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

[alsa-devel] [PATCH] ALSA: hda - Add support for CX20952
by Takashi Iwai 06 Nov '13

06 Nov '13

It's a superset of the existing CX2075x codecs, so we can reuse the existing parser code. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 993b25c17711..f0e4c56acbd1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3656,6 +3656,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f15115, .name = "CX20757", .patch = patch_conexant_auto }, + { .id = 0x14f151d7, .name = "CX20952, + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -3682,6 +3684,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15111"); MODULE_ALIAS("snd-hda-codec-id:14f15113"); MODULE_ALIAS("snd-hda-codec-id:14f15114"); MODULE_ALIAS("snd-hda-codec-id:14f15115"); +MODULE_ALIAS("snd-hda-codec-id:14f151d7"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- 1.8.4.2

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[alsa-devel] [PATCH 1/2] ALSA: cmi8328: Fix compile warnings without CONFIG_PM
by Takashi Iwai 06 Nov '13

06 Nov '13

Just add an ifdef CONFIG_PM to shut up the warnings: sound/isa/cmi8328.c:129:13: warning: ‘snd_cmi8328_cfg_save’ defined but not used [-Wunused-function] sound/isa/cmi8328.c:136:13: warning: ‘snd_cmi8328_cfg_restore’ defined but not used [-Wunused-function] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/cmi8328.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index f84f073fc1e8..ab6b2dc043f1 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -126,6 +126,7 @@ static void snd_cmi8328_cfg_write(u16 port, u8 reg, u8 val) outb(val, port + 3); /* yes, value goes to the same port as index */ } +#ifdef CONFIG_PM static void snd_cmi8328_cfg_save(u16 port, u8 cfg[]) { cfg[0] = snd_cmi8328_cfg_read(port, CFG1); @@ -139,6 +140,7 @@ static void snd_cmi8328_cfg_restore(u16 port, u8 cfg[]) snd_cmi8328_cfg_write(port, CFG2, cfg[1]); snd_cmi8328_cfg_write(port, CFG3, cfg[2]); } +#endif /* CONFIG_PM */ static int snd_cmi8328_mixer(struct snd_wss *chip) { -- 1.8.4.2

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[alsa-devel] How to apply rate conversion in asound.conf
by Dolevo Jay 06 Nov '13

06 Nov '13

Hi experts, I am currently using the following asound.conf, which is pretty much copy-paste from the link below. asound.conf ------------------------------------- pcm.!default { type plug slave.pcm multi } pcm.multi { type multi; slaves.a.pcm "hw:0,1"; slaves.a.channels 2; slaves.b.pcm "hw:1,1"; slaves.b.channels 2; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave b; bindings.2.channel 0; bindings.3.slave b; bindings.3.channel 1; } ------------------------------------- http://www.alsa-project.org/main/index.php/Asoundrc#Virtual_multi_channel_d… ------------------------------------- My purpose is to apply rate conversion to the "hw:1,1" and have 44100 rate out. There is a sample conversion like: pcm.rate_convert { type rate slave sl2 } but how can I attach this rate_convert code to the multi code? Thanks.

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[alsa-devel] [PATCH 1/1] ALSA: HDA - Limit mic boost and add mute LED for an HP machine
by David Henningsson 06 Nov '13

06 Nov '13

This machine has a mute LED as well as a noisy internal mic. Hence it needs quirks for both limiting the mic boost as well as enabling the LED. BugLink: https://bugs.launchpad.net/bugs/1248476 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 01bf812..a51f48c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3565,6 +3565,7 @@ enum { ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, @@ -3834,6 +3835,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, }, + [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269_FIXUP_HP_MUTE_LED_MIC1, + }, [ALC269VB_FIXUP_ORDISSIMO_EVE2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3915,6 +3922,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: rcar: remove un-needed select from Kconfig
by Kuninori Morimoto 06 Nov '13

06 Nov '13

config RCAR_CLK_ADG is not exist Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 56d8ff6..14011d9 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,7 +37,6 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD - select RCAR_CLK_ADG help This option enables R-Car SUR/SCU/SSIU/SSI sound support -- 1.7.9.5

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[alsa-devel] [PATCH 0/9] Kill BUG() for ASoC
by Takashi Iwai 06 Nov '13

06 Nov '13

Hi, this is the revised patches for replacing BUG() with WARN(). Indeed this gives even better information than a simple BUG(), so drop the previous patches to replace snd_BUG(). [PATCH 1/9] ASoC: wm5100: Replace BUG() with WARN() [PATCH 2/9] ASoC: wm8350: Replace BUG() with WARN() [PATCH 3/9] ASoC: wm8900: Replace BUG() with WARN() [PATCH 4/9] ASoC: wm8904: Replace BUG() with WARN() [PATCH 5/9] ASoC: wm8958: Replace BUG() with WARN() [PATCH 6/9] ASoC: wm8962: Replace BUG() with WARN() [PATCH 7/9] ASoC: wm8996: Replace BUG() with WARN() [PATCH 8/9] ASoC: wm_hubs: Replace BUG() with WARN() [PATCH 9/9] ASoC: Replace BUG() with WARN() thanks, Takashi

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[alsa-devel] [PATCH] ALSA: hda - Make sure mute LEDs stay on during runtime suspend (Realtek)
by David Henningsson 06 Nov '13

06 Nov '13

Some HP machines with Realtek codecs have mute LEDs connected to VREF pins. However when these go into runtime suspend, the pin powers down and its pin control is disabled, thus disabling the LED too. This patch fixes that issue by making sure that the pin stays in D0 with correct pin control. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/1248465 Tested-by: Franz Hsieh <franz.hsieh(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a51f48c..0acbe82 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2955,6 +2955,23 @@ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval); } +/* Make sure the led works even in runtime suspend */ +static unsigned int led_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) +{ + struct alc_spec *spec = codec->spec; + + if (power_state != AC_PWRST_D3 || nid != spec->mute_led_nid) + return power_state; + + /* Set pin ctl again, it might have just been set to 0 */ + snd_hda_set_pin_ctl(codec, nid, + snd_hda_codec_get_pin_target(codec, nid)); + + return AC_PWRST_D0; +} + static void alc269_fixup_hp_mute_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2974,6 +2991,7 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec, spec->mute_led_nid = pin - 0x0a + 0x18; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; snd_printd("Detected mute LED for %x:%d\n", spec->mute_led_nid, spec->mute_led_polarity); break; @@ -2989,6 +3007,7 @@ static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, spec->mute_led_nid = 0x18; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; } } @@ -3001,6 +3020,7 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, spec->mute_led_nid = 0x19; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; } } -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: ab8500: Revert "ASoC: ab8500: Explicitly set I/O up"
by Lee Jones 06 Nov '13

06 Nov '13

This reverts commit 2245e3c31c15c2d2a26926c4b734f4d3a37ae252, as it is one of two patches which stop audio from working on Snowball. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/codecs/ab8500-codec.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index a0394a8..ca500fd 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2473,8 +2473,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation
by Oskar Schirmer 06 Nov '13

06 Nov '13

Originally snd_hrtimer_callback() used iprtd->period_time for some jiffies based estimation to determine the right moment to call snd_pcm_period_elapsed(). As timer drifts may well be a problem, this was changed in commit b4e82b5b785670b6 to be based on buffer transmission progress, using iprtd->offset and runtime->buffer_size to calculate the amount of data since last period had elapsed. Unfortunately, iprtd->offset counts in bytes, while runtime->buffer_size counts frames, so adding these to find some delta is like comparing apples and oranges, and eventually results in negative delta values every now and then. This is no big harm, because it simply causes snd_pcm_period_elapsed() being called more often than necessary, as negative delta is taken for a large unsigned value by implicit conversion rule. Nonetheless, the calculation is broken, so one would replace the runtime->buffer_size by its equivalent in bytes. But then, there are chances snd_pcm_period_elapsed() is called late, because calculating the moment for the elapsed period into delta is based against the iprtd->last_offset, which is not necessarily the first byte of the period in question, but some random byte which the FIQ handler left us with in r8/r9 by accident. Again, negative impact is low, as there are plenty of periods already prefilled with data, and snd_pcm_period_elapsed() will probably be called latest when the following period is reached. However, the calculation is conceptually broken, and we are best off removing the clever stuff altogether. snd_pcm_period_elapsed() is now simply called once everytime snd_hrtimer_callback() is run, which may not be most accurate, but at least this way we are quite sure we dont miss an end of period. There is not much extra effort wasted by superfluous calls to snd_pcm_period_elapsed(), as the timer frequency closely matches the period size anyway. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/fsl/imx-pcm-fiq.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 34043c5..94720a6 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -39,8 +39,6 @@ struct imx_pcm_runtime_data { unsigned int period; int periods; unsigned long offset; - unsigned long last_offset; - unsigned long size; struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; @@ -54,7 +52,6 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct snd_pcm_substream *substream = iprtd->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct pt_regs regs; - unsigned long delta; if (!atomic_read(&iprtd->running)) return HRTIMER_NORESTART; @@ -66,19 +63,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) else iprtd->offset = regs.ARM_r9 & 0xffff; - /* How much data have we transferred since the last period report? */ - if (iprtd->offset >= iprtd->last_offset) - delta = iprtd->offset - iprtd->last_offset; - else - delta = runtime->buffer_size + iprtd->offset - - iprtd->last_offset; - - /* If we've transferred at least a period then report it and - * reset our poll time */ - if (delta >= iprtd->period) { - snd_pcm_period_elapsed(substream); - iprtd->last_offset = iprtd->offset; - } + snd_pcm_period_elapsed(substream); hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); @@ -95,11 +80,9 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params) ; + iprtd->period = params_period_bytes(params); iprtd->offset = 0; - iprtd->last_offset = 0; iprtd->poll_time_ns = 1000000000 / params_rate(params) * params_period_size(params); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: hda - Force buffer alignment for Haswell HDMI controllers
by Takashi Iwai 06 Nov '13

06 Nov '13

Haswell HDMI audio controllers seem to get stuck when unaligned buffer size is used. Let's enable the buffer alignment for the corresponding entries. Since AZX_DCAPS_INTEL_PCH contains AZX_DCAPS_BUFSIZE that disables the buffer alignment forcibly, define AZX_DCAPS_INTEL_HASWELL and put the necessary AZX_DCAPS bits there. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=60769 Reported-by: Alexander E. Patrakov <patrakov(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d6dcec768594..9cbd1259a5be 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -613,6 +613,11 @@ enum { #define AZX_DCAPS_INTEL_PCH \ (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) +#define AZX_DCAPS_INTEL_HASWELL \ + (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \ + AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME | \ + AZX_DCAPS_I915_POWERWELL) + /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ (AZX_DCAPS_ATI_SNOOP | AZX_DCAPS_NO_TCSEL | \ @@ -3992,14 +3997,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | - AZX_DCAPS_I915_POWERWELL }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, { PCI_DEVICE(0x8086, 0x0c0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | - AZX_DCAPS_I915_POWERWELL }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, { PCI_DEVICE(0x8086, 0x0d0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | - AZX_DCAPS_I915_POWERWELL }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_HASWELL }, /* 5 Series/3400 */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, -- 1.8.4.2

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