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

[alsa-devel] [PATCH] ASoC: Fix return value of wm8903_gpio_direction_in() and wm8903_gpio_direction_out()
by Axel Lin 02 Jan '12

02 Jan '12

We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8903.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d88b727..c91fb2f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1777,13 +1777,18 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK; val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | WM8903_GP1_DIR; - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) @@ -1803,13 +1808,18 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK; val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | (value << WM8903_GP2_LVL_SHIFT); - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) -- 1.7.5.4

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[alsa-devel] [PATCH 1/4] ASoC: Convert saarb to table based DAPM init
by Axel Lin 02 Jan '12

02 Jan '12

Also remove a unused ret variable to silence the build warning. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/saarb.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index b2be225..c34146b 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -51,7 +51,7 @@ static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { }; /* saarb machine audio map */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route saarb_audio_map[] = { {"Headset Stereophone", NULL, "HS1"}, {"Headset Stereophone", NULL, "HS2"}, @@ -121,17 +121,17 @@ static struct snd_soc_card snd_soc_card_saarb = { .owner = THIS_MODULE, .dai_link = saarb_dai, .num_links = ARRAY_SIZE(saarb_dai), + + .dapm_widgets = saarb_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), + .dapm_routes = saarb_audio_map, + .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), }; static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - snd_soc_dapm_new_controls(dapm, saarb_dapm_widgets, - ARRAY_SIZE(saarb_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); -- 1.7.5.4

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[alsa-devel] [PATCH 1/4] ASoC: Use dai_fmt in saarb machine driver
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/saarb.c | 11 ++--------- 1 files changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index 2e21712..b2be225 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -92,15 +92,6 @@ static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); return ret; @@ -119,6 +110,8 @@ static struct snd_soc_dai_link saarb_dai[] = { .platform_name = "pxa-pcm-audio", .codec_name = "88pm860x-codec", .init = saarb_pm860x_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &saarb_i2s_ops, }, }; -- 1.7.5.4

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[alsa-devel] [PATCH 1/3] ASoC: Fix return value of ak4641_pcm_set_dai_fmt()
by Axel Lin 02 Jan '12

02 Jan '12

We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4641.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 266ebea..c4d165a 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -339,6 +339,7 @@ static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u8 btif; + int ret; /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -358,7 +359,11 @@ static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); + ret = snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); + if (ret < 0) + return ret; + + return 0; } static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, -- 1.7.5.4

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[alsa-devel] [PATCH 01/14] ASoC: Convert alc5623 to devm_kzalloc()
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 11 +++-------- 1 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 6a9b621..3feee56 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1022,7 +1022,8 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); - alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL); + alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), + GFP_KERNEL); if (alc5623 == NULL) return -ENOMEM; @@ -1044,7 +1045,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, alc5623_dai.name = "alc5623-hifi"; break; default: - kfree(alc5623); return -EINVAL; } @@ -1053,20 +1053,15 @@ static int alc5623_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); - if (ret != 0) { + if (ret != 0) dev_err(&client->dev, "Failed to register codec: %d\n", ret); - kfree(alc5623); - } return ret; } static int alc5623_i2c_remove(struct i2c_client *client) { - struct alc5623_priv *alc5623 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - kfree(alc5623); return 0; } -- 1.7.5.4

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[alsa-devel] [PATCH v8 3/3 RESEND] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 02 Jan '12

02 Jan '12

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for AUX1, AUX2 and INPGA (Moved here from DAPM patch) Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 77 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0ebcbd5..ff084a6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -421,26 +447,42 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; @@ -451,14 +493,23 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], @@ -514,12 +565,21 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, @@ -528,9 +588,20 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"ADC Right", NULL, "INPGA Right"}, /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, @@ -887,6 +958,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] Period time/size in adaptive USB - alignment to nrpacks?
by Pavel Hofman 02 Jan '12

02 Jan '12

Hi, I have been analyzing USB URBs in wireshark. Output stream, adaptive mode, 16 bits, 48kHz, stereo. Each URB is issued always at nrpacks milliseconds. Each URB transfers 48 samples * 2 bytes * 2 channels * nrpacks milliseconds of bytes. The last URB of period keeps the same pace (nrpacks milliseconds), however holds only the remaining number of bytes. This creates ripples in the data stream bitrate, possibly leading to increased jitter of the PLL'd clock in the USB receiver. When period size is aligned to multiples of samplerate/1000 * audio frame bytes * nrpacks, all URBs carry constant number of bytes. I am wondering if for the xx.000 Hz samplerates it made sense to offer only these aligned multiples in snd_pcm_hw_params_set_period_time_near. Thank you for your expert opinion. Best regards, Pavel.

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 01 Jan '12

01 Jan '12

The following changes since commit c5cf4dbc7f804bb4ff02a065b927bd8688204253: ASoC: Add trivial pm_runtime usage to Samsung DAI drivers (2011-12-23 12:08:32 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 Axel Lin (8): ASoC: Convert da7210 to devm_kzalloc() ASoC: Convert lm4857 to devm_kzalloc() ASoC: Convert uda1380 to devm_kzalloc() ASoC: Convert 88pm860x-codec to devm_kzalloc() ASoC: Fix build error in sound/soc/kirkwood/kirkwood-i2s.c ASoC: Use dai_fmt in kirkwood-openrd machine driver ASoC: Use dai_fmt in kirkwood-t5325 machine driver ASoC: Convert kirkwood-t5325 to table based DAPM init Joachim Eastwood (1): ASoC: wm8776: add missing break in sample size switch Mark Brown (1): Merge branch 'for-3.2' into for-3.3 sound/soc/codecs/88pm860x-codec.c | 7 ++----- sound/soc/codecs/da7210.c | 7 ++----- sound/soc/codecs/lm4857.c | 13 ++----------- sound/soc/codecs/uda1380.c | 6 ++---- sound/soc/codecs/wm8776.c | 1 + sound/soc/kirkwood/kirkwood-i2s.c | 3 +-- sound/soc/kirkwood/kirkwood-openrd.c | 14 ++------------ sound/soc/kirkwood/kirkwood-t5325.c | 24 +++++++----------------- sound/soc/kirkwood/kirkwood.h | 1 - 9 files changed, 19 insertions(+), 57 deletions(-)

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[alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 6
by Daniel Mack 31 Dec '11

31 Dec '11

Fortunately, the effects reported by Aurélien were easy to fix, and indeed just regressions from previous version of the patch set. Those are now out of the way, so here's version 6. One thing that has also been added here is a sixth patch that adds some documentation to endpoint.c to explain the concept of the new implementation. Thanks, and please have more test results coming. Daniel Daniel Mack (6): ALSA: snd-usb: add snd_usb_audio-wide mutex ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: add support for implicit feedback ALSA: snd-usb: add some documentation sound/usb/card.c | 10 +- sound/usb/card.h | 62 ++ sound/usb/endpoint.c | 1602 ++++++++++++++++++++++++++++---------------------- sound/usb/endpoint.h | 32 +- sound/usb/pcm.c | 440 ++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 2 + 7 files changed, 1381 insertions(+), 798 deletions(-) -- 1.7.7.4

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Re: [alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 6
by Grant Diffey 31 Dec '11

31 Dec '11

> Yes, any kind of USB audio device is good. > > Thanks for heling test! :) > > > So sadly not all is well in my world the logitech is only registering as a mono device for some reason. nevyn@cetacea:~$ speaker-test -c 2 -D hw:Headset speaker-test 1.0.24.2 Playback device is hw:Headset Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Channels count (2) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument it's clearly stereo. lsusb -vv attached for reference h

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