April 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ASoC: Blackfin: drop "-codec" from codec names
by Mike Frysinger 08 Apr '11

08 Apr '11

The recent multi-component patch incorrectly added "-codec" suffixes to parts which are not MFD. Drop the suffix from the machine drivers too. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ad1836.c | 2 +- sound/soc/blackfin/bf5xx-ad193x.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 +- sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 83012da..826bae6 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -85,7 +85,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai = { .cpu_dai_name = "bf5xx-tdm", .codec_dai_name = "ad1836-hifi", .platform_name = "bf5xx-tdm-pcm-audio", - .codec_name = "ad1836-codec.0", + .codec_name = "ad1836.0", .ops = &bf5xx_ad1836_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 3367882..5794d09 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -113,7 +113,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai = { .cpu_dai_name = "bf5xx-tdm", .codec_dai_name ="ad193x-hifi", .platform_name = "bf5xx-tdm-pcm-audio", - .codec_name = "ad193x-codec.5", + .codec_name = "ad193x.5", .ops = &bf5xx_ad193x_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index d57c9c9..c147130 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -73,7 +73,7 @@ static struct snd_soc_dai_link bf5xx_board_dai = { .cpu_dai_name = "bfin-ac97", .codec_dai_name = "ad1980-hifi", .platform_name = "bfin-pcm-audio", - .codec_name = "ad1980-codec", + .codec_name = "ad1980", .ops = &bf5xx_board_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 732fb8b..1655f95 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -186,7 +186,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-pcm-audio", - .codec_name = "ad73311-codec", + .codec_name = "ad73311", .ops = &bf5xx_ad73311_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index ad28663..732a02f 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-001b", + .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, }; -- 1.7.4.1

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[alsa-devel] [PATCH] snd_hda_codec_realtek: enable configurable support for ALC271
by Guido Trentalancia 07 Apr '11

07 Apr '11

This patch enables configurable support in the Realtek HD Audio driver for the ALC271 type commonly found on some Acer laptops. It is then possible to enable such support by passing the "model=acer" parameter at boot-time. Without such parameter the driver would fall back to BIOS auto-probing and in many cases that would result in the internal microphone not working. --- linux-2.6.38.2-original/sound/pci/hda/patch_realtek.c 2011-04-07 17:56:25.053046707 +0200 +++ linux-2.6.38.2/sound/pci/hda/patch_realtek.c 2011-04-07 19:03:00.975706888 +0200 @@ -15028,6 +15028,7 @@ static const char * const alc269_models[ [ALC269_DMIC] = "laptop-dmic", [ALC269_FUJITSU] = "fujitsu", [ALC269_LIFEBOOK] = "lifebook", + [ALC271_ACER] = "acer", [ALC269_AUTO] = "auto", };

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[alsa-devel] [PATCH] ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
by David Henningsson 07 Apr '11

07 Apr '11

Codec information available here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/752792/+attachme… -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH] ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
by Aaron Plattner 07 Apr '11

07 Apr '11

The MCP7x hardware computes the audio infoframe channel count automatically, but requires the audio driver to set the audio infoframe checksum manually via the Nv_VERB_SET_Info_Frame_Checksum control verb. When audio starts playing, nvhdmi_8ch_7x_pcm_prepare sets the checksum to (0x71 - chan - chanmask). For example, for 2ch audio, chan == 1 and chanmask == 0 so the checksum is set to 0x70. When audio playback finishes and the device is closed, nvhdmi_8ch_7x_pcm_close resets the channel formats, causing the channel count to revert to 8ch. Since the checksum is not reset, the hardware starts generating audio infoframes with invalid checksums. This causes some displays to blank the video. Fix this by updating the checksum and channel mask when the device is closed and also when it is first initialized. In addition, make sure that the channel mask is appropriate for an 8ch infoframe by setting it to 0x13 (FL FR LFE FC RL RR RLC RRC). Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> Acked-by: Stephen Warren <swarren(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 70 +++++++++++++++++++++++++++---------------- 1 files changed, 44 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 251773e..715615a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1280,6 +1280,39 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec, + int channels) +{ + unsigned int chanmask; + int chan = channels ? (channels - 1) : 1; + + switch (channels) { + default: + case 0: + case 2: + chanmask = 0x00; + break; + case 4: + chanmask = 0x08; + break; + case 6: + chanmask = 0x0b; + break; + case 8: + chanmask = 0x13; + break; + } + + /* Set the audio infoframe channel allocation and checksum fields. The + * channel count is computed implicitly by the hardware. */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Channel_Allocation, chanmask); + + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Info_Frame_Checksum, + (0x71 - chan - chanmask)); +} + static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1298,6 +1331,10 @@ static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, AC_VERB_SET_STREAM_FORMAT, 0); } + /* The audio hardware sends a channel count of 0x7 (8ch) when all the + * streams are disabled. */ + nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8); + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -1308,37 +1345,16 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { int chs; - unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id; + unsigned int dataDCC1, dataDCC2, channel_id; int i; mutex_lock(&codec->spdif_mutex); chs = substream->runtime->channels; - chan = chs ? (chs - 1) : 1; - switch (chs) { - default: - case 0: - case 2: - chanmask = 0x00; - break; - case 4: - chanmask = 0x08; - break; - case 6: - chanmask = 0x0b; - break; - case 8: - chanmask = 0x13; - break; - } dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT; dataDCC2 = 0x2; - /* set the Audio InforFrame Channel Allocation */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Channel_Allocation, chanmask); - /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) snd_hda_codec_write(codec, @@ -1413,10 +1429,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, } } - /* set the Audio Info Frame Checksum */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Info_Frame_Checksum, - (0x71 - chan - chanmask)); + nvhdmi_8ch_7x_set_info_frame_parameters(codec, chs); mutex_unlock(&codec->spdif_mutex); return 0; @@ -1512,6 +1525,11 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) spec->multiout.max_channels = 8; spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x; codec->patch_ops = nvhdmi_patch_ops_8ch_7x; + + /* Initialize the audio infoframe channel mask and checksum to something + * valid */ + nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8); + return 0; } -- 1.7.1

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[PATCH] ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
by David Henningsson 07 Apr '11

07 Apr '11

In cases where there is only one internal mic connected to ADC 0x11, alc275_setup_dual_adc won't handle the case, so we need to add the ADC node to the array of candidates. Cc: stable(a)kernel.org BugLink: http://bugs.launchpad.net/bugs/752792 Reported-by: Vincenzo Pii Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12c6f45..d9f1ef7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14124,7 +14124,7 @@ static hda_nid_t alc269vb_capsrc_nids[1] = { }; static hda_nid_t alc269_adc_candidates[] = { - 0x08, 0x09, 0x07, + 0x08, 0x09, 0x07, 0x11, }; #define alc269_modes alc260_modes -- 1.7.4.1 --------------040806060705030004040103--

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[alsa-devel] spi codec names in machine drivers
by Mike Frysinger 07 Apr '11

07 Apr '11

after the recent multi-component patch, spi codec names were changed to "<name>.<spics>" in the machine drivers. so if we have an ad193x codec on spi bus 0 cs 5, the name is now "ad193x.5". this also seems to be how fmt_single_name in soc-core.c is doing things. am i reading this right ? what if i have two codecs on two different spi busses but happen to have the same cs ? shouldnt the spi code do the same as i2c and include the bus in the id naming ? -mike

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[alsa-devel] [PATCH 0/3] Add WM8580 PCM Machine driver for Samsung SoCs
by Sangbeom Kim 07 Apr '11

07 Apr '11

Hi, This patchset is for WM8580 pcm machine driver that supports PCM audio on SMDKC110, SMDKV210, SMDK6450, SMDK6440. and It is tested on SMDKC110, SMDKV210, SMDKV6450. Based on these patches WM8580 pcm machine driver supports followings: o Playback supports 8kHz sampling rates. o Capture supports 8kHz sampling rates. This patchset contains followings o To Kukjin Kim and Ben Dooks, [PATCH 1/3] ARM: S5PV210: Add PCM audio support for WM8580 [PATCH 2/3] ARM: S5P64X0: Add PCM audio support for WM8580 o To Jassi Brar, Mark Brown and Liam Girdwood, [PATCH 3/3] ASoC: SAMSUNG: Add WM8580 PCM Machine driver This patch-set is based on 2.6.39-rc1 and WM8580 machine driver is originated by Seungwhan Youn. Best Regards, SB Kim (Sangbeom Kim)

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[alsa-devel] synchronized block of register writes
by Peter Hsiang 07 Apr '11

07 Apr '11

Hi Mark and Everyone, When a codec driver control handling function needs to write a group of registers without any other register of the same codec being written to in parallel by ALSA, what is the best way to do this under the snd-soc codec driver architecture? In the current architecture, does the driver's control handling function execute serially with other ALSA automated activities? For example, the codec has filter coefficient data registers in a secondary registers "segment" i.e. a portion of the register map is banked, and the secondary bank is purely for coefficients and they do not require caching. The coefficient registers share the same register numbers as those in the normal primary segment. To write the coefficients, one would first write to a register bit to select the desired segment, and then switch back when done. This means when writing the coefficients, I use the un-cached write. Thanks in advance for your sound advice. Peter

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[alsa-devel] [PATCH 1/2] ASoC: check channel mismatch between cpu_dai and codec_dai
by Lu Guanqun 07 Apr '11

07 Apr '11

Suppose we have: cpu_dai channels_min = 1 channels_max = 1 codec_dai channels_min = 2 channels_max = 2 This is a mismatch that should not happen, however according to the current code, the result of runtime->hw will be: channels_min = 2 channels_max = 1 We better spot it early. This patch checks this mismatch. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/soc/soc-core.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0cf6997..88bb206 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -640,7 +640,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->name, cpu_dai->name); goto config_err; } - if (!runtime->hw.channels_min || !runtime->hw.channels_max) { + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", codec_dai->name, cpu_dai->name); goto config_err;

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[alsa-devel] Dynamic control commands that match the downloadable DSP firmware
by Peter Hsiang 07 Apr '11

07 Apr '11

Hi Mark and Everyone, For codecs with a programmable DSP core, one needs to provide the amixer controls for the 'soft' features. As there could be multiple DSP firmware images for different applications, it would be nice if a single codec driver source can support an unlimited combination of commands for DSP firmware features. Rather than hard coding the control commands in the driver directly, wouldn't it be cool if, in addition to the firmware being loaded from file, the control commands information can be loaded from file as well. Then the driver would dynamically create and install the control commands the same way we already do for the EQ. What do you think? Peter

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