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

Re: [alsa-devel] [PATCH v2] ALSA: hdmi: expose ELD control
by Stephen Warren 27 Sep '11

27 Sep '11

Pierre-Louis Bossart wrote at Tuesday, September 27, 2011 9:49 AM: ... > > Set the access field to SNDRV_CTL_ELEM_ACCESS_READ. > > Also, unless we implement a notification, better to set > > SNDRV_CTL_ELEM_ACCESS_VOLATILE flag, too. > > The intent was to read the ELD after the device is opened, I don't think the > ELD can change during playback, the updates can only happen on a pin-sensing > event. I can't think of a good reason why a notification would be needed? If the user unplugs an HDMI monitor and plugs a different one in, and hence the X server turns off the output and re-enables it to cater for any required new display mode, would that force the audio client to close/re-open the audio device? I don't /think/ so. As such, I guess the ELD can change dynamically, so notification would be a good idea. -- nvpublic

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[alsa-devel] [PATCH 0/6] ASoC: Remove unused control_data field from private struct
by Axel Lin 27 Sep '11

27 Sep '11

This serial removes unused control_data field from private struct. The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Note: I don't have these hardware, only compile test on these patches. Axel Lin (6): ASoC: Remove unused "control_data" field of struct wm8904_priv ASoC: Remove unused "control_data" field of struct wm8940_priv ASoC: Remove unused "control_data" field of struct wm8960_priv ASoC: Remove unused "control_data" field of struct wm8978_priv ASoC: Remove unused "control_data" field of struct wm9081_priv ASoC: Remove unused "control_data" field of struct wm9090_priv sound/soc/codecs/wm8904.c | 2 -- sound/soc/codecs/wm8940.c | 3 --- sound/soc/codecs/wm8960.c | 3 --- sound/soc/codecs/wm8978.c | 3 --- sound/soc/codecs/wm9081.c | 3 --- sound/soc/codecs/wm9090.c | 3 --- 6 files changed, 0 insertions(+), 17 deletions(-) -- 1.7.4.1

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[alsa-devel] [PATCH 1/3] mfd: Add WM1811 support
by Mark Brown 27 Sep '11

27 Sep '11

The WM1811 is mostly register compatible with the WM8994 and WM8958, providing a high performance audio hub CODEC in a small form factor suitable for ultra compact system designs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Due to the ASoC cross dependencies it seems like it'll be easier to merge this via ASoC - Samuel, does this make sense to you? drivers/mfd/wm8994-core.c | 28 ++++++++++++++++++++++++++++ include/linux/mfd/wm8994/core.h | 1 + 2 files changed, 29 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 719fee8..61894fc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -167,6 +167,18 @@ static struct mfd_cell wm8994_devs[] = { * and should be handled via the standard regulator API supply * management. */ +static const char *wm1811_main_supplies[] = { + "DBVDD1", + "DBVDD2", + "DBVDD3", + "DCVDD", + "AVDD1", + "AVDD2", + "CPVDD", + "SPKVDD1", + "SPKVDD2", +}; + static const char *wm8994_main_supplies[] = { "DBVDD", "DCVDD", @@ -227,6 +239,7 @@ static int wm8994_suspend(struct device *dev) switch (wm8994->type) { case WM8958: + case WM1811: ret = wm8994_reg_read(wm8994, WM8958_MIC_DETECT_1); if (ret < 0) { dev_err(dev, "Failed to read power status: %d\n", ret); @@ -375,6 +388,9 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) } switch (wm8994->type) { + case WM1811: + wm8994->num_supplies = ARRAY_SIZE(wm1811_main_supplies); + break; case WM8994: wm8994->num_supplies = ARRAY_SIZE(wm8994_main_supplies); break; @@ -395,6 +411,10 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) } switch (wm8994->type) { + case WM1811: + for (i = 0; i < ARRAY_SIZE(wm1811_main_supplies); i++) + wm8994->supplies[i].supply = wm1811_main_supplies[i]; + break; case WM8994: for (i = 0; i < ARRAY_SIZE(wm8994_main_supplies); i++) wm8994->supplies[i].supply = wm8994_main_supplies[i]; @@ -428,6 +448,13 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) goto err_enable; } switch (ret) { + case 0x1811: + devname = "WM1811"; + if (wm8994->type != WM1811) + dev_warn(wm8994->dev, "Device registered as type %d\n", + wm8994->type); + wm8994->type = WM1811; + break; case 0x8994: devname = "WM8994"; if (wm8994->type != WM8994) @@ -592,6 +619,7 @@ static int wm8994_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id wm8994_i2c_id[] = { + { "wm1811", WM1811 }, { "wm8994", WM8994 }, { "wm8958", WM8958 }, { } diff --git a/include/linux/mfd/wm8994/core.h b/include/linux/mfd/wm8994/core.h index a275798..f44bdb7 100644 --- a/include/linux/mfd/wm8994/core.h +++ b/include/linux/mfd/wm8994/core.h @@ -20,6 +20,7 @@ enum wm8994_type { WM8994 = 0, WM8958 = 1, + WM1811 = 2, }; struct regulator_dev; -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Add device ID for WM9093 to WM9090 driver
by Mark Brown 27 Sep '11

27 Sep '11

The WM9093 is an enhanced version of the WM9093. Add the device ID to the driver, further patches will add support for the additional features in the WM9093. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9090.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4de1220..6112ddf 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -684,6 +684,7 @@ static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) static const struct i2c_device_id wm9090_id[] = { { "wm9090", 0 }, + { "wm9093", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, wm9090_id); -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Add DRC control for WM8996
by Mark Brown 27 Sep '11

27 Sep '11

From: Karl Tsou <karl(a)opensource.wolfsonmicro.com> Signed-off-by: Karl Tsou <karl(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 833df74..b98a8f8 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -641,6 +641,14 @@ SOC_DOUBLE_R("Speaker ZC Switch", WM8996_LEFT_PDM_SPEAKER, SOC_SINGLE("DSP1 EQ Switch", WM8996_DSP1_RX_EQ_GAINS_1, 0, 1, 0), SOC_SINGLE("DSP2 EQ Switch", WM8996_DSP2_RX_EQ_GAINS_1, 0, 1, 0), + +SOC_SINGLE("DSP1 DRC TXL Switch", WM8996_DSP1_DRC_1, 0, 1, 0), +SOC_SINGLE("DSP1 DRC TXR Switch", WM8996_DSP1_DRC_1, 1, 1, 0), +SOC_SINGLE("DSP1 DRC RX Switch", WM8996_DSP1_DRC_1, 2, 1, 0), + +SOC_SINGLE("DSP2 DRC TXL Switch", WM8996_DSP2_DRC_1, 0, 1, 0), +SOC_SINGLE("DSP2 DRC TXR Switch", WM8996_DSP2_DRC_1, 1, 1, 0), +SOC_SINGLE("DSP2 DRC RX Switch", WM8996_DSP2_DRC_1, 2, 1, 0), }; static const struct snd_kcontrol_new wm8996_eq_controls[] = { -- 1.7.6.3

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[alsa-devel] davinci-pcm debug
by David Pettit 27 Sep '11

27 Sep '11

I've written a new codec driver and tested it on an EVM for a TI Davinci ARM processor. It appeared to be working there. However, now I'm trying to debug buffer under-runs in my embedded system. I've inserted a debugging trace in the function davinci_pcm_pointer() in davinci-pdm.c to print out the value of asp_count. What I'm seeing is that when I first start playing an audio file, the value of asp_count increments very slowly. After some time, the value of asp_count start incrementing by the period size and I start seeing the debugging trace "davinci-pcm: audio_set_dma_params_play ...". Is this behavior expected? If I enable XRUN DEBUG and set the value to 5, then I occasionally see the message "PCM: hw_ptr skipping! (pos=17153, delta=23938, period=512, jdelta=0/49/0, hw_ptr=17279/17279)" where delta is always high. The comments in pcm_lib.c indicate that delta should be small or zero. What does a large value indicate? Thanks, David

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Re: [alsa-devel] [PATCH v2] ALSA: hdmi: expose ELD control
by Takashi Iwai 27 Sep '11

27 Sep '11

At Tue, 27 Sep 2011 10:48:39 -0500, Pierre-Louis Bossart wrote: > > Thanks for your review Takashi! > > > Wouldn't it be better to strip the buffer data and size in the error > > case? For example, radeon driver gives often size=0. Then it's > > translated as size=128 for a workaround, then again handled as an > > error by checking the version. > > Yes, I wasn't sure what to do here and expected some comments. Should the > strategy be that if there's a single error we discard the entire ELD by > setting the size to zero? > No issues for the other changes. It's a good question. I'm inclined to discard all for any errors, in the case of ELD. AFAIK, most of errors are really broken ELD, and no need to care the rest of bits. > > Set the access field to SNDRV_CTL_ELEM_ACCESS_READ. > > Also, unless we implement a notification, better to set > > SNDRV_CTL_ELEM_ACCESS_VOLATILE flag, too. > > The intent was to read the ELD after the device is opened, I don't think the > ELD can change during playback, the updates can only happen on a pin-sensing > event. I can't think of a good reason why a notification would be needed? It's just a matter of definition. If the data may be changed without notification, the element should set VOLATILE flag. You don't have to implement the notification mechanism if you don't want :) thanks, Takashi

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[alsa-devel] [PATCH 1/3] ASoC: Allow DAI formats to be specified in the dai_link
by Mark Brown 27 Sep '11

27 Sep '11

For almost all machines the DAI format is a constant, always set to the same thing. This means that not only should we normally set it on init rather than in hw_params() (where it has been for historical reasons) we should also allow users to configure this by setting a variable in the dai_link structure. The combination of these two will make many machine drivers even more data driven. Implement a new dai_fmt field in the dai_link doing just that. Since 0 is a valid value for many format flags and we need to be able to tell if the field is actually set also add one to all the values used to configure formats. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 34 +++++++++++++++++----------------- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 22 ++++++++++++++++++++++ 3 files changed, 41 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 12d98b4..2413acc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -24,13 +24,13 @@ struct snd_pcm_substream; * Describes the physical PCM data formating and clocking. Add new formats * to the end. */ -#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ -#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ -#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */ -#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */ -#define SND_SOC_DAIFMT_AC97 5 /* AC97 */ -#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */ +#define SND_SOC_DAIFMT_I2S 1 /* I2S mode */ +#define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */ +#define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */ +#define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */ +#define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */ +#define SND_SOC_DAIFMT_AC97 6 /* AC97 */ +#define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */ /* left and right justified also known as MSB and LSB respectively */ #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J @@ -42,8 +42,8 @@ struct snd_pcm_substream; * DAI bit clocks can be be gated (disabled) when the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ -#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ -#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ +#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ +#define SND_SOC_DAIFMT_GATED (2 << 4) /* clock is gated */ /* * DAI hardware signal inversions. @@ -51,10 +51,10 @@ struct snd_pcm_substream; * Specifies whether the DAI can also support inverted clocks for the specified * format. */ -#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ -#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */ -#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ -#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */ +#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */ +#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ +#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ +#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */ /* * DAI hardware clock masters. @@ -63,10 +63,10 @@ struct snd_pcm_substream; * i.e. if the codec is clk and FRM master then the interface is * clk and frame slave. */ -#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ -#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ -#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ +#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */ +#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ +#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ +#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */ #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 diff --git a/include/sound/soc.h b/include/sound/soc.h index b499b37..a4dc699 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -713,6 +713,8 @@ struct snd_soc_dai_link { const char *cpu_dai_name; const char *codec_dai_name; + unsigned int dai_fmt; /* format to set on init */ + /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bd20154..b0cec3c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1317,6 +1317,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; + struct snd_soc_dai_link *dai_link; int ret, i, order; mutex_lock(&card->mutex); @@ -1429,6 +1430,27 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + for (i = 0; i < card->num_links; i++) { + dai_link = &card->dai_link[i]; + + if (dai_link->dai_fmt) { + pr_crit("%d %p\n", i, card->rtd[i].codec_dai); + ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].codec_dai->dev, + "Failed to set DAI format: %d\n", + ret); + + ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].cpu_dai->dev, + "Failed to set DAI format: %d\n", + ret); + } + } + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), -- 1.7.6.3

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[alsa-devel] [PATCH v2] ALSA: hdmi: expose ELD control
by Pierre-Louis Bossart 27 Sep '11

27 Sep '11

Applications may want to read ELD information to understand what codecs are supported on the HDMI receiver and handle the a-v delay for better lip-sync. ELD information is exposed in a device-specific IFACE_PCM kcontrol. Tested both with amixer and PulseAudio. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 9 ++-- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 88 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 94 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index c34f730..9bf0d8b 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -327,14 +327,14 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE || size > PAGE_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* update info */ + eld->eld_size = size; + buf = eld->eld_buffer; for (i = 0; i < size; i++) { unsigned int val = hdmi_get_eld_data(codec, nid, i); @@ -356,7 +356,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, ret = hdmi_update_eld(eld, buf, size); error: - kfree(buf); return ret; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aaefa7c..04d730f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -621,6 +621,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -645,6 +646,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3f1f6ac..944e0cd 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,86 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + + spec = codec->spec; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + + pin_idx = kcontrol->private_value; + snd_BUG_ON(pin_idx < 0); + + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + uinfo->count = eld->eld_size; + + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + int size; + + spec = codec->spec; + pin_idx = kcontrol->private_value; + + snd_BUG_ON(pin_idx < 0); + + memset(ucontrol->value.bytes.data, 0, ELD_MAX_SIZE); + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + size = eld->eld_size; + + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, size); + + /* reset rest of control */ + memset(&ucontrol->value.bytes.data[size], 0, ELD_MAX_SIZE-size); + + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, + int device) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int err; + + if (pin_idx < 0) + return -EINVAL; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = pin_idx; + kctl->id.device = device; + + err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl); + if (err < 0) + return err; + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -1193,6 +1273,14 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pin_idx); + + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, + pin_idx, + spec->pcm_rec[pin_idx].device); + + if (err < 0) + return err; } return 0; -- 1.7.6.2

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 27 Sep '11

27 Sep '11

The following changes since commit 0a742681e6072a71f30cfe6312f758f1cd185c21: ASoC: Add missed free_irq in wm5100_remove and wm5100_probe error path (2011-09-23 11:54:24 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (4): ASoC: Drop exporting sn95031_get_mic_bias ASoC: Drop exporting ad1980_dai ASoC: Remove unused fields in struct mfld_mc_private ASoC: Staticise sst_platform_dai Lars-Peter Clausen (3): ASoC: ssm2602: Re-enable oscillator after suspend ASoC: ssm2602: Set initial bias level to standby ASoC: ssm2602: Support setting the oscillator and the clock output state Mark Brown (9): ASoC: Support a wider range of sample rates on Speyside WM8962 ASoC: Rename WM8962 DMIC widget to DMIC_ENA ASoC: Convert WM8962 MICBIAS to a supply widget ASoC: Add support for on-board analogue microphones on Speyside WM8962 ASoC: Add DMIC control to Speyside WM8962 board ASoC: Don't force bias on ground referenced devices ASoC: Add platform data for WM1250 EV1 GPIOs ASoC: Set idle_bias_off for WM1250 EV1 Merge branch 'for-3.1' into for-3.2 Peter Ujfalusi (11): ASoC: omap-mcbsp: Fix compile time warning about ambiguous ‘else’ ASoC: twl6040: Read the TRIM values from the chip ASoC: twl6040: Function to fetch the TRIM values ASoC: omap-mcpdm: API to configure offset cancellation ASoC: sdp4430: Configure McPDM offset cancellation ASoC: twl6040: Rename pga_event to out_drv_event ASoC: twl6040: Combine the custom volsw get, and put functions ASoC: twl6040: Move delayed_work struct inside twl6040_output for HS/HF ASoC: twl6040: Move the delayed_work for HS detection under twl6040_jack_data ASoC: twl6040: No need to change delay during HS ramp ASoC: twl6040: No need to change delay during HF ramp include/sound/wm1250-ev1.h | 27 ++++++ sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/sn95031.c | 1 - sound/soc/codecs/ssm2602.c | 73 +++++++++++++---- sound/soc/codecs/ssm2602.h | 6 +- sound/soc/codecs/twl6040.c | 155 ++++++++++++----------------------- sound/soc/codecs/twl6040.h | 13 +++ sound/soc/codecs/wm1250-ev1.c | 122 +++++++++++++++++++++++++++- sound/soc/codecs/wm8962.c | 10 +- sound/soc/mid-x86/mfld_machine.c | 2 - sound/soc/mid-x86/sst_platform.c | 2 +- sound/soc/omap/omap-mcbsp.c | 3 +- sound/soc/omap/omap-mcpdm.c | 25 ++++++ sound/soc/omap/omap-mcpdm.h | 12 +++ sound/soc/omap/sdp4430.c | 10 ++- sound/soc/samsung/speyside_wm8962.c | 24 ++++-- sound/soc/soc-dapm.c | 7 +- 17 files changed, 350 insertions(+), 143 deletions(-) create mode 100644 include/sound/wm1250-ev1.h

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