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

Re: [alsa-devel] [PATCH V3 2/3] ASoC: Samsung: I2S: Add quirks as driver data in I2S
by Padma Venkat 08 Aug '13

08 Aug '13

Hi Tomasz, On Wed, Aug 7, 2013 at 4:43 PM, Tomasz Figa <t.figa(a)samsung.com> wrote: > Ahh, one more thing inline. > [snip] >> >> +static struct samsung_i2s_dai_data i2sv3_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv4_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_c100_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR | QUIRK_SEC_DAI | >> + QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data samsung_dai_type_sec = { >> + .dai_type = TYPE_SEC, >> +}; >> + >> static struct platform_device_id samsung_i2s_driver_ids[] = { >> { >> - .name = "samsung-i2s", >> - .driver_data = TYPE_PRI, >> + .name = "samsung,s3c6410-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv3_dai_type, >> + }, { >> + .name = "samsung,s3c6410-i2s-multi", >> + .driver_data = (kernel_ulong_t)&i2sv4_dai_type, >> + }, { >> + .name = "samsung,s5pc100-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_c100_dai_type, >> }, { >> - .name = "samsung-i2s-sec", >> - .driver_data = TYPE_SEC, >> + .name = "samsung,s5pv210-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_dai_type, >> + }, { >> + .name = "samsung-i2s-sec", >> + .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, > > I don't think you need to change the legacy platform IDs at all. If legacy platforms not required to change then I need to introduce a new samsung_i2s_dai_data structure which holds only dai_type for non-dt platforms. If I change legacy platforms it breaks the older platforms now. Is it okay adding a samsung_i2s_dai_data structure for non-dt platforms which will be removed later? > > IMHO it would be better to keep the old way of quirk retrieval from > platform_data (which I think this patch does anyway, because the probe path > without DT is unchanged), as it will be dropped in some point in time > anyway. > > This would also eliminate the need for patch 1. > > Best regards, > Tomasz > Thanks Padma

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[alsa-devel] snd_pcm_set_params typo patch
by Laurentiu Nicola 08 Aug '13

08 Aug '13

Please, somebody apply this one. I lost quite a bit of time because of it :). Also, is there anything wrong with the bug tracker? It doesn't load for me. diff -urpN alsa-lib-1.0.27.2/src/pcm/pcm.c alsa-lib-1.0.27.2-mine/src/pcm/pcm.c --- alsa-lib-1.0.27.2/src/pcm/pcm.c 2013-07-08 15:31:36.000000000 +0300 +++ alsa-lib-1.0.27.2-mine/src/pcm/pcm.c 2013-08-07 22:17:14.908673700 +0300 @@ -7932,7 +7932,7 @@ int snd_pcm_set_params(snd_pcm_t *pcm, return err; } if (rrate != rate) { - SNDERR("Rate doesn't match (requested %iHz, get %iHz)", rate, err); + SNDERR("Rate doesn't match (requested %iHz, got %iHz)", rate, rrate); return -EINVAL; } /* set the buffer time */ -- Because there are many ways to skin a cat, but you only get to choose one.

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[alsa-devel] [PATCH] ALSA: don't push static constants on stack for %*ph
by Andy Shevchenko 08 Aug '13

08 Aug '13

There is no need to pass constants via stack. The width may be explicitly specified in the format. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/isa/gus/interwave.c | 3 +-- sound/usb/6fire/firmware.c | 4 ++-- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 9942691..afef0d7 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) for (i = 0; i < 8; ++i) iwave[i] = snd_gf1_peek(gus, bank_pos + i); #ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos, - 8, iwave); + printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave); #endif if (strncmp(iwave, "INTRWAVE", 8)) continue; /* first check */ diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index b9defcd..780bf3f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version) if (!memcmp(version, known_fw_versions + i, 2)) return 0; - snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " + snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. " "please reconnect to power. if this failure " "still happens, check your firmware installation.", - 4, version); + version); return -EINVAL; } -- 1.8.4.rc1

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[alsa-devel] Alsa bug/bugtracker
by Mattia Bertoni 08 Aug '13

08 Aug '13

hi, under Mark Brown directions I submit to you my problem: recently I've gotten a crash on my machine running Archlinux x86_64 the log suggest me to report it in the Alsa bugtracker but the link on the official project (https://bugtrack.alsa-project.org/alsa-bug/) is down.... how can I do ??? thank you

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[alsa-devel] [PATCH] Fix invalid volume resolution for Logitech HD Webcam C525
by Бойко Максим 08 Aug '13

08 Aug '13

From: Maksim Boyko <maksim.a.boyko(a)gmail.com> Add the volume control quirk for avoiding the kernel warning for Logitech HD Webcam C525. The similar patch was previously reported for Logitech HD Webcam C310 (see 36691e1be6ec551eef4a5225f126a281f8c051c2). --- linux-3.10.5-orig/sound/usb/mixer.c 2013-08-04 12:51:49.000000000 +0400 +++ linux-3.10.5/sound/usb/mixer.c 2013-08-06 21:12:42.274197287 +0400 @@ -888,6 +888,7 @@ case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384.

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[alsa-devel] [PATCH] ASoC: dt: Move WM8903 pin list from Tegra board binding to CODEC binding
by Mark Brown 08 Aug '13

08 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 24 ++-------------------- Documentation/devicetree/bindings/sound/wm8903.txt | 19 +++++++++++++++++ 2 files changed, 21 insertions(+), 22 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt index 3bf722d..4b44dfb 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt @@ -11,28 +11,8 @@ Required properties: - nvidia,audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources and - sinks are the WM8903's pins, and the jacks on the board: - - WM8903 pins: - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS - - Board connectors: + sinks are the WM8903's pins (documented in the WM8903 binding document), + and the jacks on the board: * Headphone Jack * Int Spk diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt index f102cbc..94ec32c 100644 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -28,6 +28,25 @@ Optional properties: performed. If any entry has the value 0xffffffff, that GPIO's configuration will not be modified. +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + Example: codec: wm8903@1a { -- 1.8.4.rc1

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[alsa-devel] [PATCH] ALSA: hda - Fix missing mute controls for CX5051
by Takashi Iwai 08 Aug '13

08 Aug '13

We've added a fake mute control (setting the amp volume to zero) for CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but this feature was overlooked in the generic parser implementation. Now the driver lacks of mute controls on these codecs. The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE bits in each place checking the amp capabilities. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001 Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbb..e3c7ba8 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, } #define nid_has_mute(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE) + check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) #define nid_has_volume(codec, nid, dir) \ check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) @@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, if (enable) val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; } - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (!enable) val |= HDA_AMP_MUTE; } @@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec, { unsigned int mask = 0xff; - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL)) mask &= ~0x80; } -- 1.8.3.1

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[alsa-devel] [PATCH 1/2] ASoC: ak4104: Add stub DAPM support
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it easer to integrate the device with other on-board components and ensures correct operation following removal of support for non-DAPM CODECs. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4104.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c7cfdf9..9a7c89b 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -51,6 +51,14 @@ struct ak4104_private { struct regmap *regmap; }; +static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ak4104_dapm_routes[] = { + { "TX", NULL, "Playback" }, +}; + static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { @@ -214,6 +222,11 @@ static int ak4104_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, + + .dapm_widgets = ak4104_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets), + .dapm_routes = ak4104_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes), }; static const struct regmap_config ak4104_regmap = { -- 1.8.4.rc1

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[alsa-devel] [PATCH] ASoC: dt: Move WM8903 pin list from Tegra board binding to CODEC binding
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 22 ++-------------------- Documentation/devicetree/bindings/sound/wm8903.txt | 19 +++++++++++++++++++ 2 files changed, 21 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt index 3bf722d..0efe58c 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt @@ -11,26 +11,8 @@ Required properties: - nvidia,audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources and - sinks are the WM8903's pins, and the jacks on the board: - - WM8903 pins: - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS + sinks are the WM8903's pins (documented in the WM8903 binding document), + and the jacks on the board: Board connectors: diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt index f102cbc..94ec32c 100644 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -28,6 +28,25 @@ Optional properties: performed. If any entry has the value 0xffffffff, that GPIO's configuration will not be modified. +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + Example: codec: wm8903@1a { -- 1.8.4.rc1

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[alsa-devel] [PATCH] ASoC: ads711x: Add DAPM support
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it easier to hook into boards and ensures the driver continues to work with support for non-DAPM CODECs removed. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ads117x.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 506d474..8f388ed 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -23,6 +23,28 @@ #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) +static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("Input1"), +SND_SOC_DAPM_INPUT("Input2"), +SND_SOC_DAPM_INPUT("Input3"), +SND_SOC_DAPM_INPUT("Input4"), +SND_SOC_DAPM_INPUT("Input5"), +SND_SOC_DAPM_INPUT("Input6"), +SND_SOC_DAPM_INPUT("Input7"), +SND_SOC_DAPM_INPUT("Input8"), +}; + +static const struct snd_soc_dapm_route ads117x_dapm_routes[] = { + { "Capture", NULL, "Input1" }, + { "Capture", NULL, "Input2" }, + { "Capture", NULL, "Input3" }, + { "Capture", NULL, "Input4" }, + { "Capture", NULL, "Input5" }, + { "Capture", NULL, "Input6" }, + { "Capture", NULL, "Input7" }, + { "Capture", NULL, "Input8" }, +}; + static struct snd_soc_dai_driver ads117x_dai = { /* ADC */ .name = "ads117x-hifi", @@ -34,7 +56,12 @@ static struct snd_soc_dai_driver ads117x_dai = { .formats = ADS117X_FORMATS,}, }; -static struct snd_soc_codec_driver soc_codec_dev_ads117x; +static struct snd_soc_codec_driver soc_codec_dev_ads117x = { + .dapm_widgets = ads117x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets), + .dapm_routes = ads117x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes), +}; static int ads117x_probe(struct platform_device *pdev) { -- 1.8.4.rc1

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