0000400 kernel: [ 2392.885455] firewire_ohci 0000:02:00.0: AR evt_bus_reset, generation 27 kernel: [ 2392.885504] firewire_core 0000:02:00.0: phy config: new root=ffc1, gap_count=5 kernel: [ 2392.885534] firewire_ohci 0000:02:00.0: AT ack_complete, PHY 01c50000 fe3affff kernel: [ 2392.885556] firewire_ohci 0000:02:00.0: AR evt_bus_reset, generation 28 kernel: [ 2393.025757] firewire_ohci 0000:02:00.0: AT spd 0 tl 2a, ffc1 -> ffc0, ack_pending , QR req, fffff0000400 kernel: [ 2393.026007] firewire_ohci 0000:02:00.0: AR spd 2 tl 2a, ffc0 -> ffc1, ack_complete, QR resp = 041fa8cb kernel: [ 2393.026076] firewire_ohci 0000:02:00.0: AT spd 0 tl 2b, ffc1 -> ffc0, ack_pending , QR req, fffff0000404 kernel: [ 2393.026172] firewire_ohci 0000:02:00.0: AR spd 2 tl 2b, ffc0 -> ffc1, ack_complete, QR resp = 31333934 kernel: [ 2393.026251] firewire_ohci 0000:02:00.0: AT spd 0 tl 2c, ffc1 -> ffc0, ack_pending , QR req, fffff0000408 kernel: [ 2393.030558] firewire_ohci 0000:02:00.0: AR spd 2 tl 2c, ffc0 -> ffc1, ack_complete, QR resp = f0646122 3. Please send me the output since you plug-in your device (not whole) 4. And please tell me how you test, which devices you test with and what result you got == All patches My work for ALSA firewire drivers are available in my remote repository: https://github.com/takaswie/sound It includes 77 patches. I'm always welcome to receive some comments about them. Takashi Sakamoto (13): dice: Rename structure and its members dice: Move file to its own directory dice: Split transaction functionality into a file dice: Split stream functionality into a file dice: Split PCM functionality into a file dice: Split hwdep functionality into a file dice: Split proc functionality into a file dice: Add new functions for limitation of PCM parameters dice: Change the way to start stream dice: Add support for duplex streams with synchronization dice: Support for non SYT-Match sampling clock source mode dice: Add support for PCM capture dice: Add support for MIDI capture/playback sound/firewire/Kconfig | 4 +- sound/firewire/Makefile | 3 +- sound/firewire/dice-interface.h | 371 -------- sound/firewire/dice.c | 1500 -------------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-interface.h | 371 ++++++++ sound/firewire/dice/dice.c | 374 ++++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/dice/dice_hwdep.c | 190 ++++ sound/firewire/dice/dice_midi.c | 173 ++++ sound/firewire/dice/dice_pcm.c | 389 +++++++++ sound/firewire/dice/dice_proc.c | 252 ++++++ sound/firewire/dice/dice_stream.c | 410 +++++++++ sound/firewire/dice/dice_transaction.c | 359 ++++++++ 14 files changed, 2712 insertions(+), 1876 deletions(-) delete mode 100644 sound/firewire/dice-interface.h delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-interface.h create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/dice/dice_hwdep.c create mode 100644 sound/firewire/dice/dice_midi.c create mode 100644 sound/firewire/dice/dice_pcm.c create mode 100644 sound/firewire/dice/dice_proc.c create mode 100644 sound/firewire/dice/dice_stream.c create mode 100644 sound/firewire/dice/dice_transaction.c -- 1.8.3.2

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[alsa-devel] [PATCH] ALSA: hda: fix tegra build
by Arnd Bergmann 26 May '14

26 May '14

When CONFIG_PM is disabled, the CONFIG_SND_HDA_POWER_SAVE_DEFAULT symbol does not get defined, which causes a build error for the hda-tegra driver: hda/hda_tegra.c:80:25: error: 'CONFIG_SND_HDA_POWER_SAVE_DEFAULT' undeclared here (not in a function) static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; ^ /git/arm-soc/sound/pci/hda/hda_tegra.c:235:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function] static void hda_tegra_disable_clocks(struct hda_tegra *data) ^ This works around the problem by not referencing that macro when CONFIG_PM is disabled. Instead, we assume that it's disabled unconditionally and cannot be enabled at runtime. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> Reviewed-by: Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 086fd06..a366ba9 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -77,10 +77,14 @@ struct hda_tegra { void __iomem *regs; }; +#ifdef CONFIG_PM static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); +#else +static int power_save = 0; +#endif /* * DMA page allocation ops.

2 2

[alsa-devel] [PATCH 1/2] ASoC: add api for dapm kcontrol configiuration
by Vinod Koul 26 May '14

26 May '14

From: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> For DSPs we need to set and get the value for snd_kcontrol. This is currently done by dapm_kcontrol_set/get_value, so create a wrapper snd_soc_dapm_kcontrol_get/set_value APIs to let drivers use this Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc-dapm.h | 4 ++++ sound/soc/soc-dapm.c | 13 +++++++++++++ 2 files changed, 17 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 75020f5..9e826e9 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -368,6 +368,10 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +bool snd_soc_dapm_kcontrol_set_value( + const struct snd_kcontrol *kcontrol, unsigned int value); +unsigned int snd_soc_dapm_kcontrol_get_value( + const struct snd_kcontrol *kcontrol); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 98c1dc6..b7dfa1a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -333,6 +333,12 @@ static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) return data->value; } +unsigned int snd_soc_dapm_kcontrol_get_value( + const struct snd_kcontrol *kcontrol) +{ + dapm_kcontrol_get_value(kcontrol); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_get_value); static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) @@ -350,6 +356,13 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, return true; } +bool snd_soc_dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, + unsigned int value) +{ + return dapm_kcontrol_set_value(kcontrol, value); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_set_value); + /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol -- 1.7.0.4

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[alsa-devel] Fwd: Re: No sound for a rather old laptop (ALC260).
by Fidel Leon Martin 26 May '14

26 May '14

Hello, list. Is there any way to know if this support request has been taken by anyone...? ---------- Forwarded Message ---------- Subject: Re: [alsa-devel] No sound for a rather old laptop (ALC260). Date: Tuesday 06 May 2014, 16:40:56 From: Fidel Leon <fidelleon(a)mykolab.com> To: Takashi Iwai <tiwai(a)suse.de>, alsa-devel(a)alsa-project.org On Tuesday 06 May 2014 11:27:07 you wrote: > At Mon, 05 May 2014 16:34:38 +0200, > > Fidel Leon wrote: > > I own a rather old laptop which features a snd_hda_intel soundcard, with a > > ALC260 chipset. > > > > I know it works when using rather old kernels (Debian 7, for example), and > > > > I already have an alsa-info.txt file to debug, may I post it here? > > Please attach in your post. (But maybe better to compress so that it > won't hit the max size of ML.) Also, at best, take alsa-info.sh from > the working state with the old kernel, so that we can compare > directly. Many thanks, Takashi. I've tested a Debian 6 live image and my soundcard runs fine, so I can tell my soundcard works. Attached are alsa-infos from kernel 2.6 (which works) and 3.14 (Fedora 20, which doesnt work). Regards, ----------------------------------------- -- Fidel Leon fidelleon(a)mykolab.com GPG: AD74FB97

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[alsa-devel] [PATCH] ASoC: samsung: Use params_width()
by Tushar Behera 26 May '14

26 May '14

commit 8c5178fca4ce ("ALSA: Add params_width() helpers") introduces a helper to get the sample width. Updating Samsung related sound drivers to use this helper. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- sound/soc/samsung/i2s.c | 8 ++++---- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/s3c-i2s-v2.c | 8 ++++---- sound/soc/samsung/s3c2412-i2s.c | 6 +++--- sound/soc/samsung/s3c24xx-i2s.c | 6 +++--- sound/soc/samsung/smdk_wm8580.c | 8 +++----- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/spdif.c | 4 ++-- 8 files changed, 22 insertions(+), 24 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 07ff3e7..6b96168 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -672,8 +672,8 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, if (is_manager(i2s)) mod &= ~MOD_BLC_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: if (is_secondary(i2s)) mod |= MOD_BLCS_8BIT; else @@ -681,7 +681,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, if (is_manager(i2s)) mod |= MOD_BLC_8BIT; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: if (is_secondary(i2s)) mod |= MOD_BLCS_16BIT; else @@ -689,7 +689,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, if (is_manager(i2s)) mod |= MOD_BLC_16BIT; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: if (is_secondary(i2s)) mod |= MOD_BLCS_24BIT; else diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index a3c9c9c..4c5f97f 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -283,8 +283,8 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); /* Strictly check for sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; default: return -EINVAL; diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 77a2ae5..0ff4bbe 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -322,13 +322,13 @@ static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, iismod &= ~S3C64XX_IISMOD_BLC_MASK; /* Sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: iismod |= S3C64XX_IISMOD_BLC_8BIT; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iismod |= S3C64XX_IISMOD_BLC_24BIT; break; } diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 843f315..08c059b 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -120,11 +120,11 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: iismod |= S3C2412_IISMOD_8BIT; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: iismod &= ~S3C2412_IISMOD_8BIT; break; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 4a6d206..9aba9fb 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -248,12 +248,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); pr_debug("hw_params r: IISMOD: %x\n", iismod); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: iismod &= ~S3C2410_IISMOD_16BIT; dma_data->dma_size = 1; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: iismod |= S3C2410_IISMOD_16BIT; dma_data->dma_size = 2; break; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 7a16b32..b1a519f 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -37,13 +37,11 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, unsigned int pll_out; int bfs, rfs, ret; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: bfs = 16; break; - case SNDRV_PCM_FORMAT_U16_LE: - case SNDRV_PCM_FORMAT_S16_LE: + case 16: bfs = 32; break; default: diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index fc25cc0..3d6272a 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -57,7 +57,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, int ret; /* AIF1CLK should be >=3MHz for optimal performance */ - if (params_format(params) == SNDRV_PCM_FORMAT_S24_LE) + if (params_width(params) == 24) pll_out = params_rate(params) * 384; else if (params_rate(params) == 8000 || params_rate(params) == 11025) pll_out = params_rate(params) * 512; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index e93a93e..d9ffc48 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -211,8 +211,8 @@ static int spdif_hw_params(struct snd_pcm_substream *substream, con |= CON_PCM_DATA; con &= ~CON_PCM_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: con |= CON_PCM_16BIT; break; default: -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: sirf-audio-codec: Simplify the new bitmask value in regmap_update_bits
by Axel Lin 26 May '14

26 May '14

Having the binary ones complement operator in the new bitmak value makes the code hard to read. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- I think the binary ones complement operator in such use case is error prone: For example: To set BIT(1) and clear both BIT(2) and BIT(3): The following code has wrong result: regmap_update_bits(regmap, the_register, BIT(1), BIT(2), BIT(3), BIT(1) | ~BIT(2) | ~BIT(3)); Axel sound/soc/codecs/sirf-audio-codec.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index c5177bc..d90cb0f 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -109,7 +109,7 @@ static void enable_and_reset_codec(struct regmap *regmap, { regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, codec_enable_bits | codec_reset_bits, - codec_enable_bits | ~codec_reset_bits); + codec_enable_bits); msleep(20); regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, codec_reset_bits, codec_reset_bits); @@ -128,8 +128,7 @@ static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(sirf_audio_codec->regmap, - AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS, - ~ATLAS6_CODEC_ENABLE_BITS); + AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS, 0); break; default: break; @@ -151,8 +150,7 @@ static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(sirf_audio_codec->regmap, - AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS, - ~PRIMA2_CODEC_ENABLE_BITS); + AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS, 0); break; default: break; -- 1.8.3.2

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[alsa-devel] [PATCH RFC 03/11] sound: ASoC: kirkwood: Remove ARCH_KIRKWOOD dependency
by Andrew Lunn 26 May '14

26 May '14

mach-kirkwood has been removed, now that kirkwood lives in mach-mvebu. Signed-off-by: Andrew Lunn <andrew(a)lunn.ch> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/kirkwood/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 1f7c7ee3527a..6a0f79d2bb33 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST + depends on ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- 2.0.0.rc2

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[alsa-devel] [PATCH 1/4] ASoC: tegra: Replace instances of rtd->codec->card with rtd->card
by Lars-Peter Clausen 26 May '14

26 May '14

No need to go via the CODEC to get a pointer to the card. This will help to eventually remove the card field from the snd_soc_codec struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/tegra/tegra_alc5632.c | 5 ++--- sound/soc/tegra/tegra_max98090.c | 5 ++--- sound/soc/tegra/tegra_rt5640.c | 5 ++--- sound/soc/tegra/tegra_wm8753.c | 3 +-- sound/soc/tegra/tegra_wm8903.c | 5 ++--- sound/soc/tegra/trimslice.c | 3 +-- 6 files changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c61ea3a..21ad9bd 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -41,8 +41,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -105,7 +104,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 0283cfb..b867b27 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -49,8 +49,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -127,7 +126,7 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_max98090 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 4511c5a..d720bd7 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -51,8 +51,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -110,7 +109,7 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, &tegra_rt5640_hp_jack); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index 8e774d1..769e28f 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -55,8 +55,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 4ac7373..e795424 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -60,8 +60,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -173,7 +172,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); if (gpio_is_valid(machine->gpio_hp_det)) { diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 734bfcd..589d2d9 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -50,8 +50,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; int err; -- 1.8.0

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[alsa-devel] [PATCH V2 0/2] ASoC: max98090/max98095: Add master clock handing
by Tushar Behera 26 May '14

26 May '14

These CODECs are used on Snow/Peach-pit boards and the master clock needs to be set at 24MHz. Adding support in respective codec drivers so that we can update the master clock frequency from sound card driver through set_sysclk. Changes for V2: 1. Added PROBE_DEFER if clock driver has not yet been probed 2. Moved clk_prepare_enable from .probe to .set_bias_level 3. Moved clk_disable_unprepare from .remove to .set_bias_level Tushar Behera (2): ASoC: max98090: Add master clock handling ASoC: max98095: Add master clock handling .../devicetree/bindings/sound/max98090.txt | 6 +++++ .../devicetree/bindings/sound/max98095.txt | 6 +++++ sound/soc/codecs/max98090.c | 23 +++++++++++++++++++ sound/soc/codecs/max98090.h | 1 + sound/soc/codecs/max98095.c | 24 ++++++++++++++++++++ 5 files changed, 60 insertions(+) -- 1.7.9.5

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