December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare
by Jarkko Nikula 02 Dec '13

02 Dec '13

N810 audio driver has stopped working at some point. Probably when OMAP2 was converted to common clock framework since now call to clk_enable dumps the stack trace in drivers/clk/clk.c: __clk_enable() due clk->prepare_count is zero. Fix this by converting clk_enable/_disable calls to those that take care of clock prepare/unprepare. I'm not queueing this to linux-stable since OMAP2 common clock framework conversion in commit ed1ebc4948fd ("ARM: OMAP2: clock: Convert to common clk") happened before N810 was really usable in mainline and user base for N810 is anyway small. Potential linux-stable candidates are only those after commit 3d3a6d18abc6 ("watchdog: introduce retu_wdt driver"). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- sound/soc/omap/n810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6d216cb6c19b..3fde9e402710 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); n810_ext_control(&codec->dapm); - return clk_enable(sys_clkout2); + return clk_prepare_enable(sys_clkout2); } static void n810_shutdown(struct snd_pcm_substream *substream) { - clk_disable(sys_clkout2); + clk_disable_unprepare(sys_clkout2); } static int n810_hw_params(struct snd_pcm_substream *substream, -- 1.8.4.3

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[alsa-devel] [PATCH v3] ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface
by Nicolin Chen 02 Dec '13

02 Dec '13

The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v3: * Fixed conflict for git apply. v2: * Moved i2s_mode to ssi_private so that we can save and retore it as needed * And dropped the horrible static thing. sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fb8f52a..83e4ed8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -143,6 +143,7 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -354,14 +355,13 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -378,7 +378,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -508,6 +508,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -537,6 +538,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -649,14 +655,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.8.4

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[alsa-devel] Why does `aplay -L' not show PCM's defined in ~/.asoundrc ?
by Gabriel M. Beddingfield 02 Dec '13

02 Dec '13

Hi guys, Why does `aplay -L' not list PCM's that are defined in ~/.asoundrc (i.e. PCM's implemented by plugins)? How do you get a listing that includes these? I was helping a user today who needed to set up a device using the `multi' plugin -- and the fact that the new virtual pcm device didn't show up in any standard listing made things very confusing. -gabriel

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[alsa-devel] [PATCH] ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 02 Dec '13

02 Dec '13

The snd_soc_dai_digital_mute() here will be never executed because we only decrease codec->active in snd_soc_close(). Thus correct it. Signed-off-by: Nicolin Chen <b42378(a)freescale.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 22af038..7d19117 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -689,7 +689,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - if (!codec->active) + if (codec->active == 1) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ -- 1.8.4

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[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface
by Nicolin Chen 02 Dec '13

02 Dec '13

The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v2: * Moved i2s_mode to ssi_private so that we can save and retore it as needed * And dropped the horrible static thing. sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f43be6d..cba5c4d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -144,6 +144,7 @@ struct fsl_ssi_private { bool imx_ac97; bool use_dma; bool use_dual_fifo; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -325,14 +326,13 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -349,7 +349,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -517,6 +517,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -546,6 +547,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -658,14 +664,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.8.4

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[alsa-devel] [PATCH 1/2] ALSA: hda/analog - Handle inverted EAPD properly in vmaster hook
by Takashi Iwai 02 Dec '13

02 Dec '13

ad_vmaster_eapd_hook() needs to handle the inverted EAPD case properly, too. Otherwise the output gets broken on Lenovo N100 with AD1986A codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 34d86ec5d3dd..f6351b8e0f5d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -147,6 +147,8 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) if (!spec->eapd_nid) return; + if (codec->inv_eapd) + enabled = !enabled; snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); -- 1.8.4.4

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[alsa-devel] [PATCH] ALSA: hda - Another fixup for ASUS laptop with ALC660 codec
by Takashi Iwai 02 Dec '13

02 Dec '13

ASUS Z35HL laptop also needs the very same fix as the previous one that was applied to ASUS W7J. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a98f3296b891..9ca83001fc37 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4559,6 +4559,7 @@ static const struct hda_fixup alc861_fixups[] = { static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J), + SND_PCI_QUIRK(0x1043, 0x1263, "ASUS Z35HL", ALC660_FIXUP_ASUS_W7J), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), -- 1.8.4.4

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[alsa-devel] HD-Audio: How to reduce driver initializaton time if multiple codecs present on the bus?
by Lin, Mengdong 02 Dec '13

02 Dec '13

Hi Takashi, We're trying to reduce the HD-A driver initialization time when more than one codecs are connected to the bus, but are blocked. Would you please share some advices on this? Usually, there is one HD-A controller connecting to two codecs: one on-board codec and one integrated display codec. During initialization, the codecs are created and configured in a serial way. Creating a codec may cost 6~20ms, and then building controls make cost about 15~30ms. So I had thought it's helpful to build controls for different codecs in parallel in function snd_hda_build_controls(), as we did in driver suspend/resume. But the test shows this doesn't work: (1) Using a workqueue without WQ_UNBOUND, the work items, which calls snd_hda_codec_build_controls(), are processed one after one (I think because there is no explicit sleep in the work function). So the total time does not change. (2) Using an unbouned work queue for the bus or using separate kthreads for each codec, the work items for different codecs are processed in parallel. However, the total time is still the sum of two codec as that without parallel. I think that because snd_hda_codec_read() are used during initialization, the command execution on bus is serialized in the low level. So even if we parallel initializing two codecs in high level, commands on bus are still processed one by one so the total time cannot reduce. We're blocked here. Is there any other means to reduce the time? Thanks Mengdong

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[alsa-devel] [PATCH] ALSA: hda - Fix bad EAPD setup for HP machines with AD1984A
by Takashi Iwai 02 Dec '13

02 Dec '13

It seems that EAPD on NID 0x16 is the only control over all outputs on HP machines with AD1984A while turning EAPD on NID 0x12 breaks the output. Thus we need to avoid fiddling EAPD on NID. As a quick workaround, just set own_eapd_ctrl flag for the wrong EAPD, then implement finer EAPD controls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66321 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a83559f4cbd..34d86ec5d3dd 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -962,6 +962,7 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook; + spec->gen.own_eapd_ctl = 1; snd_hda_sequence_write_cache(codec, gpio_init_verbs); break; case HDA_FIXUP_ACT_PROBE: -- 1.8.4.4

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[alsa-devel] [PATCH] ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced situation
by Oskar Schirmer 02 Dec '13

02 Dec '13

Unbalanced calls to imx_ssi_trigger() may result in endless SSI activity and thus provoke eternal sound. While on the first glance, the switch statement looks pretty symmetric, the SUSPEND/RESUME pair is not: the suspend case comes along snd_pcm_suspend_all(), which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(), but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME. This way userland may provoke an unbalanced resume, which might cause the ssi->enabled counter to increase and never return to zero again, so eventually SSI_SCR_SSIEN is never disabled. Simply removing the ssi->enabled will solve the problem, as long as one never goes play and capture game simultaneously, but beware trying both at once, the early TRIGGER_STOP will cut off the other activity prematurely. So now playing and capturing is scrutinized separately, instead of by counting. This is essentially the same stuff as in sound/soc/fsl/imx-pcm-fiq.c which I send a patch for three days ago. Astonishing enough this highly fragile scheme is used twice in parallel to serve the very same control function, synchronously: Once out of sync you are lost until reboot. Note, that these fixes wont prevent state machine distortion on alsa level to cut sound or the like. It just makes sure we have a chance to synchronise again later on. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/fsl/imx-ssi.c | 19 ++++++++++++------- sound/soc/fsl/imx-ssi.h | 3 ++- 2 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f5f248c..c68499b 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -298,27 +298,32 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr |= SSI_SCR_TE; - else + ssi->playing = 1; + } else { scr |= SSI_SCR_RE; + ssi->capturing = 1; + } sier |= sier_bits; - if (++ssi->enabled == 1) - scr |= SSI_SCR_SSIEN; + scr |= SSI_SCR_SSIEN; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr &= ~SSI_SCR_TE; - else + ssi->playing = 0; + } else { scr &= ~SSI_SCR_RE; + ssi->capturing = 0; + } sier &= ~sier_bits; - if (--ssi->enabled == 0) + if (!ssi->playing && !ssi->capturing) scr &= ~SSI_SCR_SSIEN; break; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index 560c40f..a3c90d5 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -213,7 +213,8 @@ struct imx_ssi { int fiq_init; int dma_init; - int enabled; + int playing; + int capturing; }; #endif /* _IMX_SSI_H */ -- 1.7.9.5

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