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[alsa-devel] [PATCH v3 0/2] usb: Add native DSD support for Denon/Marantz
by Jurgen Kramer 28 Nov '14

28 Nov '14

This patchset adds native DSD support for Denon/Marantz devices. A specific vendor command is needed to switch between PCM and DSD mode. This is implemented as a new quirk function. Changes from v2: - switch) -> switch ( - combine PCM and DSD mode select to a single USB ctrl msg Changes from v1: - fix compare in err check - remove unneeded brackets Jurgen Kramer (2): usb: Add native DSD support for Denon/Marantz DACs usb: Add mode select quirk for Denon/Marantz DACs sound/usb/pcm.c | 5 +++++ sound/usb/quirks.c | 47 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 3 +++ 3 files changed, 55 insertions(+) -- 1.9.3

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[alsa-devel] [PATCH v2 0/2] usb: add native DSD support for Denon/Marantz
by Jurgen Kramer 28 Nov '14

28 Nov '14

This patchset adds native DSD support for Denon/Marantz devices. A specific vendor command is needed to switch between PCM and DSD mode. This is implemented as a new quirk function. Changes from v1: - fix compare in err check - remove unneeded brackets Jurgen Kramer (2): usb: Add native DSD support for Denon/Marantz DACs usb: Add mode select quirk for Denon/Marantz DACs sound/usb/pcm.c | 5 +++++ sound/usb/quirks.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 3 +++ 3 files changed, 61 insertions(+) -- 1.9.3

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[alsa-devel] [PATCH] ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell
by Jie Yang 28 Nov '14

28 Nov '14

Move capture PCM pin from PCM4 to PCM0 for Broadwell. This will allow us to integrate with pulseaudio better for usually default device is set to 0. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 15 ++------------- sound/soc/intel/sst-haswell-pcm.c | 34 ++++++++++++++++------------------ 2 files changed, 18 insertions(+), 31 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 52cb764..c256764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { /* Front End DAI links */ { .name = "System PCM", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -180,6 +180,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .init = broadwell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -214,18 +215,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture PCM", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - /* Back End DAI links */ { /* SSP0 - Codec */ diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 5f70d81..0180b38 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -309,7 +309,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -396,8 +396,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, /* DSP stream type depends on DAI ID */ switch (rtd->cpu_dai->id) { case 0: - stream_type = SST_HSW_STREAM_TYPE_SYSTEM; - module_id = SST_HSW_MODULE_PCM_SYSTEM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + stream_type = SST_HSW_STREAM_TYPE_SYSTEM; + module_id = SST_HSW_MODULE_PCM_SYSTEM; + } + else { + stream_type = SST_HSW_STREAM_TYPE_CAPTURE; + module_id = SST_HSW_MODULE_PCM_CAPTURE; + } break; case 1: case 2: @@ -410,10 +416,6 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, path_id = SST_HSW_STREAM_PATH_SSP0_OUT; module_id = SST_HSW_MODULE_PCM_REFERENCE; break; - case 4: - stream_type = SST_HSW_STREAM_TYPE_CAPTURE; - module_id = SST_HSW_MODULE_PCM_CAPTURE; - break; default: dev_err(rtd->dev, "error: invalid DAI ID %d\n", rtd->cpu_dai->id); @@ -779,6 +781,13 @@ static struct snd_soc_dai_driver hsw_dais[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, + }, }, { /* PCM */ @@ -815,17 +824,6 @@ static struct snd_soc_dai_driver hsw_dais[] = { .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, - { - .name = "Capture Pin", - .id = HSW_PCM_DAI_ID_CAPTURE, - .capture = { - .stream_name = "Analog Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, - }, - }, }; static const struct snd_soc_dapm_widget widgets[] = { -- 1.9.1

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[alsa-devel] [PATCH 0/2] usb: add native DSD support for Denon/Marantz
by Jurgen Kramer 28 Nov '14

28 Nov '14

This patchset adds native DSD support for Denon/Marantz devices. A specific vendor command is needed to switch between PCM and DSD mode. This is implemented as a new quirk function. Jurgen Kramer (2): Add native DSD support for Denon/Marantz DACs Add mode select quirk for Denon/Marantz DACs sound/usb/pcm.c | 6 ++++++ sound/usb/quirks.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 3 +++ 3 files changed, 62 insertions(+) -- 1.9.3

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[alsa-devel] [PATCH] ASoC: Intel: Correct the xmax volume
by Jie Yang 28 Nov '14

28 Nov '14

The xmax volume should be corrected to ARRAY_SIZE(volume_map)-1, otherwise, the xmax value will be mapped to 0 wrongly. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 20561ca..5f70d81 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -298,19 +298,19 @@ static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1); static const struct snd_kcontrol_new hsw_volume_controls[] = { /* Global DSP volume */ SOC_DOUBLE_EXT_TLV("Master Playback Volume", 0, 0, 8, - ARRAY_SIZE(volume_map) -1, 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_volume_get, hsw_volume_put, hsw_vol_tlv), /* Offload 0 volume */ SOC_DOUBLE_EXT_TLV("Media0 Playback Volume", 1, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Offload 1 volume */ SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; -- 1.9.1

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[alsa-devel] [PATCH 1/3] ASoC: mxs-sgtl5000: Remove MCLK restriction
by Fabio Estevam 27 Nov '14

27 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> According to the sgtl5000 datasheet the MCLK frequency range restriction of 8 to 27 MHz only applies when the PLL is used - synchronous SYS_MCLK input mode. mxs-sgtl5000 machine sets the codec as slave, and mx28 generates MCLK in the range of 256*fs, 384*fs or 512*fs, which is called asynchronous SYS_MCLK input. In asynchronous SYS_MCLK we cannot have the 8 to 27 MHz check because if we want to play a 8KHz sample rate track, with a MCLK of 8k * 512 = 4.096MHz the current check would return -EINVAL, which is not correct. Remove the 8 to 27MHz frequency check, since this only applies to the synchronous SYS_MCLK input case. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 7515cd4..6f1916b 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -49,13 +49,6 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, break; } - /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ - if (mclk < 8000000 || mclk > 27000000) { - dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n", - mclk / 1000000, mclk / 1000 % 1000); - return -EINVAL; - } - /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */ ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0); if (ret) { -- 1.9.1

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[alsa-devel] [PATCH 01/11] ASoC: cq93vc: Remove unused state struct
by Lars-Peter Clausen 27 Nov '14

27 Nov '14

While two of the fields in the cq93vc driver state struct are initialized none of them are ever acutally read again. So remove the whole struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/mfd/davinci_voicecodec.h | 7 ------- sound/soc/codecs/cq93vc.c | 8 -------- 2 files changed, 15 deletions(-) diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h index cb01496..8e1cdbe 100644 --- a/include/linux/mfd/davinci_voicecodec.h +++ b/include/linux/mfd/davinci_voicecodec.h @@ -99,12 +99,6 @@ struct davinci_vcif { dma_addr_t dma_rx_addr; }; -struct cq93vc { - struct platform_device *pdev; - struct snd_soc_codec *codec; - u32 sysclk; -}; - struct davinci_vc; struct davinci_vc { @@ -122,7 +116,6 @@ struct davinci_vc { /* Client devices */ struct davinci_vcif davinci_vcif; - struct cq93vc cq93vc; }; #endif diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 537327c..036a877 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -62,14 +62,10 @@ static int cq93vc_mute(struct snd_soc_dai *dai, int mute) static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct davinci_vc *davinci_vc = codec->dev->platform_data; - switch (freq) { case 22579200: case 27000000: case 33868800: - davinci_vc->cq93vc.sysclk = freq; return 0; } @@ -135,10 +131,6 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; - - davinci_vc->cq93vc.codec = codec; - /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- 1.8.0

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[alsa-devel] [PATCH] ASoC: Intel: Remove useless loopback volume control for Broadwell
by Jie Yang 27 Nov '14

27 Nov '14

On Broadwell, the ADSP FW don't support loopback record volume tuning, so here remove this control. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 331237f..20561ca 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -308,10 +308,6 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8, ARRAY_SIZE(volume_map), 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), - /* Loopback volume */ - SOC_DOUBLE_EXT_TLV("Loopback Capture Volume", 3, 0, 8, - ARRAY_SIZE(volume_map), 0, - hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, ARRAY_SIZE(volume_map), 0, -- 1.9.1

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[alsa-devel] [patch] ASoC: sigmadsp: uninitialized variable in sigmadsp_activate_ctrl()
by Dan Carpenter 27 Nov '14

27 Nov '14

The "changed" variable should be set to false at the start. Fixes: a35daac77a03 ('ASoC: sigmadsp: Add support for fw v2') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 34fdc40..d53680a 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -661,7 +661,8 @@ static void sigmadsp_activate_ctrl(struct sigmadsp *sigmadsp, struct snd_card *card = sigmadsp->component->card->snd_card; struct snd_kcontrol_volatile *vd; struct snd_ctl_elem_id id; - bool active, changed; + bool active; + bool changed = false; active = sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask);

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[alsa-devel] [PATCH] ASoC: wm8960: Move register initialisation to I2C driver probe()
by Zidan Wang 27 Nov '14

27 Nov '14

We must ensure that the clocking configuration is valid as rapidly as possible. And do software reset before the others registers updates, or the registers will be reset to the default state. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 41 ++++++++++++++++++++--------------------- 1 file changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index bc8793cd..031a1ae 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -128,7 +128,7 @@ struct wm8960_priv { struct wm8960_data pdata; }; -#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) +#define wm8960_reset(c) regmap_write(c, WM8960_RESET, 0) /* enumerated controls */ static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted", @@ -947,31 +947,12 @@ static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct wm8960_data *pdata = &wm8960->pdata; - int ret; if (pdata->capless) wm8960->set_bias_level = wm8960_set_bias_level_capless; else wm8960->set_bias_level = wm8960_set_bias_level_out3; - ret = wm8960_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - - /* Latch the update bits */ - snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LADC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RADC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LDAC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RDAC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LOUT1, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_ROUT1, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100); - snd_soc_add_codec_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); wm8960_add_widgets(codec); @@ -1030,7 +1011,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, else if (i2c->dev.of_node) wm8960_set_pdata_from_of(i2c, &wm8960->pdata); - if (pdata && pdata->shared_lrclk) { + ret = wm8960_reset(wm8960->regmap); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + return ret; + } + + if (wm8960->pdata.shared_lrclk) { ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 0x4, 0x4); if (ret != 0) { @@ -1040,6 +1027,18 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, } } + /* Latch the update bits */ + regmap_update_bits(wm8960->regmap, WM8960_LINVOL, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RINVOL, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LADC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RADC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LDAC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RDAC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LOUT1, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_ROUT1, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LOUT2, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_ROUT2, 0x100, 0x100); + i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- 1.9.1

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[alsa-devel] [patch] ALSA: hda - using uninitialized data
by Dan Carpenter 27 Nov '14

27 Nov '14

In olden times the snd_hda_param_read() function always set "*start_id" but in 2007 we introduced a new return and it causes uninitialized data bugs in a couple of the callers: print_codec_info() and hdmi_parse_codec(). Fixes: e8a7f136f5ed ('[ALSA] hda-intel - Improve HD-audio codec probing robustness') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This is from static analysis. Untested. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b2d5899..2fe86d2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -346,8 +346,10 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT); - if (parm == -1) + if (parm == -1) { + *start_id = 0; return 0; + } *start_id = (parm >> 16) & 0x7fff; return (int)(parm & 0x7fff); }

2 1

[alsa-devel] [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 27 Nov '14

27 Nov '14

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This set of patches adds support for audio on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This encapsulates the MI2S port, which is what these drivers are configured to use. The I2S pins are connected to an external DAC/amp chip. In addition, a single GPIO is connected to the same DAC/amp, which gives the SOC enable/disable control. The specific drivers added are: - a machine driver that handles the board-specific pins - a native driver that handles hardware access to the LPAIF - a CPU DAI driver for controlling the LPAIF block - a PCM MI2S platform driver Corresponding additions to the device tree for the ipq806x and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. - Ken Kenneth Westfield (9): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Add device tree binding docs ASoC: ipq806x: add native LPAIF driver ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add I2S PCM platform driver ASoC: ipq806x: Add machine driver for IPQ806X SOC ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../bindings/sound/qcom,ipq806x-snd-card.txt | 41 ++ .../bindings/sound/qcom,lpass-cpu-dai.txt | 20 + .../devicetree/bindings/sound/qcom,lpass-lpaif.txt | 21 + .../bindings/sound/qcom,lpass-pcm-mi2s.txt | 12 + MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 33 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/qcom/Kconfig | 43 ++ sound/soc/qcom/Makefile | 11 + sound/soc/qcom/ipq806x.c | 221 ++++++++++ sound/soc/qcom/lpass-cpu-dai.c | 307 +++++++++++++ sound/soc/qcom/lpass-lpaif.c | 488 +++++++++++++++++++++ sound/soc/qcom/lpass-lpaif.h | 181 ++++++++ sound/soc/qcom/lpass-pcm-mi2s.c | 390 ++++++++++++++++ sound/soc/qcom/lpass-pcm-mi2s.h | 40 ++ 16 files changed, 1817 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,ipq806x-snd-card.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu-dai.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-lpaif.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-pcm-mi2s.txt create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/ipq806x.c create mode 100644 sound/soc/qcom/lpass-cpu-dai.c create mode 100644 sound/soc/qcom/lpass-lpaif.c create mode 100644 sound/soc/qcom/lpass-lpaif.h create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.c create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.h -- 1.8.2.1

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[alsa-devel] [PATCH v2] ALSA: korg1212: cleanup of printk
by Sudip Mukherjee 26 Nov '14

26 Nov '14

From: Sudip Mukherjee <sudip(a)vectorindia.org> replaced all references of the debug messages via printk with dev_* macro (mostly dev_dbg). one reference was changed to pr_err as there the card might have been uninitialized. this patch will generate warning from checkpatch about broken quoted strings. but that was not fixed intentionally to improve the readability. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v2: the build warnings of v1 are fixed. hi Takashi, in your review of v1, you said about some lines which are not ending with \n. but i was not able to find them. did i miss them somewhere? sound/pci/korg1212/korg1212.c | 407 +++++++++++++++++++++++------------------- 1 file changed, 220 insertions(+), 187 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 59d21c9..faebe81 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -39,21 +39,6 @@ #include <asm/io.h> // ---------------------------------------------------------------------------- -// Debug Stuff -// ---------------------------------------------------------------------------- -#define K1212_DEBUG_LEVEL 0 -#if K1212_DEBUG_LEVEL > 0 -#define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) -#else -#define K1212_DEBUG_PRINTK(fmt,...) -#endif -#if K1212_DEBUG_LEVEL > 1 -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args) -#else -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,...) -#endif - -// ---------------------------------------------------------------------------- // Record/Play Buffer Allocation Method. If K1212_LARGEALLOC is defined all // buffers are alocated as a large piece inside KorgSharedBuffer. // ---------------------------------------------------------------------------- @@ -530,12 +515,12 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, int rc = K1212_CMDRET_Success; if (!korg1212->outDoorbellPtr) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: CardUninitialized\n"); + pr_err("K1212_DEBUG: CardUninitialized\n"); return K1212_CMDRET_CardUninitialized; } - K1212_DEBUG_PRINTK("K1212_DEBUG: Card <- 0x%08x 0x%08x [%s]\n", - doorbellVal, mailBox0Val, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Card <- 0x%08x 0x%08x [%s]\n", + doorbellVal, mailBox0Val, stateName[korg1212->cardState]); for (retryCount = 0; retryCount < MAX_COMMAND_RETRIES; retryCount++) { writel(mailBox3Val, korg1212->mailbox3Ptr); writel(mailBox2Val, korg1212->mailbox2Ptr); @@ -562,7 +547,8 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, mailBox3Lo = readl(korg1212->mailbox3Ptr); if (mailBox3Lo & COMMAND_ACK_MASK) { if ((mailBox3Lo & DOORBELL_VAL_MASK) == (doorbellVal & DOORBELL_VAL_MASK)) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- Success\n"); + dev_dbg(korg1212->card->dev, + "Card <- Success\n"); rc = K1212_CMDRET_Success; break; } @@ -571,7 +557,7 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, korg1212->cmdRetryCount += retryCount; if (retryCount >= MAX_COMMAND_RETRIES) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- NoAckFromCard\n"); + dev_dbg(korg1212->card->dev, "Card <- NoAckFromCard\n"); rc = K1212_CMDRET_NoAckFromCard; } @@ -612,20 +598,21 @@ static void snd_korg1212_timer_func(unsigned long data) korg1212->stop_pending_cnt = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Stop ack'ed [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Stop ack'ed [%s]\n", + stateName[korg1212->cardState]); } else { if (--korg1212->stop_pending_cnt > 0) { /* reprogram timer */ korg1212->timer.expires = jiffies + 1; add_timer(&korg1212->timer); } else { - snd_printd("korg1212_timer_func timeout\n"); + dev_dbg(korg1212->card->dev, + "korg1212_timer_func timeout\n"); korg1212->sharedBufferPtr->cardCommand = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); - K1212_DEBUG_PRINTK("K1212_DEBUG: Stop timeout [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Stop timeout [%s]\n", + stateName[korg1212->cardState]); } } spin_unlock_irqrestore(&korg1212->lock, flags); @@ -660,8 +647,8 @@ static inline void snd_korg1212_setCardState(struct snd_korg1212 * korg1212, enu static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: OpenCard [%s] %d\n", - stateName[korg1212->cardState], korg1212->opencnt); + dev_dbg(korg1212->card->dev, "OpenCard [%s] %d\n", + stateName[korg1212->cardState], korg1212->opencnt); mutex_lock(&korg1212->open_mutex); if (korg1212->opencnt++ == 0) { snd_korg1212_TurnOffIdleMonitor(korg1212); @@ -674,8 +661,8 @@ static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard [%s] %d\n", - stateName[korg1212->cardState], korg1212->opencnt); + dev_dbg(korg1212->card->dev, "CloseCard [%s] %d\n", + stateName[korg1212->cardState], korg1212->opencnt); mutex_lock(&korg1212->open_mutex); if (--(korg1212->opencnt)) { @@ -687,8 +674,9 @@ static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_StopPlay, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "CloseCard - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { mutex_unlock(&korg1212->open_mutex); return 0; @@ -711,8 +699,8 @@ static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->setcnt); + dev_dbg(korg1212->card->dev, "SetupForPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->setcnt); if (korg1212->setcnt++) return 0; @@ -721,8 +709,8 @@ static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212) rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_SetupPlay, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "SetupForPlay - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { return 1; } @@ -734,8 +722,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->playcnt); + dev_dbg(korg1212->card->dev, "K1212_DEBUG: TriggerPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->playcnt); if (korg1212->playcnt++) return 0; @@ -743,8 +731,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) snd_korg1212_setCardState(korg1212, K1212_STATE_PLAYING); rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_TriggerPlay, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "TriggerPlay - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { return 1; } @@ -754,8 +742,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) /* spinlock already held */ static int snd_korg1212_StopPlay(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: StopPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->playcnt); + dev_dbg(korg1212->card->dev, "StopPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->playcnt); if (--(korg1212->playcnt)) return 0; @@ -866,8 +854,9 @@ static int snd_korg1212_SetRate(struct snd_korg1212 *korg1212, int rate) ClockSourceSelector[korg1212->clkSrcRate], 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Set Clock Source Selector - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); return 0; } @@ -902,8 +891,8 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) u16 count; unsigned long flags; - K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "WriteADCSensivity [%s]\n", + stateName[korg1212->cardState]); // ---------------------------------------------------------------------------- // initialize things. The local init bit is always set when writing to the @@ -1026,8 +1015,9 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_MonitorOn, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "WriteADCSensivity - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); } spin_unlock_irqrestore(&korg1212->lock, flags); @@ -1039,8 +1029,8 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) { int channel, rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is complete. [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "DSP download is complete. [%s]\n", + stateName[korg1212->cardState]); // ---------------------------------------------------- // tell the card to boot @@ -1048,8 +1038,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_BootFromDSPPage4, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Boot from Page 4 - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Boot from Page 4 - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); msleep(DSP_BOOT_DELAY_IN_MS); // -------------------------------------------------------------------------------- @@ -1065,8 +1056,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Buffer Memory - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Configure Buffer Memory - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); udelay(INTERCOMMAND_DELAY); @@ -1079,8 +1071,8 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Misc Memory - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Configure Misc Memory - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); // -------------------------------------------------------------------------------- // Initialize the routing and volume tables, then update the card's state. @@ -1100,15 +1092,16 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ClockSourceSelector[korg1212->clkSrcRate], 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Set Clock Source Selector - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); rc = snd_korg1212_TurnOnIdleMonitor(korg1212); snd_korg1212_setCardState(korg1212, K1212_STATE_READY); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Monitor On - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Set Monitor On - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); snd_korg1212_setCardState(korg1212, K1212_STATE_DSP_COMPLETE); } @@ -1133,9 +1126,10 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) switch (doorbellValue) { case K1212_DB_DSPDownloadDone: - K1212_DEBUG_PRINTK("K1212_DEBUG: IRQ DNLD count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ DNLD count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); if (korg1212->cardState == K1212_STATE_DSP_IN_PROCESS) { korg1212->dsp_is_loaded = 1; wake_up(&korg1212->wait); @@ -1146,10 +1140,11 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) // an error occurred - stop the card // ------------------------------------------------------------------------ case K1212_DB_DMAERROR: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DMAE count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); - snd_printk(KERN_ERR "korg1212: DMA Error\n"); + dev_dbg(korg1212->card->dev, + "IRQ DMAE count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); + dev_err(korg1212->card->dev, "DMA Error\n"); korg1212->errorcnt++; korg1212->totalerrorcnt++; korg1212->sharedBufferPtr->cardCommand = 0; @@ -1161,16 +1156,19 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) // the semaphore in case someone is waiting for this. // ------------------------------------------------------------------------ case K1212_DB_CARDSTOPPED: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ CSTP count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ CSTP count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); korg1212->sharedBufferPtr->cardCommand = 0; break; default: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", - korg1212->irqcount, doorbellValue, - korg1212->currentBuffer, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", + korg1212->irqcount, doorbellValue, + korg1212->currentBuffer, + stateName[korg1212->cardState]); if ((korg1212->cardState > K1212_STATE_SETUP) || korg1212->idleMonitorOn) { korg1212->currentBuffer++; @@ -1206,8 +1204,8 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is starting... [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "DSP download is starting... [%s]\n", + stateName[korg1212->cardState]); // --------------------------------------------------------------- // verify the state of the card before proceeding. @@ -1221,8 +1219,9 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) UpperWordSwap(korg1212->dma_dsp.addr), 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Start DSP Download RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Start DSP Download RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); korg1212->dsp_is_loaded = 0; wait_event_timeout(korg1212->wait, korg1212->dsp_is_loaded, HZ * CARD_BOOT_TIMEOUT); @@ -1281,20 +1280,20 @@ static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int coun struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos; int i; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n", - pos, offset, size, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n", + pos, offset, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", - dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", + dst, i); return -EFAULT; } -#endif memset((void*) dst + offset, 0, size); dst++; } @@ -1307,22 +1306,25 @@ static int snd_korg1212_copy_to(struct snd_korg1212 *korg1212, void __user *dst, struct KorgAudioFrame * src = korg1212->recordDataBufsPtr[0].bufferData + pos; int i, rc; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_to pos=%d offset=%d size=%d\n", - pos, offset, size); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to pos=%d offset=%d size=%d\n", + pos, offset, size); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) src < (void *) korg1212->recordDataBufsPtr || (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } -#endif rc = copy_to_user(dst + offset, src, size); if (rc) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } src++; @@ -1337,23 +1339,26 @@ static int snd_korg1212_copy_from(struct snd_korg1212 *korg1212, void __user *sr struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos; int i, rc; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n", - pos, offset, size, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n", + pos, offset, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } -#endif rc = copy_from_user((void*) dst + offset, src, size); if (rc) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } dst++; @@ -1367,8 +1372,8 @@ static void snd_korg1212_free_pcm(struct snd_pcm *pcm) { struct snd_korg1212 *korg1212 = pcm->private_data; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_free_pcm [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_free_pcm [%s]\n", + stateName[korg1212->cardState]); korg1212->pcm = NULL; } @@ -1379,8 +1384,8 @@ static int snd_korg1212_playback_open(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_open [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_open [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_OpenCard(korg1212); @@ -1408,8 +1413,8 @@ static int snd_korg1212_capture_open(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_open [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_open [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_OpenCard(korg1212); @@ -1435,8 +1440,8 @@ static int snd_korg1212_playback_close(struct snd_pcm_substream *substream) unsigned long flags; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_close [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_close [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_silence(korg1212, 0, K1212_MAX_SAMPLES, 0, korg1212->channels * 2); @@ -1457,8 +1462,8 @@ static int snd_korg1212_capture_close(struct snd_pcm_substream *substream) unsigned long flags; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_close [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_close [%s]\n", + stateName[korg1212->cardState]); spin_lock_irqsave(&korg1212->lock, flags); @@ -1475,14 +1480,18 @@ static int snd_korg1212_capture_close(struct snd_pcm_substream *substream) static int snd_korg1212_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_ioctl: cmd=%d\n", cmd); + struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); + + dev_dbg(korg1212->card->dev, "snd_korg1212_ioctl: cmd=%d\n", cmd); if (cmd == SNDRV_PCM_IOCTL1_CHANNEL_INFO ) { struct snd_pcm_channel_info *info = arg; info->offset = 0; info->first = info->channel * 16; info->step = 256; - K1212_DEBUG_PRINTK("K1212_DEBUG: channel_info %d:, offset=%ld, first=%d, step=%d\n", info->channel, info->offset, info->first, info->step); + dev_dbg(korg1212->card->dev, + "channel_info %d:, offset=%ld, first=%d, step=%d\n", + info->channel, info->offset, info->first, info->step); return 0; } @@ -1498,8 +1507,8 @@ static int snd_korg1212_hw_params(struct snd_pcm_substream *substream, pid_t this_pid; pid_t other_pid; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_hw_params [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_hw_params [%s]\n", + stateName[korg1212->cardState]); spin_lock_irqsave(&korg1212->lock, flags); @@ -1546,15 +1555,16 @@ static int snd_korg1212_prepare(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_prepare [%s]\n", + stateName[korg1212->cardState]); spin_lock_irq(&korg1212->lock); /* FIXME: we should wait for ack! */ if (korg1212->stop_pending_cnt > 0) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare - Stop is pending... [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "snd_korg1212_prepare - Stop is pending... [%s]\n", + stateName[korg1212->cardState]); spin_unlock_irq(&korg1212->lock); return -EAGAIN; /* @@ -1579,8 +1589,8 @@ static int snd_korg1212_trigger(struct snd_pcm_substream *substream, struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_trigger [%s] cmd=%d\n", - stateName[korg1212->cardState], cmd); + dev_dbg(korg1212->card->dev, "snd_korg1212_trigger [%s] cmd=%d\n", + stateName[korg1212->cardState], cmd); spin_lock(&korg1212->lock); switch (cmd) { @@ -1621,8 +1631,9 @@ static snd_pcm_uframes_t snd_korg1212_playback_pointer(struct snd_pcm_substream pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_pointer [%s] %ld\n", - stateName[korg1212->cardState], pos); + dev_dbg(korg1212->card->dev, + "snd_korg1212_playback_pointer [%s] %ld\n", + stateName[korg1212->cardState], pos); return pos; } @@ -1634,8 +1645,8 @@ static snd_pcm_uframes_t snd_korg1212_capture_pointer(struct snd_pcm_substream * pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_pointer [%s] %ld\n", - stateName[korg1212->cardState], pos); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_pointer [%s] %ld\n", + stateName[korg1212->cardState], pos); return pos; } @@ -1648,8 +1659,9 @@ static int snd_korg1212_playback_copy(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_copy [%s] %ld %ld\n", - stateName[korg1212->cardState], pos, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_playback_copy [%s] %ld %ld\n", + stateName[korg1212->cardState], pos, count); return snd_korg1212_copy_from(korg1212, src, pos, count, 0, korg1212->channels * 2); @@ -1662,8 +1674,8 @@ static int snd_korg1212_playback_silence(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_silence [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_silence [%s]\n", + stateName[korg1212->cardState]); return snd_korg1212_silence(korg1212, pos, count, 0, korg1212->channels * 2); } @@ -1676,8 +1688,9 @@ static int snd_korg1212_capture_copy(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_copy [%s] %ld %ld\n", - stateName[korg1212->cardState], pos, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_capture_copy [%s] %ld %ld\n", + stateName[korg1212->cardState], pos, count); return snd_korg1212_copy_to(korg1212, dst, pos, count, 0, korg1212->channels * 2); } @@ -2138,7 +2151,7 @@ snd_korg1212_free(struct snd_korg1212 *korg1212) static int snd_korg1212_dev_free(struct snd_device *device) { struct snd_korg1212 *korg1212 = device->device_data; - K1212_DEBUG_PRINTK("K1212_DEBUG: Freeing device\n"); + dev_dbg(korg1212->card->dev, "Freeing device\n"); return snd_korg1212_free(korg1212); } @@ -2210,19 +2223,17 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, ioport_size = pci_resource_len(korg1212->pci, 1); iomem2_size = pci_resource_len(korg1212->pci, 2); - K1212_DEBUG_PRINTK("K1212_DEBUG: resources:\n" - " iomem = 0x%lx (%d)\n" - " ioport = 0x%lx (%d)\n" - " iomem = 0x%lx (%d)\n" - " [%s]\n", - korg1212->iomem, iomem_size, - korg1212->ioport, ioport_size, - korg1212->iomem2, iomem2_size, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "resources:\n\tiomem = 0x%lx (%d)\n\tioport = 0x%lx (%d)\n\tiomem = 0x%lx (%d)\n\t[%s]\n", + korg1212->iomem, iomem_size, + korg1212->ioport, ioport_size, + korg1212->iomem2, iomem2_size, + stateName[korg1212->cardState]); if ((korg1212->iobase = ioremap(korg1212->iomem, iomem_size)) == NULL) { - snd_printk(KERN_ERR "korg1212: unable to remap memory region 0x%lx-0x%lx\n", korg1212->iomem, - korg1212->iomem + iomem_size - 1); + dev_err(korg1212->card->dev, + "unable to remap memory region 0x%lx-0x%lx\n", + korg1212->iomem, korg1212->iomem + iomem_size - 1); snd_korg1212_free(korg1212); return -EBUSY; } @@ -2232,7 +2243,8 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, KBUILD_MODNAME, korg1212); if (err) { - snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq); + dev_err(korg1212->card->dev, "unable to grab IRQ %d\n", + pci->irq); snd_korg1212_free(korg1212); return -EBUSY; } @@ -2252,40 +2264,45 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->sensRegPtr = (u16 __iomem *) (korg1212->iobase + SENS_CONTROL_OFFSET); korg1212->idRegPtr = (u32 __iomem *) (korg1212->iobase + DEV_VEND_ID_OFFSET); - K1212_DEBUG_PRINTK("K1212_DEBUG: card registers:\n" - " Status register = 0x%p\n" - " OutDoorbell = 0x%p\n" - " InDoorbell = 0x%p\n" - " Mailbox0 = 0x%p\n" - " Mailbox1 = 0x%p\n" - " Mailbox2 = 0x%p\n" - " Mailbox3 = 0x%p\n" - " ControlReg = 0x%p\n" - " SensReg = 0x%p\n" - " IDReg = 0x%p\n" - " [%s]\n", - korg1212->statusRegPtr, - korg1212->outDoorbellPtr, - korg1212->inDoorbellPtr, - korg1212->mailbox0Ptr, - korg1212->mailbox1Ptr, - korg1212->mailbox2Ptr, - korg1212->mailbox3Ptr, - korg1212->controlRegPtr, - korg1212->sensRegPtr, - korg1212->idRegPtr, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "card registers:\n" + " Status register = 0x%p\n" + " OutDoorbell = 0x%p\n" + " InDoorbell = 0x%p\n" + " Mailbox0 = 0x%p\n" + " Mailbox1 = 0x%p\n" + " Mailbox2 = 0x%p\n" + " Mailbox3 = 0x%p\n" + " ControlReg = 0x%p\n" + " SensReg = 0x%p\n" + " IDReg = 0x%p\n" + " [%s]\n", + korg1212->statusRegPtr, + korg1212->outDoorbellPtr, + korg1212->inDoorbellPtr, + korg1212->mailbox0Ptr, + korg1212->mailbox1Ptr, + korg1212->mailbox2Ptr, + korg1212->mailbox3Ptr, + korg1212->controlRegPtr, + korg1212->sensRegPtr, + korg1212->idRegPtr, + stateName[korg1212->cardState]); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), sizeof(struct KorgSharedBuffer), &korg1212->dma_shared) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%Zd bytes)\n", sizeof(struct KorgSharedBuffer)); + dev_err(korg1212->card->dev, + "can not allocate shared buffer memory (%Zd bytes)\n", + sizeof(struct KorgSharedBuffer)); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->sharedBufferPtr = (struct KorgSharedBuffer *)korg1212->dma_shared.area; korg1212->sharedBufferPhy = korg1212->dma_shared.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Shared Buffer Area = 0x%p (0x%08lx), %d bytes\n", korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, sizeof(struct KorgSharedBuffer)); + dev_dbg(korg1212->card->dev, + "Shared Buffer Area = 0x%p (0x%08lx), %lu bytes\n", + korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, + sizeof(struct KorgSharedBuffer)); #ifndef K1212_LARGEALLOC @@ -2293,27 +2310,35 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), korg1212->DataBufsSize, &korg1212->dma_play) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate play data buffer memory (%d bytes)\n", korg1212->DataBufsSize); + dev_err(korg1212->card->dev, + "can not allocate play data buffer memory (%d bytes)\n", + korg1212->DataBufsSize); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->playDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_play.area; korg1212->PlayDataPhy = korg1212->dma_play.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Play Data Area = 0x%p (0x%08x), %d bytes\n", - korg1212->playDataBufsPtr, korg1212->PlayDataPhy, korg1212->DataBufsSize); + dev_dbg(korg1212->card->dev, + "Play Data Area = 0x%p (0x%08x), %d bytes\n", + korg1212->playDataBufsPtr, korg1212->PlayDataPhy, + korg1212->DataBufsSize); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), korg1212->DataBufsSize, &korg1212->dma_rec) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate record data buffer memory (%d bytes)\n", korg1212->DataBufsSize); + dev_err(korg1212->card->dev, + "can not allocate record data buffer memory (%d bytes)\n", + korg1212->DataBufsSize); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->recordDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_rec.area; korg1212->RecDataPhy = korg1212->dma_rec.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Record Data Area = 0x%p (0x%08x), %d bytes\n", - korg1212->recordDataBufsPtr, korg1212->RecDataPhy, korg1212->DataBufsSize); + dev_dbg(korg1212->card->dev, + "Record Data Area = 0x%p (0x%08x), %d bytes\n", + korg1212->recordDataBufsPtr, korg1212->RecDataPhy, + korg1212->DataBufsSize); #else // K1212_LARGEALLOC @@ -2334,22 +2359,25 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, err = request_firmware(&dsp_code, "korg/k1212.dsp", &pci->dev); if (err < 0) { release_firmware(dsp_code); - snd_printk(KERN_ERR "firmware not available\n"); + dev_err(korg1212->card->dev, "firmware not available\n"); snd_korg1212_free(korg1212); return err; } if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), dsp_code->size, &korg1212->dma_dsp) < 0) { - snd_printk(KERN_ERR "korg1212: cannot allocate dsp code memory (%zd bytes)\n", dsp_code->size); + dev_err(korg1212->card->dev, + "cannot allocate dsp code memory (%zd bytes)\n", + dsp_code->size); snd_korg1212_free(korg1212); release_firmware(dsp_code); return -ENOMEM; } - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", - korg1212->dma_dsp.area, korg1212->dma_dsp.addr, dsp_code->size, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", + korg1212->dma_dsp.area, (int)korg1212->dma_dsp.addr, + (int)dsp_code->size, stateName[korg1212->cardState]); memcpy(korg1212->dma_dsp.area, dsp_code->data, dsp_code->size); @@ -2358,7 +2386,8 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_RebootCard, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Reboot Card - RC = %d [%s]\n", rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Reboot Card - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, korg1212, &ops)) < 0) { snd_korg1212_free(korg1212); @@ -2372,18 +2401,22 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, if (snd_korg1212_downloadDSPCode(korg1212)) return -EBUSY; - K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], " - "PlayDataPhy = %08x L[%08x]\n" - "korg1212: RecDataPhy = %08x L[%08x], " - "VolumeTablePhy = %08x L[%08x]\n" - "korg1212: RoutingTablePhy = %08x L[%08x], " - "AdatTimeCodePhy = %08x L[%08x]\n", - (int)korg1212->dma_dsp.addr, UpperWordSwap(korg1212->dma_dsp.addr), - korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy), - korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy), - korg1212->VolumeTablePhy, LowerWordSwap(korg1212->VolumeTablePhy), - korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy), - korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy)); + dev_dbg(korg1212->card->dev, "dspMemPhy = %08x U[%08x], " + "PlayDataPhy = %08x L[%08x]\n" + "korg1212: RecDataPhy = %08x L[%08x], " + "VolumeTablePhy = %08x L[%08x]\n" + "korg1212: RoutingTablePhy = %08x L[%08x], " + "AdatTimeCodePhy = %08x L[%08x]\n", + (int)korg1212->dma_dsp.addr, + UpperWordSwap(korg1212->dma_dsp.addr), + korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy), + korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy), + korg1212->VolumeTablePhy, + LowerWordSwap(korg1212->VolumeTablePhy), + korg1212->RoutingTablePhy, + LowerWordSwap(korg1212->RoutingTablePhy), + korg1212->AdatTimeCodePhy, + LowerWordSwap(korg1212->AdatTimeCodePhy)); if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0) return err; @@ -2446,7 +2479,7 @@ snd_korg1212_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, korg1212->iomem, korg1212->irq); - K1212_DEBUG_PRINTK("K1212_DEBUG: %s\n", card->longname); + dev_dbg(&pci->dev, "%s\n", card->longname); if ((err = snd_card_register(card)) < 0) { snd_card_free(card); -- 1.8.1.2

3 6

[alsa-devel] [PATCH] ASoC: rt5631: Fixing compilation warning when DT is disabled
by Krishna Mohan Dani 26 Nov '14

26 Nov '14

Fixes the following compilation warning: Warning: 'rt5631_i2c_dt_ids' defined but not used - when DT is not used. Signed-off-by: Claude Youn <claude.youn(a)gmail.com> Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- sound/soc/codecs/rt5631.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3b7d5e4..5c65915 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1691,7 +1691,7 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); -static struct of_device_id rt5631_i2c_dt_ids[] = { +static struct of_device_id rt5631_i2c_dt_ids[] __maybe_unused = { { .compatible = "realtek,rt5631"}, { .compatible = "realtek,alc5631"}, { } -- 1.7.9.5

3 3

[alsa-devel] [PATCH] ASoC: Samsung: Add arndale_rt5631 machine driver and binding
by Krishna Mohan Dani 26 Nov '14

26 Nov '14

Adding machine driver to instantiate I2S based realtek's ALC5631 sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec hence support for ALC5631 based machine driver is being added. This patch also documents the device tree binding for the Arndale board based machine driver. Signed-off-by: Claude Youn <claude.youn(a)gmail.com> Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- .../devicetree/bindings/sound/arndale.txt | 24 ++++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 150 ++++++++++++++++++++ 4 files changed, 182 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt new file mode 100644 index 0000000..0e76946 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arndale.txt @@ -0,0 +1,24 @@ +Audio Binding for Arndale boards + +Required properties: +- compatible : Can be the following, + "samsung,arndale-rt5631" + +- samsung,audio-cpu: The phandle of the Samsung I2S controller +- samsung,audio-codec: The phandle of the audio codec + +Optional: +- samsung,model: The name of the sound-card + +Arndale Boards has many audio daughter cards, one of them is +rt5631/alc5631. Below example shows audio bindings for rt5631/ +alc5631 based codec. + +Example: + +sound { + compatible = "samsung,arndale-rt5631"; + + samsung,audio-cpu = <&i2s0> + samsung,audio-codec = <&rt5631>; +}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 55a3869..80b5c61 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -239,3 +239,9 @@ config SND_SOC_ODROIDX2 select SND_SAMSUNG_I2S help Say Y here to enable audio support for the Odroid-X2/U3. + +config SND_SOC_ARNDALE_RT5631_ALC5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on SND_SOC_SAMSUNG + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 91505dd..31e3dba 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -45,6 +45,7 @@ snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o +snd-soc-arndale-rt5631-objs := arndale_rt5631.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o +obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c new file mode 100644 index 0000000..1e2b61c --- /dev/null +++ b/sound/soc/samsung/arndale_rt5631.c @@ -0,0 +1,150 @@ +/* + * arndale_rt5631.c + * + * Copyright (c) 2014, Insignal Co., Ltd. + * + * Author: Claude <claude(a)insginal.co.kr> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "i2s.h" + +static int arndale_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int rfs, ret; + unsigned long rclk; + + rfs = 256; + + rclk = params_rate(params) * rfs; + + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, + 0, SND_SOC_CLOCK_OUT); + + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops arndale_ops = { + .hw_params = arndale_hw_params, +}; + +static struct snd_soc_dai_link arndale_rt5631_dai[] = { + { + .name = "RT5631 HiFi", + .stream_name = "Primary", + .codec_dai_name = "rt5631-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ops = &arndale_ops, + }, +}; + +static struct snd_soc_card arndale_rt5631 = { + .name = "Arndale RT5631", + .dai_link = arndale_rt5631_dai, + .num_links = ARRAY_SIZE(arndale_rt5631_dai), +}; + +static int arndale_audio_probe(struct platform_device *pdev) +{ + int n, ret; + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &arndale_rt5631; + + card->dev = &pdev->dev; + + for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { + if (!arndale_rt5631_dai[n].cpu_dai_name) { + arndale_rt5631_dai[n].cpu_of_node = of_parse_phandle(np, + "samsung,audio-cpu", n); + + if (!arndale_rt5631_dai[n].cpu_of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } + } + if (!arndale_rt5631_dai[n].platform_name) + arndale_rt5631_dai[n].platform_of_node = + arndale_rt5631_dai[n].cpu_of_node; + + arndale_rt5631_dai[n].codec_name = NULL; + arndale_rt5631_dai[n].codec_of_node = of_parse_phandle(np, + "samsung,audio-codec", n); + if (!arndale_rt5631_dai[0].codec_of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-codec' missing or invalid\n"); + return -EINVAL; + } + } + + ret = devm_snd_soc_register_card(card->dev, card); + + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); + + return ret; +} + +static int arndale_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { + { .compatible = "samsung,arndale-rt5631", }, + { .compatible = "samsung,arndale-alc5631", }, + {}, +}; +MODULE_DEVICE_TABLE(of, samsung_arndale_rt5631_of_match); + +static struct platform_driver arndale_audio_driver = { + .driver = { + .name = "arndale-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), + }, + .probe = arndale_audio_probe, + .remove = arndale_audio_remove, +}; + +module_platform_driver(arndale_audio_driver); + +MODULE_AUTHOR("Claude <claude(a)insignal.co.kr>"); +MODULE_DESCRIPTION("ALSA SoC Driver for Arndale Board"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: bt-sco: support wide band speech
by Barry Song 26 Nov '14

26 Nov '14

From: Yibo Cai <Yibo.Cai(a)csr.com> HFP1.6 requires support for wide band speech(16KHz) over Bluetooth. Signed-off-by: Yibo Cai <Yibo.Cai(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/bt-sco.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index c4cf069..3878491 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -52,6 +52,14 @@ static struct snd_soc_codec_driver soc_codec_dev_bt_sco = { static int bt_sco_probe(struct platform_device *pdev) { + const struct platform_device_id *id = platform_get_device_id(pdev); + + /* Check wide band SCO(16KHz) support */ + if (id && (strcmp(id->name, "bt-sco-wb") == 0)) { + bt_sco_dai.playback.rates |= SNDRV_PCM_RATE_16000; + bt_sco_dai.capture.rates |= SNDRV_PCM_RATE_16000; + } + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, &bt_sco_dai, 1); } @@ -70,6 +78,9 @@ static struct platform_device_id bt_sco_driver_ids[] = { { .name = "bt-sco", }, + { + .name = "bt-sco-wb", + }, {}, }; MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); -- 2.1.3

3 2

[alsa-devel] [PATCH] ASoC: wm8960: LRCM operation should after software reset
by Zidan Wang 26 Nov '14

26 Nov '14

When shared_lrclk is set, LRCM will be enabled. But the following software reset in wm8960_probe will reset it to the default state. So LRCM operation should after software reset. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index bc8793cd..21db90c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -960,6 +960,14 @@ static int wm8960_probe(struct snd_soc_codec *codec) return ret; } + if (pdata->shared_lrclk) { + ret = snd_soc_update_bits(codec, WM8960_ADDCTL2, 0x4, 0x4); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable LRCM: %d\n", ret); + return ret; + } + } + /* Latch the update bits */ snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); @@ -1030,15 +1038,6 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, else if (i2c->dev.of_node) wm8960_set_pdata_from_of(i2c, &wm8960->pdata); - if (pdata && pdata->shared_lrclk) { - ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, - 0x4, 0x4); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", - ret); - return ret; - } - } i2c_set_clientdata(i2c, wm8960); -- 1.9.1

3 2

[alsa-devel] [PATCH V2 0/6] Add support for the WM8280 and WM8281 codecs
by Richard Fitzgerald 26 Nov '14

26 Nov '14

This set of patches adds support for the WM8280 and WM8281 codecs to the Wolfson Arizona drivers. Only change since V1 series is update to device tree binding documentation in patch 2 Richard Fitzgerald (6): mfd: arizona: add support for WM8280/WM8281 Documentation: devicetree: arizona: Add bindings for WM8280 regulator: arizona-micsupp: Add support for WM8280/WM8281 gpio: arizona: Add support for WM8280/WM8281 extcon: arizona: Add support for WM8280/WM8281 ASoC: arizona: Add support for WM8280/WM8281 Documentation/devicetree/bindings/mfd/arizona.txt | 15 +++++++++++---- drivers/extcon/extcon-arizona.c | 1 + drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 5 +++-- drivers/mfd/arizona-core.c | 14 ++++++++++++-- drivers/mfd/arizona-i2c.c | 2 ++ drivers/mfd/arizona-irq.c | 1 + drivers/mfd/arizona-spi.c | 2 ++ drivers/regulator/arizona-micsupp.c | 1 + include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 2 ++ 11 files changed, 37 insertions(+), 8 deletions(-) -- 1.7.2.5

2 7

[alsa-devel] Where to find Alsa experienced programmer?
by Andrej Falout 25 Nov '14

25 Nov '14

Hello, I have a client that would like to hire services of an skilled programmer with Alsa related experience, for a particular project contract task. They advertised locally, but came up with nothing, which is probably not a big surprise. They have accepted remote/online working as an option. Could anyone suggest the appropriate method/place to seek such a person? Any suggestions gratefully received. Kind regards, Andrej Falout

2 1

Re: [alsa-devel] [PATCH 26/29] ALSA: oxfw: Add support AMDTP in-stream
by Clemens Ladisch 25 Nov '14

25 Nov '14

Takashi Sakamoto wrote: > On Nov 24 2014 22:54, Clemens Ladisch wrote: >> In theory, it would be possible to use atomic_t to synchronize multiple >> streams, but this requires that >> 1. two CPUs that start a stream at the same time do not both think they >> are the first, ... and >> 2. when one CPU is still starting a stream, code on all other CPUs must >> be able to work correctly, without synchronization. [...] > > If I rewrite your pseudo code, like this: > >> void start_some_substream(...) >> { >> if (atomic_inc_and_test(substreams)) { >> mutex_lock(); >> // lots of stuff to start DMA ... > > if (wait_for_first_packet() == ETIMEDOUT) { > stop_dma(); > release_streams(); > mutex_unlock(); > return ETIMEDOUT; > } >> mutex_unlock(); >> } >> ... >> } This is an even better example of why atomic_t is incorrect: when the startup fails, the second call to start_some_substream() at the same time, which just sees that substreams>0, cannot notice the error, and thinks its own substream has been started successfully. Regards, Clemens

2 1

Re: [alsa-devel] [PATCH v3] ASoC: Remove 'const' from the device_node pointers
by Mark Brown 25 Nov '14

25 Nov '14

On Tue, Nov 25, 2014 at 12:14:48PM +0100, Jean-Francois Moine wrote: > As Russell King's explained it, there should not be pointers to > struct device_node: OK, found a user that needs this so applying now. Please do try to make your changelogs clearer about why things are being done, it's often hard to figure things out which slows everything down.

1 0

[alsa-devel] Remove my address from alsa-devel@alsa-project,org
by John Houston 25 Nov '14

25 Nov '14

jbhdbr(a)gmail.com Sent from my iPhone

1 0

[alsa-devel] [PATCH] ASoC: Disable regmap helpers if regmap is disabled
by Lars-Peter Clausen 25 Nov '14

25 Nov '14

If regmap is disabled there will be no users of the ASoC regmap helpers. Furthermore regmap_exit() will no be defined causing the following compile error: sound/soc/soc-core.c: In function 'snd_soc_component_exit_regmap': sound/soc/soc-core.c:2645:2: error: implicit declaration of function 'regmap_exit' [-Werror=implicit-function-declaration] So disable the helpers if regmap is disabled. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Fixes: 20feb881988c ASoC: Add helper functions for deferred regmap setup") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 ++++ sound/soc/soc-core.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 879e2b3..4cef90cc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1283,6 +1283,8 @@ void snd_soc_component_async_complete(struct snd_soc_component *component); int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +#ifdef CONFIG_REGMAP + void snd_soc_component_init_regmap(struct snd_soc_component *component, struct regmap *regmap); void snd_soc_component_exit_regmap(struct snd_soc_component *component); @@ -1318,6 +1320,8 @@ static inline void snd_soc_codec_exit_regmap(struct snd_soc_codec *codec) snd_soc_component_exit_regmap(&codec->component); } +#endif + /* device driver data */ static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cc52ea1..7904eda 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2612,6 +2612,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component) component->val_bytes = val_bytes; } +#ifdef CONFIG_REGMAP + /** * snd_soc_component_init_regmap() - Initialize regmap instance for the component * @component: The component for which to initialize the regmap instance @@ -2647,6 +2649,8 @@ void snd_soc_component_exit_regmap(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(snd_soc_component_exit_regmap); +#endif + static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { if (!component->write && !component->read) { -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: qi_lb60: Pass flags to gpiod_get()
by Lars-Peter Clausen 25 Nov '14

25 Nov '14

Pass flags to gpiod_get() to automatically configure the GPIOs. This is shorter and not passing any flags to gpiod_get() will eventually no longer be supported. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/qi_lb60.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 5cb91f9..0fb7d2a 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -77,25 +77,18 @@ static int qi_lb60_probe(struct platform_device *pdev) { struct qi_lb60 *qi_lb60; struct snd_soc_card *card = &qi_lb60_card; - int ret; qi_lb60 = devm_kzalloc(&pdev->dev, sizeof(*qi_lb60), GFP_KERNEL); if (!qi_lb60) return -ENOMEM; - qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd"); + qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd", GPIOD_OUT_LOW); if (IS_ERR(qi_lb60->snd_gpio)) return PTR_ERR(qi_lb60->snd_gpio); - ret = gpiod_direction_output(qi_lb60->snd_gpio, 0); - if (ret) - return ret; - qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp"); + qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp", GPIOD_OUT_LOW); if (IS_ERR(qi_lb60->amp_gpio)) return PTR_ERR(qi_lb60->amp_gpio); - ret = gpiod_direction_output(qi_lb60->amp_gpio, 0); - if (ret) - return ret; card->dev = &pdev->dev; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] alsa-lib: Fix assorted tstamp_type bugs/omissions.
by Tim Cussins 25 Nov '14

25 Nov '14

Just a couple of quick fixes related to tstamp_type. - snd_pcm_sw_params(): copy tstamp_type field - snd_pcm_tstamp_type_name(): fix argument type - snd_pcm_dump_sw_setup(): fix dumping of tstamp_type field --- src/pcm/pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 3904256..baa47c7 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -897,6 +897,7 @@ int snd_pcm_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) if (err < 0) return err; pcm->tstamp_mode = params->tstamp_mode; + pcm->tstamp_type = params->tstamp_type; pcm->period_step = params->period_step; pcm->avail_min = params->avail_min; pcm->period_event = sw_get_period_event(params); @@ -1843,7 +1844,7 @@ const char *snd_pcm_tstamp_mode_name(snd_pcm_tstamp_t mode) * \param mode PCM tstamp type * \return ascii name of PCM tstamp type setting */ -const char *snd_pcm_tstamp_type_name(snd_pcm_tstamp_t type) +const char *snd_pcm_tstamp_type_name(snd_pcm_tstamp_type_t type) { if (type > SND_PCM_TSTAMP_TYPE_LAST) return NULL; @@ -1924,7 +1925,7 @@ int snd_pcm_dump_sw_setup(snd_pcm_t *pcm, snd_output_t *out) return -EIO; } snd_output_printf(out, " tstamp_mode : %s\n", snd_pcm_tstamp_mode_name(pcm->tstamp_mode)); - snd_output_printf(out, " tstamp_type : %s\n", snd_pcm_tstamp_type_name(pcm->tstamp_mode)); + snd_output_printf(out, " tstamp_type : %s\n", snd_pcm_tstamp_type_name(pcm->tstamp_type)); snd_output_printf(out, " period_step : %d\n", pcm->period_step); snd_output_printf(out, " avail_min : %ld\n", pcm->avail_min); snd_output_printf(out, " period_event : %i\n", pcm->period_event); -- 1.7.10.4

2 1

[alsa-devel] Cache issue on BayTrail (ValleyView2) HD-audio
by Takashi Iwai 25 Nov '14

25 Nov '14

Hi Mengdong, right now I'm playing onboard HD-audio on a Baytrail (ValleyView2) machine, not DP/HDMI output, and I noticed that the snooping doesn't seem working with this. The playback sometimes stutters / gets noises. This is a typical behavior with caching mismatch and indeed it works when I disable snooping via snoop=0 option. Could you check whether this can be seen generically? Also, can anyone check whether snoop=0 for Baytrail HD-audio causes any issues? In general, snoop=0 isn't too bad (IIRC, Windows does it as default), so we can turn it off at least for Baytrail as default. thanks, Takashi

1 1

Re: [alsa-devel] [PATCH v3] ASoC: Remove 'const' from the device_node pointers
by Mark Brown 25 Nov '14

25 Nov '14

On Tue, Nov 25, 2014 at 12:14:48PM +0100, Jean-Francois Moine wrote: > As Russell King's explained it, there should not be pointers to > struct device_node: > > "struct device_node is a ref-counted structure. That means if you > store a reference to it, you should "get" it, and you should "put" > it once you've done. The act of "put"ing the pointed-to structure > involves writing to that structure, so it is totally unappropriate > to store a device_node structure as a const pointer. It forces you > to have to cast it back to a non-const pointer at various points > in time to use various OF function calls." So, we're not holding references here (we're just doing comparisons, the references need to be owned before we get into the core) and I'm not seeing anything here removing casts?

2 2

[alsa-devel] [PATCH] ASoC: Intel: Fix stream volume set no effect issue on Broadwell
by Jie Yang 25 Nov '14

25 Nov '14

The volume setting control for capture stream doesn't take effect on intel Broadwell platform. Root cause it at 2 points: 1. set stream volume with channel=2 is wrongly bapassed; 2. the saved stream volume should be restored after stream is commit. Here correct these 2 items to fix the stream volume set issue. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 35 ++++++++++++++++++++++++++--------- sound/soc/intel/sst-haswell-ipc.h | 1 + sound/soc/intel/sst-haswell-pcm.c | 21 ++++++++++++--------- 3 files changed, 39 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index ffd5728..3f8c482 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1042,14 +1042,9 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, trace_ipc_request("set stream volume", stream->reply.stream_hw_id); - if (channel > 1) + if (channel >= 2 && channel != SST_HSW_CHANNELS_ALL) return -EINVAL; - if (stream->mute[channel]) { - stream->mute_volume[channel] = volume; - return 0; - } - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); @@ -1057,9 +1052,28 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, header |= (stage_id << IPC_STG_ID_SHIFT); req = &stream->vol_req; - req->channel = channel; req->target_volume = volume; + /* set both at same time ? */ + if (channel == SST_HSW_CHANNELS_ALL) { + if (hsw->mute[0] && hsw->mute[1]) { + hsw->mute_volume[0] = hsw->mute_volume[1] = volume; + return 0; + } else if (hsw->mute[0]) + req->channel = 1; + else if (hsw->mute[1]) + req->channel = 0; + else + req->channel = SST_HSW_CHANNELS_ALL; + } else { + /* set only 1 channel */ + if (hsw->mute[channel]) { + hsw->mute_volume[channel] = volume; + return 0; + } + req->channel = channel; + } + ret = ipc_tx_message_wait(hsw, header, req, sizeof(*req), NULL, 0); if (ret < 0) { dev_err(hsw->dev, "error: set stream volume failed\n"); @@ -1138,8 +1152,11 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, trace_ipc_request("set mixer volume", volume); + if (channel >= 2 && channel != SST_HSW_CHANNELS_ALL) + return -EINVAL; + /* set both at same time ? */ - if (channel == 2) { + if (channel == SST_HSW_CHANNELS_ALL) { if (hsw->mute[0] && hsw->mute[1]) { hsw->mute_volume[0] = hsw->mute_volume[1] = volume; return 0; @@ -1148,7 +1165,7 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, else if (hsw->mute[1]) req.channel = 0; else - req.channel = 0xffffffff; + req.channel = SST_HSW_CHANNELS_ALL; } else { /* set only 1 channel */ if (hsw->mute[channel]) { diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 387511f..138e894 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -24,6 +24,7 @@ #define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 #define SST_HSW_DX_CONTEXT_SIZE (640 * 1024) +#define SST_HSW_CHANNELS_ALL 0xffffffff #define SST_HSW_FW_LOG_CONFIG_DWORDS 12 #define SST_HSW_GLOBAL_LOG 15 diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 7eb9afc..331237f 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -189,7 +189,8 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0] == ucontrol->value.integer.value[1]) { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 2, volume); + /* apply volume value to all channels */ + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, SST_HSW_CHANNELS_ALL, volume); } else { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 0, volume); @@ -255,7 +256,7 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[1]) { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); - sst_hsw_mixer_set_volume(hsw, 0, 2, volume); + sst_hsw_mixer_set_volume(hsw, 0, SST_HSW_CHANNELS_ALL, volume); } else { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); @@ -525,7 +526,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, dev_err(rtd->dev, "error: failed to commit stream %d\n", ret); return ret; } - pcm_data->allocated = true; + + if (!pcm_data->allocated) { + /* Set previous saved volume */ + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 0, pcm_data->volume[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 1, pcm_data->volume[1]); + pcm_data->allocated = true; + } ret = sst_hsw_stream_pause(hsw, pcm_data->stream, 1); if (ret < 0) @@ -632,12 +641,6 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) return -EINVAL; } - /* Set previous saved volume */ - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, - 0, pcm_data->volume[0]); - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, - 1, pcm_data->volume[1]); - mutex_unlock(&pcm_data->mutex); return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: fsl: Use dynamic slot width for ESAI.
by Nicolin Chen 25 Nov '14

25 Nov '14

This series of patches add the support of dynamic slot width for ESAI driver. To do this, the PATCH-1 removes the fixed width. In order not to break those platforms that use fixed width, there comes the PATCH-2 to apply an override of slot_width. @Shengjiu, Will you be available to test this series on Sabre Auto for both Master and Slave cases? I'd like to wait for your Test-by. Thanks. Nicolin Chen (2): ASoC: fsl_esai: Use dynamic slot width as default ASoC: fsl-asoc-card: Add slot_width setting for cpu-dai sound/soc/fsl/fsl-asoc-card.c | 12 ++++++++++++ sound/soc/fsl/fsl_esai.c | 12 +++++++----- 2 files changed, 19 insertions(+), 5 deletions(-) -- 1.9.1

2 9

[alsa-devel] [PATCH v2] ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed
by Oder Chiou 25 Nov '14

25 Nov '14

After the patch "ASoC: rt5677: Use specific r/w function for DSP mode", the regmap_range "rt5677_ranges" was not registered in rt5677_regmap_physical, and it caused that the regmap_range "rt5677_ranges" cannot be accessed by the specific r/w function. The patch fixes this issue. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 37 ++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 2 +- 2 files changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2211f1..133010d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4287,6 +4287,7 @@ static int rt5677_probe(struct snd_soc_codec *codec) } mutex_init(&rt5677->dsp_cmd_lock); + mutex_init(&rt5677->dsp_pri_lock); return 0; } @@ -4344,10 +4345,19 @@ static int rt5677_read(void *context, unsigned int reg, unsigned int *val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_read(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + } + } else { regmap_read(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4357,10 +4367,20 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_write(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA, + val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + } + } else { regmap_write(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4495,10 +4515,13 @@ static const struct regmap_config rt5677_regmap_physical = { .reg_bits = 8, .val_bits = 16, - .max_register = RT5677_VENDOR_ID2 + 1, + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), .readable_reg = rt5677_readable_register, .cache_type = REGCACHE_NONE, + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), }; static const struct regmap_config rt5677_regmap = { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a02f64c..dbd9ffd 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1670,7 +1670,7 @@ struct rt5677_priv { struct rt5677_platform_data pdata; struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; - struct mutex dsp_cmd_lock; + struct mutex dsp_cmd_lock, dsp_pri_lock; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: rt5670: check if asrc is useable
by Bard Liao 25 Nov '14

25 Nov '14

To use ASRC, the sysclk should be faster than 384 times sample rate of I2S1. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- *Change source->codec to snd_soc_dapm_to_codec(source->dapm) --- sound/soc/codecs/rt5670.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index b842be0..fc26742 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -576,6 +576,18 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int can_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (rt5670->sysclk > rt5670->lrck[RT5670_AIF1] * 384) + return 1; + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, @@ -1639,8 +1651,8 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc }, { "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc }, - { "I2S1", NULL, "I2S1 ASRC" }, - { "I2S2", NULL, "I2S2 ASRC" }, + { "I2S1", NULL, "I2S1 ASRC", can_use_asrc}, + { "I2S2", NULL, "I2S2 ASRC", can_use_asrc}, { "DMIC1", NULL, "DMIC L1" }, { "DMIC1", NULL, "DMIC R1" }, -- 1.8.1.1.439.g50a6b54

3 5

[alsa-devel] [PATCH 1/2] ASoC: rt5645: multiple JD mode support
by Bard Liao 25 Nov '14

25 Nov '14

There are 3 JD modes in RT5645. This patch configure register values according to platform data. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5645.h | 1 + sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5645.h | 7 +++++++ 3 files changed, 42 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index 937f421..120d961 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -26,6 +26,7 @@ struct rt5645_platform_data { /* true if codec's jd function is used */ bool en_jd_func; + unsigned int jd_mode; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 31b5b1a..a7789a8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2241,7 +2241,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); - snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); snd_soc_dapm_sync(&codec->dapm); } @@ -2545,6 +2546,38 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, RT5645_IRQ_CLK_INT, RT5645_IRQ_CLK_INT); } + if (rt5645->pdata.jd_mode) { + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_IRQ_JD_1_1_EN, RT5645_IRQ_JD_1_1_EN); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_JD_PSV_MODE, RT5645_JD_PSV_MODE); + regmap_update_bits(rt5645->regmap, RT5645_HPO_MIXER, + RT5645_IRQ_PSV_MODE, RT5645_IRQ_PSV_MODE); + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_MIC2_OVCD_EN, RT5645_MIC2_OVCD_EN); + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + switch (rt5645->pdata.jd_mode) { + case 1: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_0); + break; + case 2: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_1); + break; + case 3: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_2); + break; + default: + break; + } + } + if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index c72220a..a815e36 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -594,6 +594,7 @@ #define RT5645_M_DAC1_HM_SFT 14 #define RT5645_M_HPVOL_HM (0x1 << 13) #define RT5645_M_HPVOL_HM_SFT 13 +#define RT5645_IRQ_PSV_MODE (0x1 << 12) /* SPK Left Mixer Control (0x46) */ #define RT5645_G_RM_L_SM_L_MASK (0x3 << 14) @@ -1350,6 +1351,10 @@ #define RT5645_PWR_CLK25M_PU (0x1 << 4) #define RT5645_IRQ_CLK_MCLK (0x0 << 3) #define RT5645_IRQ_CLK_INT (0x1 << 3) +#define RT5645_JD1_MODE_MASK (0x3 << 0) +#define RT5645_JD1_MODE_0 (0x0 << 0) +#define RT5645_JD1_MODE_1 (0x1 << 0) +#define RT5645_JD1_MODE_2 (0x2 << 0) /* VAD Control 4 (0x9d) */ #define RT5645_VAD_SEL_MASK (0x3 << 8) @@ -1638,6 +1643,7 @@ #define RT5645_OT_P_SFT 10 #define RT5645_OT_P_NOR (0x0 << 10) #define RT5645_OT_P_INV (0x1 << 10) +#define RT5645_IRQ_JD_1_1_EN (0x1 << 9) /* IRQ Control 2 (0xbe) */ #define RT5645_IRQ_MB1_OC_MASK (0x1 << 15) @@ -2120,6 +2126,7 @@ enum { #define RT5645_RXDP2_SEL_SFT (3) /* General Control3 (0xfc) */ +#define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) #define RT5645_MICINDET_MANU (0x1 << 7) -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v3] ASoC: Remove 'const' from the device_node pointers
by Jean-Francois Moine 25 Nov '14

25 Nov '14

As Russell King's explained it, there should not be pointers to struct device_node: "struct device_node is a ref-counted structure. That means if you store a reference to it, you should "get" it, and you should "put" it once you've done. The act of "put"ing the pointed-to structure involves writing to that structure, so it is totally unappropriate to store a device_node structure as a const pointer. It forces you to have to cast it back to a non-const pointer at various points in time to use various OF function calls." Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v3: change the patch comment (Mark Brown) v2: remove const from the pointers instead of modifying the OF functions (Russell King) --- include/sound/soc.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 879e2b3..13e1648 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -883,7 +883,7 @@ struct snd_soc_platform_driver { struct snd_soc_dai_link_component { const char *name; - const struct device_node *of_node; + struct device_node *of_node; const char *dai_name; }; @@ -985,7 +985,7 @@ struct snd_soc_codec_conf { * DT/OF node, but not both. */ const char *dev_name; - const struct device_node *of_node; + struct device_node *of_node; /* * optional map of kcontrol, widget and path name prefixes that are @@ -1002,7 +1002,7 @@ struct snd_soc_aux_dev { * DT/OF node, but not both. */ const char *codec_name; - const struct device_node *codec_of_node; + struct device_node *codec_of_node; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_component *component); -- 2.1.3

1 0

[alsa-devel] [PATCH] ALSA: hda - Add AZX_DCAPS_SNOOP_OFF (and refactor snoop setup)
by Takashi Iwai 25 Nov '14

25 Nov '14

Add a new driver_caps bit, AZX_DCAPS_SNOOP_OFF, to set the snoop off as default. This new bit is used for the checks in azx_check_snoop_available(). Most of case-switches are replaced with the new dcaps in each entry. While working on it, for avoiding to spend more bits, combine three bits AZX_DCAPS_SNOOP_SCH, AZX_DCAPS_SNOOP_ATI and AZX_DCAPS_SNOOP_NVIDIA bits into a flat type of two bits. This reduces the bits usages, and assign AZX_DCAPS_OFF to this empty bit now. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 71 +++++++++++++++++++++++++---------------------- sound/pci/hda/hda_priv.h | 12 ++++++-- 2 files changed, 47 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91fa959d05fe..b1b4a72cf29c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -271,42 +271,51 @@ enum { AZX_NUM_DRIVERS, /* keep this as last entry */ }; +#define azx_get_snoop_type(chip) \ + (((chip)->driver_caps & AZX_DCAPS_SNOOP_MASK) >> 10) +#define AZX_DCAPS_SNOOP_TYPE(type) ((AZX_SNOOP_TYPE_ ## type) << 10) + /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_NOPM \ - (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_REVERSE_ASSIGN) + (AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ + AZX_DCAPS_REVERSE_ASSIGN | AZX_DCAPS_SNOOP_TYPE(SCH)) #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) + (AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ + AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ + AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ - (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \ - AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_PM_RUNTIME | \ - AZX_DCAPS_I915_POWERWELL) + (AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ + AZX_DCAPS_SNOOP_TYPE(SCH)) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ - (AZX_DCAPS_ATI_SNOOP | AZX_DCAPS_NO_TCSEL | \ - AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB) + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_SNOOP_TYPE(ATI)) /* quirks for ATI/AMD HDMI */ #define AZX_DCAPS_PRESET_ATI_HDMI \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB) +/* quirks for ATI HDMI with snoop off */ +#define AZX_DCAPS_PRESET_ATI_HDMI_NS \ + (AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF) + /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ - (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ - AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT |\ - AZX_DCAPS_CORBRP_SELF_CLEAR) + (AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI | AZX_DCAPS_ALIGN_BUFSIZE |\ + AZX_DCAPS_NO_64BIT | AZX_DCAPS_CORBRP_SELF_CLEAR |\ + AZX_DCAPS_SNOOP_TYPE(NVIDIA)) #define AZX_DCAPS_PRESET_CTHDA \ - (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY) + (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF) /* * VGA-switcher support @@ -435,6 +444,8 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg, static void azx_init_pci(struct azx *chip) { + int snoop_type = azx_get_snoop_type(chip); + /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44) * TCSEL == Traffic Class Select Register, which sets PCI express QOS * Ensuring these bits are 0 clears playback static on some HD Audio @@ -449,7 +460,7 @@ static void azx_init_pci(struct azx *chip) /* For ATI SB450/600/700/800/900 and AMD Hudson azalia HD audio, * we need to enable snoop. */ - if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_ATI) { dev_dbg(chip->card->dev, "Setting ATI snoop: %d\n", azx_snoop(chip)); update_pci_byte(chip->pci, @@ -458,7 +469,7 @@ static void azx_init_pci(struct azx *chip) } /* For NVIDIA HDA, enable snoop */ - if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_NVIDIA) { dev_dbg(chip->card->dev, "Setting Nvidia snoop: %d\n", azx_snoop(chip)); update_pci_byte(chip->pci, @@ -473,7 +484,7 @@ static void azx_init_pci(struct azx *chip) } /* Enable SCH/PCH snoop if needed */ - if (chip->driver_caps & AZX_DCAPS_SCH_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_SCH) { unsigned short snoop; pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); if ((!azx_snoop(chip) && !(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)) || @@ -1360,8 +1371,8 @@ static void azx_check_snoop_available(struct azx *chip) { bool snoop = chip->snoop; - switch (chip->driver_type) { - case AZX_DRIVER_VIA: + if (azx_get_snoop_type(chip) == AZX_SNOOP_TYPE_NONE && + chip->driver_type == AZX_DRIVER_VIA) { /* force to non-snoop mode for a new VIA controller * when BIOS is set */ @@ -1371,17 +1382,11 @@ static void azx_check_snoop_available(struct azx *chip) if (!(val & 0x80) && chip->pci->revision == 0x30) snoop = false; } - break; - case AZX_DRIVER_ATIHDMI_NS: - /* new ATI HDMI requires non-snoop */ - snoop = false; - break; - case AZX_DRIVER_CTHDA: - case AZX_DRIVER_CMEDIA: - snoop = false; - break; } + if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF) + snoop = false; + if (snoop != chip->snoop) { dev_info(chip->card->dev, "Force to %s mode\n", snoop ? "snoop" : "non-snoop"); @@ -2112,13 +2117,13 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xaa98), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x9902), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaaa0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaaa8), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaab0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, @@ -2165,7 +2170,7 @@ static const struct pci_device_id azx_ids[] = { /* CM8888 */ { PCI_DEVICE(0x13f6, 0x5011), .driver_data = AZX_DRIVER_CMEDIA | - AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB }, + AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_SNOOP_OFF }, /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* VMware HDAudio */ diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index 949cd437eeb2..602536c2147d 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -152,9 +152,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* bits 0-7 are used for indicating driver type */ #define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ #define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ -#define AZX_DCAPS_ATI_SNOOP (1 << 10) /* ATI snoop enable */ -#define AZX_DCAPS_NVIDIA_SNOOP (1 << 11) /* Nvidia snoop enable */ -#define AZX_DCAPS_SCH_SNOOP (1 << 12) /* SCH/PCH snoop enable */ +#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ +#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ #define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ #define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ @@ -172,6 +171,13 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ +enum { + AZX_SNOOP_TYPE_NONE , + AZX_SNOOP_TYPE_SCH, + AZX_SNOOP_TYPE_ATI, + AZX_SNOOP_TYPE_NVIDIA, +}; + /* HD Audio class code */ #define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 -- 2.1.3

1 0

Re: [alsa-devel] ASoC: Intel: sst: Add IPC handling
by Dan Carpenter 25 Nov '14

25 Nov '14

Hello Vinod Koul, The patch ea12aa4acd70: "ASoC: Intel: sst: Add IPC handling" from Oct 16, 2014, leads to the following static checker warning: sound/soc/intel/sst/sst_ipc.c:340 sst_process_reply_mrfld() warn: we tested 'drv_id' before and it was 'true' sound/soc/intel/sst/sst_ipc.c 330 drv_id = msg_high.part.drv_id; 331 332 /* Check for async messages first */ 333 if (drv_id == SST_ASYNC_DRV_ID) { ^^^^^^^^^^^^^^^^ This is zero. 334 /*FW sent async large message*/ 335 process_fw_async_msg(sst_drv_ctx, msg); 336 return; 337 } 338 339 /* FW sent short error response for an IPC */ 340 if (msg_high.part.result && drv_id && !msg_high.part.large) { ^^^^^^ So this is non-zero. The thinking behind these warnings is that maybe a different test was intended. What does a non-zero drv_id mean which is different from checking SST_ASYNC_DRV_ID? The intent is not clear. In other words, would it be more readable to test "drv_id != SST_ASYNC_DRV_ID"? But in that case, we would just leave the test out since we obviously tested for that earlier. 341 /* 32-bit FW error code in msg_low */ 342 dev_err(sst_drv_ctx->dev, "FW sent error response 0x%x", msg_low); 343 sst_wake_up_block(sst_drv_ctx, msg_high.part.result, 344 msg_high.part.drv_id, 345 msg_high.part.msg_id, NULL, 0); 346 return; 347 } 348 regards, dan carpenter

1 0

[alsa-devel] [PATCH] ASoC: rockchip: i2s: Add dependency to I2S device driver
by Andreas Ruprecht 24 Nov '14

24 Nov '14

Currently, CONFIG_SND_SOC_ROCKCHIP_I2S could also be selected without having CONFIG_SND_SOC_ROCKCHIP enabled. As this makes no sense (and breaks the build, when the option CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM - which would be selected by CONFIG_SND_SOC_ROCKCHIP - is not enabled), a Kconfig dependency is added to CONFIG_SND_SOC_ROCKCHIP_I2S. This will make the item visible only if CONFIG_SND_SOC_ROCKCHIP is enabled. Signed-off-by: Andreas Ruprecht <rupran(a)einserver.de> Reported-by: Jim Davis <jim.epost(a)gmail.com> --- sound/soc/rockchip/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index b1fc0ca..74380b4 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -9,7 +9,7 @@ config SND_SOC_ROCKCHIP config SND_SOC_ROCKCHIP_I2S tristate "Rockchip I2S Device Driver" - depends on CLKDEV_LOOKUP + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP help Say Y or M if you want to add support for I2S driver for Rockchip I2S device. The device supports upto maximum of -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Fix off by one error in the loop stucture.
by Jyri Sarha 24 Nov '14

24 Nov '14

Fix off by one read beyond the end of a table. Reported-by: David Binderman <dcb314(a)hotmail.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 145fe5b..93de5dd 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -911,12 +911,13 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, } aic31xx->p_div = i; - for (i = 0; aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) { - if (i == ARRAY_SIZE(aic31xx_divs)) { - dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", - __func__, freq); - return -EINVAL; - } + for (i = 0; i < ARRAY_SIZE(aic31xx_divs) && + aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) + ; + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; } /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH 01/32] ASoC: wm8350: Cleanup manual bias level transitions
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8350.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 628ec77..87f664b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1242,19 +1242,6 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8350_suspend(struct snd_soc_codec *codec) -{ - wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8350_resume(struct snd_soc_codec *codec) -{ - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) @@ -1565,9 +1552,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); - - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } @@ -1596,8 +1580,6 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) * wait for its completion */ flush_delayed_work(&codec->dapm.delayed_work); - wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); - wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); return 0; @@ -1613,10 +1595,9 @@ static struct regmap *wm8350_get_regmap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .probe = wm8350_codec_probe, .remove = wm8350_codec_remove, - .suspend = wm8350_suspend, - .resume = wm8350_resume, .get_regmap = wm8350_get_regmap, .set_bias_level = wm8350_set_bias_level, + .suspend_bias_off = true, .controls = wm8350_snd_controls, .num_controls = ARRAY_SIZE(wm8350_snd_controls), -- 1.8.0

3 64

[alsa-devel] [PATCH 1/3] ASoC: alc5623: Cleanup bias level transitions
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Also remove the manual sequencing back to SND_SOC_BIAS_ON in resume as this is already handled by the ASoC core. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/alc5623.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 9d0755a..bdf8c5a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -866,7 +866,6 @@ static int alc5623_suspend(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(alc5623->regmap, true); return 0; @@ -887,15 +886,6 @@ static int alc5623_resume(struct snd_soc_codec *codec) return ret; } - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* charge alc5623 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - alc5623_set_bias_level(codec, codec->dapm.bias_level); - } - return 0; } @@ -906,9 +896,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) alc5623_reset(codec); - /* power on device */ - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (alc5623->add_ctrl) { snd_soc_write(codec, ALC5623_ADD_CTRL_REG, alc5623->add_ctrl); @@ -964,19 +951,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5623_remove(struct snd_soc_codec *codec) -{ - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .probe = alc5623_probe, - .remove = alc5623_remove, .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config alc5623_regmap = { -- 1.8.0

2 5

[alsa-devel] [PATCH 1/3] ASoC: uda134x: Remove is_powered_on_standby from platform data
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

According to its documentation the is_powered_on_standby field of the uda134x platform data is supposed to prevent the the driver from shutting down the ADC and DAC in standby mode. This behavior was broken in commit commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") almost 5 years ago and all the flag does now is cause the driver to go to SND_SOC_BIAS_ON in probe, just for the ASoC core to put it back into SND_SOC_BIAS_STANDBY right after probe. Apparently the intended behavior has not been missed, so just remove is_powered_on_standby from the platform data struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/uda134x.h | 12 ------------ sound/soc/codecs/uda134x.c | 5 +---- 2 files changed, 1 insertion(+), 16 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index e475659..509efb0 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,18 +18,6 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; - /* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you unset is_powered_on_standby the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - set is_powered_on_standby, you will have slightly higher power - consumption. Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 32b2f78..54240f1 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -518,10 +518,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) uda134x_reset(codec); - if (pd->is_powered_on_standby) - uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); - else - uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (pd->model == UDA134X_UDA1341) { widgets = uda1341_dapm_widgets; -- 1.8.0

2 3

[alsa-devel] [PATCH 1/3] ASoC: max98090: Fix digital microphone
by Jarkko Nikula 24 Nov '14

24 Nov '14

Commit e409dfbfccf9 ("ASoC: dapm: Add a few supply widget sanity checks") broke digital microphone support in max98090.c: max98090 i2c-193C9890:00: Conditional paths are not supported for supply widgets (DMICL_ENA -> [DMIC] -> DMIC Mux) max98090 i2c-193C9890:00: ASoC: no dapm match for DMICL_ENA --> DMIC --> DMIC Mux max98090 i2c-193C9890:00: ASoC: Failed to add route DMICL_ENA -> DMIC -> DMIC Mux max98090 i2c-193C9890:00: Conditional paths are not supported for supply widgets (DMICR_ENA -> [DMIC] -> DMIC Mux) max98090 i2c-193C9890:00: ASoC: no dapm match for DMICR_ENA --> DMIC --> DMIC Mux max98090 i2c-193C9890:00: ASoC: Failed to add route DMICR_ENA -> DMIC -> DMIC Mux Problem is partially caused by commit f69e3caa9e18 ("ASoC: max98090: Enable both DMIC channels also when using mono configuration") which connects "DMICL_ENA" and "DMICR_ENA" supply widgets to "DMIC Mux". Fix the breakage by reverting f69e3caa9e18 and then by adding additional "DMICR_ENA" to "DMICL" and "DMICL_ENA" to "DMICR" cross-connections. This disconnects these supply widgets from the mux and makes sure that both DMIC data channels are still enabled together. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98090.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 2ad381c4ec57..19258a910974 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1311,6 +1311,10 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICL", NULL, "DMICR_ENA"}, + {"DMICR", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, {"DMICL", NULL, "AHPF"}, {"DMICR", NULL, "AHPF"}, @@ -1368,8 +1372,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"DMIC Mux", "ADC", "ADCR"}, {"DMIC Mux", "DMIC", "DMICL"}, {"DMIC Mux", "DMIC", "DMICR"}, - {"DMIC Mux", "DMIC", "DMICL_ENA"}, - {"DMIC Mux", "DMIC", "DMICR_ENA"}, {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, -- 2.1.3

2 5

[alsa-devel] [PATCH] ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed
by Oder Chiou 24 Nov '14

24 Nov '14

After the commit "19ba484d7b15c8650b30377aad6e65b34d3cf3d5", it will cause that the regmap_range "rt5677_ranges" cannot be accessed. The patch fixes this issue. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 37 ++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 2 +- 2 files changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2211f1..133010d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4287,6 +4287,7 @@ static int rt5677_probe(struct snd_soc_codec *codec) } mutex_init(&rt5677->dsp_cmd_lock); + mutex_init(&rt5677->dsp_pri_lock); return 0; } @@ -4344,10 +4345,19 @@ static int rt5677_read(void *context, unsigned int reg, unsigned int *val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_read(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + } + } else { regmap_read(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4357,10 +4367,20 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_write(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA, + val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + } + } else { regmap_write(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4495,10 +4515,13 @@ static const struct regmap_config rt5677_regmap_physical = { .reg_bits = 8, .val_bits = 16, - .max_register = RT5677_VENDOR_ID2 + 1, + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), .readable_reg = rt5677_readable_register, .cache_type = REGCACHE_NONE, + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), }; static const struct regmap_config rt5677_regmap = { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a02f64c..dbd9ffd 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1670,7 +1670,7 @@ struct rt5677_priv { struct rt5677_platform_data pdata; struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; - struct mutex dsp_cmd_lock; + struct mutex dsp_cmd_lock, dsp_pri_lock; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: Arndale: Add machine device tree binding documentation for Arndale board
by Krishna Mohan Dani 24 Nov '14

24 Nov '14

Document the device tree binding for the Arndale board based machine driver. Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- .../devicetree/bindings/sound/arndale.txt | 24 ++++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt new file mode 100644 index 0000000..0e76946 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arndale.txt @@ -0,0 +1,24 @@ +Audio Binding for Arndale boards + +Required properties: +- compatible : Can be the following, + "samsung,arndale-rt5631" + +- samsung,audio-cpu: The phandle of the Samsung I2S controller +- samsung,audio-codec: The phandle of the audio codec + +Optional: +- samsung,model: The name of the sound-card + +Arndale Boards has many audio daughter cards, one of them is +rt5631/alc5631. Below example shows audio bindings for rt5631/ +alc5631 based codec. + +Example: + +sound { + compatible = "samsung,arndale-rt5631"; + + samsung,audio-cpu = <&i2s0> + samsung,audio-codec = <&rt5631>; +}; -- 1.7.9.5

2 1

[alsa-devel] TLV320AIC3104 Sample rate problem
by Catalin Crenguta 24 Nov '14

24 Nov '14

Hello, I'm using an Audio Cape Rev. B connected to BeagleBone Black (MCLK=12MHz) and I'm trying to record audio at sample rate 8KHz. Recording and playback using ALSA is OK, but when playing the recorded samples on PC you can hear that the actual sample rate is lower (the pitch shifted to a lower tone). I have added debug messages throughout the alsa driver tlv320aic3x.c, and I found out that the PLL parameters are computed correctly as in TLV320AIC3104 Datasheet Table found at page 27. (p: 1 j: 8 d: 1920 r: 1). Has anyone else encountered a problem like this? Can someone with more experience check the driver? Kernel: https://github.com/beagleboard/linux , commit 2bcc89c, 3.14.19+. Best regards, Catalin Crenguta

1 0

[alsa-devel] [PATCH 00/29 v2] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 24 Nov '14

24 Nov '14

This patchset is to enable ALSA Dice driver to handle more models, to improve ALSA firewire-speakers driver as OXFW driver. This patchset is for linux-next branch and all of my work for Linux 3.19. == Features: Currently, ALSA Dice driver is limited for playback-only devices. The aim of this patchset is to add supports for: - Full-duplex streams with synchronization - Both PCM capture and playback - Both MIDI capture and playback Currently, ALSA firewire-speakers driver is only for two playback-only models. And this driver has a bug for one of them. The aim of this patchset is to fix the bug and to add supports for more models with: - Both PCM capture and playback - Both MIDI capture and playback The renew drivers doesn't support: - Synchronization between multiple devices on the same bus For this purpose, please join in user-space driver project (FFADO) - Any control functionality for internal DSP For this purpose, please use ffado-dbus-server/ffado-mixer, or write your own application == Updates from my previous post: [alsa-devel] [PATCH 00/14] ALSA: Dice enhancement http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/081890.ht… - Fix typo - Minor code change [alsa-devel] [RFC][PATCH 00/15 v4] OXFW driver, a succesor to firewire-speakers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/076581.html - Support more Mackie (Loud) models by name detection - Fix a bug to handle FireWave - Use AV/C Stream Format Information command to set sampling rate == Remarks: - When the Dice driver is loaded, user-space driver fails to register notification address, thus cannot handle device for streaming. See: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081731.… - OXFW chipset seems to transmit wrong values for timestamp, thus the driver don't apply full-duplex streams with synchronization == Supported Dice-based models (as long as I know): TC Electronic - Desktop Konnekt 6 - Impact Twin - Konnekt 8/24D/Live - Studio Konnekt 48 PreSonus - FireStudio 26x26/Mobile/Project/Tube/LightPipe - StudioLive 1602/1642/2442 - FireStation Mackie - Onyx Blackbird - Onyx i-series (latter models) Focusrite - Saffire Pro 14/24/24 DSP/26/40/56 M-Audio - Profire 610/2626 Alessis - IO 24/26 - MultiMix 12/16 Weiss - ADC2 - Vesta - Mineva - AFI 1 - TAG DAC 1 - INT 202 - DAC 202 - Flex 5000 DnR - Axus FireWire I/O card 16x16 Lexicon - I-ONIX FW 810S Allen and Heath - Zed R16 Midas - Venice F32 == Supported OXFW-based models (as long as I know): Griffin: - FireWave LaCie: - Firewire Speakers Behringer: - F-Control Audio 202 Mackie (Loud): - Onyx-i series (former models) - Onyx Satellite - Tapco Link.Firewire - d.2 pro/d.4 pro - U.420/U.420d Takashi Sakamoto (29): ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc interface into a file ALSA: dice: Add new functions for constraints of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: dice: remove experimental state ALSA: speakers: Rename to oxfw and rename some members ALSA: oxfw: Move to its own directory ALSA: oxfw: Split stream functionality to a new file and add a header file ALSA: oxfw: Split PCM functionality to a new file ALSA: oxfw: Split control functionality to a new file ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 26 +- sound/firewire/Makefile | 6 +- sound/firewire/dice.c | 1511 ---------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/{ => dice}/dice-interface.h | 0 sound/firewire/dice/dice-midi.c | 176 ++++ sound/firewire/dice/dice-pcm.c | 404 ++++++++ sound/firewire/dice/dice-proc.c | 252 +++++ sound/firewire/dice/dice-stream.c | 423 ++++++++ sound/firewire/dice/dice-transaction.c | 382 +++++++ sound/firewire/dice/dice.c | 359 +++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw-command.c | 153 +++ sound/firewire/oxfw/oxfw-control.c | 283 ++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++ sound/firewire/oxfw/oxfw-midi.c | 189 ++++ sound/firewire/oxfw/oxfw-pcm.c | 399 ++++++++ sound/firewire/oxfw/oxfw-proc.c | 113 +++ sound/firewire/oxfw/oxfw-stream.c | 702 +++++++++++++ sound/firewire/oxfw/oxfw.c | 310 ++++++ sound/firewire/oxfw/oxfw.h | 145 +++ sound/firewire/speakers.c | 792 --------------- 26 files changed, 4887 insertions(+), 2319 deletions(-) delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c rename sound/firewire/{ => dice}/dice-interface.h (100%) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-control.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-pcm.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c create mode 100644 sound/firewire/oxfw/oxfw-stream.c create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h delete mode 100644 sound/firewire/speakers.c -- 1.9.1

4 42

[alsa-devel] [PATCH] ASoC: bt-sco: support wide band speech
by Barry Song 24 Nov '14

24 Nov '14

From: Yibo Cai <yibo.cai(a)csr.com> HFP1.6 requires support for wide band speech(16KHz) over Bluetooth. Signed-off-by: Yibo Cai <yibo.cai(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/bt-sco.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index c4cf069..fc0664b 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -31,14 +31,14 @@ static struct snd_soc_dai_driver bt_sco_dai = { .stream_name = "Playback", .channels_min = 1, .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -- 2.1.1

2 3

[alsa-devel] [PATCH] ASoC: rt5631: Fixing compilation warning when DT is disabled
by Krishna Mohan Dani 24 Nov '14

24 Nov '14

Fixes the following compilation warning: Warning: 'rt5631_i2c_dt_ids' defined but not used - when DT is not used. Signed-off-by: Claude Youn <claude.youn(a)gmail.com> Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- sound/soc/codecs/rt5631.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3b7d5e4..9425545 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1691,12 +1691,14 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); +#ifdef CONFIG_OF static struct of_device_id rt5631_i2c_dt_ids[] = { { .compatible = "realtek,rt5631"}, { .compatible = "realtek,alc5631"}, { } }; MODULE_DEVICE_TABLE(of, rt5631_i2c_dt_ids); +#endif static const struct regmap_config rt5631_regmap_config = { .reg_bits = 8, -- 1.7.9.5

4 4

[alsa-devel] [PATCH 0/4 V3] Arndale: Adding Sound Machine Driver for ALC5631 Codec
by Krishna Mohan Dani 24 Nov '14

24 Nov '14

These patches add machine driver to instantiate I2S based realtek's ALC5631 based sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec based card hence support for ALC5631 based machine driver is being added. Most of the code has been pulled from insignal's git for Arndale and modified to suit the maintain syntax. Machine driver has been made DT compatible. Changes done: - Removed ifdef for of_device_id structure as commented by Mark Brown - Supporting both variants of device name:alc5631 and rt5631 as commented by Mark Brown - Added "return -EINVAL" if cpu_of_node or codec_of_node is not found in dtb, as commented by Anil Kumar - Removed i2s specific changes - Added signed off of claude - Removed CONFIG_OF check from rt5631.c codec driver and other changes Krishna Mohan Dani (4): ASoC: ALC5631/RT5631: Add device tree binding documentation ASoC: Samsung: Add arndale_rt5631 machine driver Sound: Kconfig: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631/RT5631 codec driver Documentation/devicetree/bindings/sound/rt5631.txt | 48 +++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/rt5631.c | 9 ++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 150 ++++++++++++++++++++ 6 files changed, 217 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c -- 1.7.9.5

3 9

[alsa-devel] [PATCH] ASoC: soc-pcm: skip dpcm path checking with incapable/unready FE
by Qiao Zhou 24 Nov '14

24 Nov '14

Skip dpcm path checking for playback or capture, if corresponding FE doesn't support playback or capture, or currently is not ready. It can reduce the unnecessary cost to search connected widgets. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311a..3b068d4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2248,7 +2248,13 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) fe->dai_link->name); /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min) + if (!fe->cpu_dai->driver->playback.channels_min + || !fe->codec_dai->driver->playback.channels_min) + goto capture; + + /* skip if FE doesn't do playback */ + if (!fe->cpu_dai->playback_active + || !fe->codec_dai->playback_active) goto capture; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); @@ -2278,7 +2284,13 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min) + if (!fe->cpu_dai->driver->capture.channels_min + || !fe->codec_dai->driver->capture.channels_min) + continue; + + /* skip if FE doesn't do capture */ + if (!fe->cpu_dai->capture_active + || !fe->codec_dai->capture_active) continue; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); -- 1.7.9.5

2 2

[alsa-devel] Debian Bug#770122: patch as workaround/solution
by Elimar Riesebieter 23 Nov '14

23 Nov '14

Hi, Iam forwarding this Debian bug to alsa-devel due to a not maintained BugTracker at alsa project. Could one please have a look at it? Thanks Elimar -- Alles was viel bedacht wird ist bedenklich!;-) Friedrich Nietzsche

1 1

[alsa-devel] [PATCH 0/10] fix error return code
by Julia Lawall 23 Nov '14

23 Nov '14

These patches fix cases where the return code appears to be unintentially nonnegative. The complete semantic match that finds the problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ f(...) { <+... ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ...+> } @r exists@ identifier ret,l,ok.f; expression e1,e2,e3; statement S; position p1,p2,p3; @@ f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(...) { ... when != ret = e2 when forall return@p3 ret; } ) ... when any } @bad exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any ret = e1 ... when any if@p2(...) S2 @bad2@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; @@ ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when any if@p2(...) S2 @script:python depends on !bad && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

2 12

[alsa-devel] Debian Bug#770122: patch as workaround/solution
by Elimar Riesebieter 23 Nov '14

23 Nov '14

Hi, I am forwarding this Debian bug [0] to alsa-devel due to a not maintained BugTracker at alsa project. Could one please have a look at it? [0] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=770122 Thanks Elimar -- Alles was viel bedacht wird ist bedenklich!;-) Friedrich Nietzsche

1 0

[alsa-devel] [PATCH 1/2] amixer: Make "dB" case-insensitive in set commands
by Takashi Iwai 23 Nov '14

23 Nov '14

We don't have to be necessarily too strict about case-sensitivity of "dB" suffix used in set commands. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index cf82892bfa2f..6a2fdb96c62f 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -346,7 +346,7 @@ static int set_volume_simple(snd_mixer_elem_t *elem, if (*p == '%') { percent = 1; p++; - } else if (p[0] == 'd' && p[1] == 'B') { + } else if (toupper(p[0]) == 'D' && toupper(p[1]) == 'B') { vol_type = VOL_DB; p += 2; scale = 100; -- 2.1.3

1 1

[alsa-devel] [PATCH 00/10] ALSA: asihpi: updates
by eliot＠blennerhassett.gen.nz 23 Nov '14

23 Nov '14

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Major change - Add support for interrupt driven 'low latency' mode PCM streams on some AudioScience adapters. Various other minor updates from AudioScience Eliot Blennerhassett (10): ALSA: asihpi: Minor string and dead code cleanup ALSA: asihpi: New I/O types - AVB & BLUlink, DAB Rf receiver ALSA: asihpi: Logging format improvements ALSA: asihpi: Use CONFIG_64BIT directly ALSA: asihpi: Refactor control cache code. ALSA: asihpi: Add support for stream interrupt. ALSA: asihpi: Turn off msg/resp logging after DSP has crashed. ALSA: asihpi: Use standard printk helpers ALSA: asihpi: don't fail probe if adapter mode read fails ALSA: asihpi: used parts of message/response are zeroed before use sound/pci/asihpi/asihpi.c | 326 +++++++++++++++++++++++++++------------- sound/pci/asihpi/hpi.h | 16 +- sound/pci/asihpi/hpi6205.c | 43 +++++- sound/pci/asihpi/hpi_internal.h | 20 ++- sound/pci/asihpi/hpicmn.c | 107 +++++++------ sound/pci/asihpi/hpicmn.h | 19 ++- sound/pci/asihpi/hpimsginit.c | 30 ++-- sound/pci/asihpi/hpimsgx.c | 15 +- sound/pci/asihpi/hpioctl.c | 123 +++++++++++++-- sound/pci/asihpi/hpios.h | 8 +- 10 files changed, 508 insertions(+), 199 deletions(-) -- 1.9.1

3 12

[alsa-devel] randconfig errors in ts3a227e.c
by devendra.aaru 23 Nov '14

23 Nov '14

Hello All, While testing the build, i have got the following errors: sound/soc/codecs/ts3a227e.c: In function ‘ts3a227e_enable_jack_detect’: sound/soc/codecs/ts3a227e.c:223:2: error: implicit declaration of function ‘snd_jack_set_key’ [-Werror=implicit-function-declaration] snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); ^ cc1: some warnings being treated as errors make[3]: *** [sound/soc/codecs/ts3a227e.o] Error 1 The CONFIG_SND_JACK is not enabled in the .config and the #else portion of include/sound/jack.h doesn't contain snd_jack_set_key stub. This resulting in the build error. I am unsure if adding stub in include/sound/jack.h or adding a Kconfig knob to deselect the ts3a227e.c fixes the issue ? full randconfig attached. (sorry about the 55k file size)

1 1

[alsa-devel] [patch] sound: oss: uart401: remove unneeded NULL check
by Dan Carpenter 22 Nov '14

22 Nov '14

"devc" can't be NULL here so there is no need to check. Also I removed the "devc = NULL" assignment because devc is stored on stack so it's a no-op. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index 279bc56..dae4d43 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -412,13 +412,10 @@ void unload_uart401(struct address_info *hw_config) if (!devc->share_irq) free_irq(devc->irq, devc); - if (devc) - { - kfree(midi_devs[devc->my_dev]->converter); - kfree(midi_devs[devc->my_dev]); - kfree(devc); - devc = NULL; - } + kfree(midi_devs[devc->my_dev]->converter); + kfree(midi_devs[devc->my_dev]); + kfree(devc); + /* This kills midi_devs[x] */ sound_unload_mididev(hw_config->slots[4]); }

2 1

[alsa-devel] Headset jack and intel based Macmini6,2
by Marco 22 Nov '14

22 Nov '14

/cc David Henningsson that has done something like that for some Dell HW and might have some suggestions Hello, even if poorly advertised, recent mac.* have an headset jack (amplified) along with the non amplified line jack. For example, for MacMini, from: apple.com/mac-mini/specs/ > Support for Apple iPhone headset with microphone Where iPhone headset means a simple headset not preamplified, really. I own a MacMini6,2 (the october 2012 one) and with Mac OSX it works... under linux... well... it is a simple out jack. This jeopardize the ability to use any video conference software (who doesn't today?) on linux if you don't own a USB audio device (or you don't want that extra piece of relatively large hw) The card is a HDA based Cirrus Logic CS4206/4207 and the two specific models do not work (imac27 and mbp55) while auto works (except the headset of course :) I tried to change the pinconfig but on the intel HDA documentation it is not really clear how the code for an headset jack (in/out) would look like. I saw that for card ALC269/270/275/276/28x/29x there are models that support the headset codes but... well.. they don't really seem "standard" intel codes I tried to look if it was possible to extract these codes from OSX but I haven't found anything interesting (but I might have missed something) Any suggestion dear esteemed Alsa developers? :) Any help appreciated, Thanks! Marco

1 0

[alsa-devel] [PATCH] ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672
by mengdong.lin＠intel.com 21 Nov '14

21 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5672 codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..7479ce0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -16,6 +16,9 @@ config SND_SST_MFLD_PLATFORM config SND_SST_IPC tristate +config SND_SST_MACHINE + tristate + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI @@ -76,3 +79,20 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5672_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5670 + select SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_SST + select SND_SST_IPC + select SND_SST_MACHINE + default n + + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y if you have such a device + If unsure select "N". + diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..4069d3f 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -31,6 +31,7 @@ obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SST_MACHINE) += board/ # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/board/Makefile b/sound/soc/intel/board/Makefile new file mode 100644 index 0000000..9ecc227 --- /dev/null +++ b/sound/soc/intel/board/Makefile @@ -0,0 +1,2 @@ +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o diff --git a/sound/soc/intel/board/cht_bsw_rt5672.c b/sound/soc/intel/board/cht_bsw_rt5672.c new file mode 100644 index 0000000..dffd8b1 --- /dev/null +++ b/sound/soc/intel/board/cht_bsw_rt5672.c @@ -0,0 +1,286 @@ +/* + * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5672 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> + * Mengdong Lin <mengdong.lin(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt5670.h" +#include "../sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + unsigned int fmt; + + if (strncmp(codec_dai->name, "rt5670-aif1", 11)) + return 0; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, + SNDRV_PCM_FORMAT_GSM); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* TDM slave Mode */ + fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec DAI fmt %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + return 0; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int cht_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + break; + default: + dev_err(card->dev, "Invalid bias level=%d\n", level); + return -EINVAL; + } + card->dapm.bias_level = level; + return 0; +} + +static int cht_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + + /* Set card bias level */ + cht_set_bias_level(card, &card->dapm, SND_SOC_BIAS_OFF); + card->dapm.idle_bias_off = true; + + ret = snd_soc_dapm_sync(&card->dapm); + if (ret) { + dev_err(card->dev, "unable to sync dapm\n"); + return ret; + } + return ret; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .init = cht_init, + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5670-aif1", + .codec_name = "i2c-10EC5670:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailcraudio", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .set_bias_level = cht_set_bias_level, + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = snd_soc_register_card(&snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5672", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5672"); -- 1.9.1

4 24

[alsa-devel] [PATCH 1/1] ALSA: sb: Deletion of unnecessary checks before two function calls
by SF Markus Elfring 21 Nov '14

21 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 21 Nov 2014 19:32:02 +0100 The functions release_and_free_resource() and snd_util_memhdr_free() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/isa/sb/emu8000_synth.c | 3 +-- sound/isa/sb/sb_common.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c index 4e3fcfb..95b39be 100644 --- a/sound/isa/sb/emu8000_synth.c +++ b/sound/isa/sb/emu8000_synth.c @@ -105,8 +105,7 @@ static int snd_emu8000_delete_device(struct snd_seq_device *dev) snd_device_free(dev->card, hw->pcm); if (hw->emu) snd_emux_free(hw->emu); - if (hw->memhdr) - snd_util_memhdr_free(hw->memhdr); + snd_util_memhdr_free(hw->memhdr); hw->emu = NULL; hw->memhdr = NULL; return 0; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3ef9906..f22b448 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -184,8 +184,7 @@ static int snd_sbdsp_probe(struct snd_sb * chip) static int snd_sbdsp_free(struct snd_sb *chip) { - if (chip->res_port) - release_and_free_resource(chip->res_port); + release_and_free_resource(chip->res_port); if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); #ifdef CONFIG_ISA -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: es1688_lib: Deletion of an unnecessary check before the function call "release_and_free_resource"
by SF Markus Elfring 21 Nov '14

21 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 21 Nov 2014 19:05:50 +0100 The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/isa/es1688/es1688_lib.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index b3b4f15..fb259491 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -614,8 +614,7 @@ static int snd_es1688_free(struct snd_es1688 *chip) { if (chip->hardware != ES1688_HW_UNDEF) snd_es1688_init(chip, 0); - if (chip->res_port) - release_and_free_resource(chip->res_port); + release_and_free_resource(chip->res_port); if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); if (chip->dma8 >= 0) { -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: core: Deletion of unnecessary checks before two function calls
by SF Markus Elfring 21 Nov '14

21 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 21 Nov 2014 18:34:48 +0100 The functions snd_seq_oss_timer_delete() and vunmap() perform also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/core/seq/oss/seq_oss_init.c | 9 +++------ sound/core/sgbuf.c | 3 +-- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b9184d2..b0e32e1 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -403,14 +403,11 @@ free_devinfo(void *private) { struct seq_oss_devinfo *dp = (struct seq_oss_devinfo *)private; - if (dp->timer) - snd_seq_oss_timer_delete(dp->timer); + snd_seq_oss_timer_delete(dp->timer); - if (dp->writeq) - snd_seq_oss_writeq_delete(dp->writeq); + snd_seq_oss_writeq_delete(dp->writeq); - if (dp->readq) - snd_seq_oss_readq_delete(dp->readq); + snd_seq_oss_readq_delete(dp->readq); kfree(dp); } diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 0a41850..84fffab 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -39,8 +39,7 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) if (! sgbuf) return -EINVAL; - if (dmab->area) - vunmap(dmab->area); + vunmap(dmab->area); dmab->area = NULL; tmpb.dev.type = SNDRV_DMA_TYPE_DEV; -- 2.1.3

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs42l51: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs42l51.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 669c38f..b395152 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -153,15 +153,17 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + switch (event) { case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + snd_soc_update_bits(codec, CS42L51_POWER_CTL1, CS42L51_POWER_CTL1_PDN, CS42L51_POWER_CTL1_PDN); break; default: case SND_SOC_DAPM_POST_PMD: - snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + snd_soc_update_bits(codec, CS42L51_POWER_CTL1, CS42L51_POWER_CTL1_PDN, 0); break; } -- 1.8.0

3 4

[alsa-devel] [PATCH v3] ASoC: wm8960: Add device tree support
by Zidan Wang 21 Nov '14

21 Nov '14

Document the device tree binding for the WM8960 codec, and modify the driver to extract the platform data from device tree, if present. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 31 ++++++++++++++++ sound/soc/codecs/wm8960.c | 41 ++++++++++++++++------ 2 files changed, 62 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt new file mode 100644 index 0000000..2deb8a3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -0,0 +1,31 @@ +WM8960 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8960" + + - reg : the I2C address of the device. + +Optional properties: + - wlf,shared-lrclk: This is a boolean property. If present, the LRCM bit of + R24 (Additional control 2) gets set, indicating that ADCLRC and DACLRC pins + will be disabled only when ADC (Left and Right) and DAC (Left and Right) + are disabled. + When wm8960 works on synchronize mode and DACLRC pin is used to supply + frame clock, it will no frame clock for captrue unless enable DAC to enable + DACLRC pin. If shared-lrclk is present, no need to enable DAC for captrue. + + - wlf,capless: This is a boolean property. If present, OUT3 pin will be + enabled and disabled together with HP_L and HP_R pins in response to jack + detect events. + +Example: + +codec: wm8960@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + + wlf,shared-lrclk; +}; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4dc4e85..99d6457 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -125,6 +125,7 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + struct wm8960_data pdata; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) @@ -440,8 +441,8 @@ static const struct snd_soc_dapm_route audio_paths_capless[] = { static int wm8960_add_widgets(struct snd_soc_codec *codec) { - struct wm8960_data *pdata = codec->dev->platform_data; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + struct wm8960_data *pdata = &wm8960->pdata; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; @@ -961,17 +962,13 @@ static int wm8960_resume(struct snd_soc_codec *codec) static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - struct wm8960_data *pdata = dev_get_platdata(codec->dev); + struct wm8960_data *pdata = &wm8960->pdata; int ret; - wm8960->set_bias_level = wm8960_set_bias_level_out3; - - if (!pdata) { - dev_warn(codec->dev, "No platform data supplied\n"); - } else { - if (pdata->capless) - wm8960->set_bias_level = wm8960_set_bias_level_capless; - } + if (pdata->capless) + wm8960->set_bias_level = wm8960_set_bias_level_capless; + else + wm8960->set_bias_level = wm8960_set_bias_level_out3; ret = wm8960_reset(codec); if (ret < 0) { @@ -1029,6 +1026,18 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; +static void wm8960_set_pdata_from_of(struct i2c_client *i2c, + struct wm8960_data *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + + if (of_property_read_bool(np, "wlf,capless")) + pdata->capless = true; + + if (of_property_read_bool(np, "wlf,shared-lrclk")) + pdata->shared_lrclk = true; +} + static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1045,6 +1054,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); + if (pdata) + memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); + else if (i2c->dev.of_node) + wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (pdata && pdata->shared_lrclk) { ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 0x4, 0x4); @@ -1075,10 +1089,17 @@ static const struct i2c_device_id wm8960_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); +static const struct of_device_id wm8960_of_match[] = { + { .compatible = "wlf,wm8960", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8960_of_match); + static struct i2c_driver wm8960_i2c_driver = { .driver = { .name = "wm8960", .owner = THIS_MODULE, + .of_match_table = wm8960_of_match, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: adau17x1: Mark DSP parameter memory as readable and precious
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

To be able to read back data from the DSP parameter memory the register range needs to be marked as readable. At the same time we do not want them to e.g. appear in debugfs output so mark them as precious as well. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adau1761.c | 1 + sound/soc/codecs/adau1781.c | 1 + sound/soc/codecs/adau17x1.c | 14 ++++++++++++++ sound/soc/codecs/adau17x1.h | 1 + 4 files changed, 17 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 109fd78..a1baeee 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -798,6 +798,7 @@ const struct regmap_config adau1761_regmap_config = { .num_reg_defaults = ARRAY_SIZE(adau1761_reg_defaults), .readable_reg = adau1761_readable_register, .volatile_reg = adau17x1_volatile_register, + .precious_reg = adau17x1_precious_register, .cache_type = REGCACHE_RBTREE, }; EXPORT_SYMBOL_GPL(adau1761_regmap_config); diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 17966b6..35581f4 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -472,6 +472,7 @@ const struct regmap_config adau1781_regmap_config = { .num_reg_defaults = ARRAY_SIZE(adau1781_reg_defaults), .readable_reg = adau1781_readable_register, .volatile_reg = adau17x1_volatile_register, + .precious_reg = adau17x1_precious_register, .cache_type = REGCACHE_RBTREE, }; EXPORT_SYMBOL_GPL(adau1781_regmap_config); diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index d49bc7e..fa2e690 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -707,8 +707,22 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(adau17x1_set_micbias_voltage); +bool adau17x1_precious_register(struct device *dev, unsigned int reg) +{ + /* SigmaDSP parameter memory */ + if (reg < 0x400) + return true; + + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_precious_register); + bool adau17x1_readable_register(struct device *dev, unsigned int reg) { + /* SigmaDSP parameter memory */ + if (reg < 0x400) + return true; + switch (reg) { case ADAU17X1_CLOCK_CONTROL: case ADAU17X1_PLL_CONTROL: diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 6861aa3..e13583e 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -56,6 +56,7 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); bool adau17x1_volatile_register(struct device *dev, unsigned int reg); +bool adau17x1_precious_register(struct device *dev, unsigned int reg); int adau17x1_resume(struct snd_soc_codec *codec); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -- 1.8.0

2 2

[alsa-devel] [PATCH 1/3] ASoC: max98088: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98088.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index bb892b3..805b3f8 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -875,7 +875,7 @@ static const struct snd_kcontrol_new max98088_right_ADC_mixer_controls[] = { static int max98088_mic_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -905,7 +905,7 @@ static int max98088_mic_event(struct snd_soc_dapm_widget *w, static int max98088_line_pga(struct snd_soc_dapm_widget *w, int event, int line, u8 channel) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); u8 *state; -- 1.8.0

2 5

[alsa-devel] [PATCH] ASoC: sirf-audio-codec: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. While we are at it also replace dev_get_drvdata() with snd_soc_codec_get_drvdata(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sirf-audio-codec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 06ba492..07eea20 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -120,7 +120,8 @@ static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, { #define ATLAS6_CODEC_ENABLE_BITS (1 << 29) #define ATLAS6_CODEC_RESET_BITS (1 << 28) - struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: enable_and_reset_codec(sirf_audio_codec->regmap, @@ -142,7 +143,8 @@ static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, { #define PRIMA2_CODEC_ENABLE_BITS (1 << 27) #define PRIMA2_CODEC_RESET_BITS (1 << 26) - struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: enable_and_reset_codec(sirf_audio_codec->regmap, -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Fix the issue that the private value cannot be accessed
by Oder Chiou 21 Nov '14

21 Nov '14

Fix the issue that the private value cannot be accessed. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 37 ++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 2 +- 2 files changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2211f1..133010d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4287,6 +4287,7 @@ static int rt5677_probe(struct snd_soc_codec *codec) } mutex_init(&rt5677->dsp_cmd_lock); + mutex_init(&rt5677->dsp_pri_lock); return 0; } @@ -4344,10 +4345,19 @@ static int rt5677_read(void *context, unsigned int reg, unsigned int *val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_read(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + } + } else { regmap_read(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4357,10 +4367,20 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_write(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA, + val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + } + } else { regmap_write(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4495,10 +4515,13 @@ static const struct regmap_config rt5677_regmap_physical = { .reg_bits = 8, .val_bits = 16, - .max_register = RT5677_VENDOR_ID2 + 1, + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), .readable_reg = rt5677_readable_register, .cache_type = REGCACHE_NONE, + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), }; static const struct regmap_config rt5677_regmap = { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a02f64c..dbd9ffd 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1670,7 +1670,7 @@ struct rt5677_priv { struct rt5677_platform_data pdata; struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; - struct mutex dsp_cmd_lock; + struct mutex dsp_cmd_lock, dsp_pri_lock; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: samsung: ASoC: samsung: Fix IISMOD setting in i2s_set_sysclk()
by Padmavathi Venna 21 Nov '14

21 Nov '14

In the i2s_set_sysclk() callback we are currently clearing all bits of the IISMOD register in i2s_set_sysclk. It's due to an incorrect mask used for the AND operation which is introduced in commit a5a56871f804edac93a53b5e871c0e9818fb9033 (ASoC: samsung: add support for exynos7 I2S controller) and also adds the missing break statement. Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- sound/soc/samsung/i2s.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0df6547..e1ace52 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -494,7 +494,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) mod |= 1 << i2s_regs->cdclkcon_off; else - mod &= 0 << i2s_regs->cdclkcon_off; + mod &= ~(1 << i2s_regs->cdclkcon_off); i2s->rfs = rfs; break; @@ -551,10 +551,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, } if (clk_id == 0) - mod &= 0 << i2s_regs->rclksrc_off; + mod &= ~(1 << i2s_regs->rclksrc_off); else mod |= 1 << i2s_regs->rclksrc_off; + break; default: dev_err(&i2s->pdev->dev, "We don't serve that!\n"); return -EINVAL; -- 1.7.4.4

3 2

[alsa-devel] [PATCH 1/6] ASoC: sigmadsp: Refuse to load firmware files with a non-supported version
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

Make sure to check the version field of the firmware header to make sure to not accidentally try to parse a firmware file with a different layout. Trying to do so can result in loading invalid firmware code to the device. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/sigmadsp.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index f2de7e0..81a38dd 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -159,6 +159,13 @@ int _process_sigma_firmware(struct device *dev, goto done; } + if (ssfw_head->version != 1) { + dev_err(dev, + "Failed to load firmware: Invalid version %d. Supported firmware versions: 1\n", + ssfw_head->version); + goto done; + } + crc = crc32(0, fw->data + sizeof(*ssfw_head), fw->size - sizeof(*ssfw_head)); pr_debug("%s: crc=%x\n", __func__, crc); -- 1.8.0

2 8

[alsa-devel] [PATCH] alsa-kernel: Add big-endian DSD sample formats and fix XMOS DSD sample format
by Jussi Laako 21 Nov '14

21 Nov '14

This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds DSD_U16_BE and DSD_U32_BE sample formats. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ sound/core/pcm_misc.c | 8 ++++++++ sound/usb/quirks.c | 4 ++-- 5 files changed, 17 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 40289ec..1e7f74a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -184,6 +184,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_DSD_U8 _SNDRV_PCM_FMTBIT(DSD_U8) #define SNDRV_PCM_FMTBIT_DSD_U16_LE _SNDRV_PCM_FMTBIT(DSD_U16_LE) #define SNDRV_PCM_FMTBIT_DSD_U32_LE _SNDRV_PCM_FMTBIT(DSD_U32_LE) +#define SNDRV_PCM_FMTBIT_DSD_U16_BE _SNDRV_PCM_FMTBIT(DSD_U16_BE) +#define SNDRV_PCM_FMTBIT_DSD_U32_BE _SNDRV_PCM_FMTBIT(DSD_U32_BE) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 6ee5867..941d32f 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -220,7 +220,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ #define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ #define SNDRV_PCM_FORMAT_DSD_U32_LE ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_LE +#define SNDRV_PCM_FORMAT_DSD_U16_BE ((__force snd_pcm_format_t) 51) /* DSD, 2-byte samples DSD (x16), big endian */ +#define SNDRV_PCM_FORMAT_DSD_U32_BE ((__force snd_pcm_format_t) 52) /* DSD, 4-byte samples DSD (x32), big endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_BE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8f624b7..cfc56c8 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -216,6 +216,8 @@ static char *snd_pcm_format_names[] = { FORMAT(DSD_U8), FORMAT(DSD_U16_LE), FORMAT(DSD_U32_LE), + FORMAT(DSD_U16_BE), + FORMAT(DSD_U32_BE), }; /** diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index ae7a0fe..ebe8444 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -152,6 +152,14 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { .width = 32, .phys = 32, .le = 1, .signd = 0, .silence = { 0x69, 0x69, 0x69, 0x69 }, }, + [SNDRV_PCM_FORMAT_DSD_U16_BE] = { + .width = 16, .phys = 16, .le = 0, .signd = 0, + .silence = { 0x69, 0x69 }, + }, + [SNDRV_PCM_FORMAT_DSD_U32_BE] = { + .width = 32, .phys = 32, .le = 0, .signd = 0, + .silence = { 0x69, 0x69, 0x69, 0x69 }, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2c1018e..bc9ebb6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1202,12 +1202,12 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* iFi Audio micro/nano iDSD */ case USB_ID(0x20b1, 0x3008): if (fp->altsetting == 2) - return SNDRV_PCM_FMTBIT_DSD_U32_LE; + return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2009): if (fp->altsetting == 3) - return SNDRV_PCM_FMTBIT_DSD_U32_LE; + return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; default: break; -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: samsung: In the i2s_set_sysclk() callback we are currently clearing all bits of the IISMOD register in i2s_set_sysclk. It's due to an incorrect mask used for the AND operation which is introduced in commit a5a56871f804edac93a53b5e871c0e9818fb9033 (ASoC: samsung: add support for exynos7 I2S controller) and also adds the missing break statement.
by Padmavathi Venna 21 Nov '14

21 Nov '14

Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- sound/soc/samsung/i2s.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0df6547..e1ace52 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -494,7 +494,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) mod |= 1 << i2s_regs->cdclkcon_off; else - mod &= 0 << i2s_regs->cdclkcon_off; + mod &= ~(1 << i2s_regs->cdclkcon_off); i2s->rfs = rfs; break; @@ -551,10 +551,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, } if (clk_id == 0) - mod &= 0 << i2s_regs->rclksrc_off; + mod &= ~(1 << i2s_regs->rclksrc_off); else mod |= 1 << i2s_regs->rclksrc_off; + break; default: dev_err(&i2s->pdev->dev, "We don't serve that!\n"); return -EINVAL; -- 1.7.4.4

4 3

Re: [alsa-devel] [PATCH 0/1] Add big-endian DSD sample formats and fix XMOS DSD sample format
by Takashi Iwai 21 Nov '14

21 Nov '14

At Fri, 21 Nov 2014 12:00:26 +0200, Jussi Laako wrote: > > Resend, previous had non matching subject for cover letter vs patch. > > This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds > DSD_U16_BE and DSD_U32_BE sample formats. > > Jussi Laako (1): > Add big-endian DSD sample formats and fix XMOS DSD sample format The changes there look good (supposing you tested with the real device). Since DSD_U32 was added in 3.18-rc1, I think we can put this as a fix for 3.18. Could you submit the patch for alsa-lib as well? thanks, Takashi

2 2

Re: [alsa-devel] [PATCH 0/1] Add big-endian DSD sample formats and fix XMOS DSD sample format
by Jurgen Kramer 21 Nov '14

21 Nov '14

On Fri, 2014-11-21 at 12:00 +0200, Jussi Laako wrote: > Resend, previous had non matching subject for cover letter vs patch. > > This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds > DSD_U16_BE and DSD_U32_BE sample formats. > > Jussi Laako (1): > Add big-endian DSD sample formats and fix XMOS DSD sample format > > include/sound/pcm.h | 2 ++ > include/uapi/sound/asound.h | 4 +++- > sound/core/pcm.c | 2 ++ > sound/core/pcm_misc.c | 8 ++++++++ > sound/usb/quirks.c | 4 ++-- > 5 files changed, 17 insertions(+), 3 deletions(-) > Acked-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> I have tested the complete chain (added DSD_BE to kernel, alsa-lib, alsa-utils) with the change to BE. Everything still works. Jurgen

1 0

[alsa-devel] HP mute led for ALC286
by Kailang 21 Nov '14

21 Nov '14

Hi Takashi, I had sent the discussion mail about the DMI string for HP. But no body response. So, I just send the patch to fix this issue. BR, Kailang

2 5

[alsa-devel] [PATCH] ALSA: hda - One more HP machine needs to change mute led quirk
by Hui Wang 21 Nov '14

21 Nov '14

The machine originally use the quirk ALC269_FIXUP_HP_GPIO_MIC1_LED, but the LED doesn't work at all. After this change, the machine will change to use ALC269_FIXUP_HP_MUTE_LED_MIC1 through pin_fixup_tbl[], and the LED works well. BugLink: https://bugs.launchpad.net/bugs/1389497 Tested-by: TieFu Chen <tienfu.chen(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cd08d8a..2ac9e23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4831,7 +4831,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), -- 1.9.1

2 1

[alsa-devel] [PATCH] pcm: add new 32-bit DSD sample format
by Jurgen Kramer 21 Nov '14

21 Nov '14

Add the new DSD_U32_LE sample format to alsa-lib. NB include/pcm.h and include/sound/asound.h are updated so a new sync with the kernel headers is not needed Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- include/pcm.h | 4 +++- include/sound/asound.h | 3 ++- src/pcm/pcm.c | 2 ++ src/pcm/pcm_misc.c | 6 ++++++ 4 files changed, 13 insertions(+), 2 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 11e9f0d..db88ad5 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -211,7 +211,9 @@ typedef enum _snd_pcm_format { SND_PCM_FORMAT_DSD_U8, /* Direct Stream Digital (DSD) in 2-byte samples (x16) */ SND_PCM_FORMAT_DSD_U16_LE, - SND_PCM_FORMAT_LAST = SND_PCM_FORMAT_DSD_U16_LE, + /* Direct Stream Digital (DSD) in 4-byte samples (x32) */ + SND_PCM_FORMAT_DSD_U32_LE, + SND_PCM_FORMAT_LAST = SND_PCM_FORMAT_DSD_U32_LE, #if __BYTE_ORDER == __LITTLE_ENDIAN /** Signed 16 bit CPU endian */ diff --git a/include/sound/asound.h b/include/sound/asound.h index 32168f7..6ee5867 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -219,7 +219,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ #define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE +#define SNDRV_PCM_FORMAT_DSD_U32_LE ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_LE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1399a5b..2e24338 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1565,6 +1565,7 @@ static const char *const snd_pcm_format_names[] = { FORMAT(G723_40_1B), FORMAT(DSD_U8), FORMAT(DSD_U16_LE), + FORMAT(DSD_U32_LE), }; static const char *const snd_pcm_format_aliases[SND_PCM_FORMAT_LAST+1] = { @@ -1624,6 +1625,7 @@ static const char *const snd_pcm_format_descriptions[] = { FORMATD(G723_40_1B, "G.723 (ADPCM) 40 kbit/s, 1 sample in 1 byte"), FORMATD(DSD_U8, "Direct Stream Digital, 1-byte (x8), oldest bit in MSB"), FORMATD(DSD_U16_LE, "Direct Stream Digital, 2-byte (x16), little endian, oldest bits in MSB"), + FORMATD(DSD_U32_LE, "Direct Stream Digital, 4-byte (x32), little endian, oldest bits in MSB"), }; static const char *const snd_pcm_type_names[] = { diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 46fc771..9272179 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -64,6 +64,7 @@ int snd_pcm_format_signed(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3BE: case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0; default: return -EINVAL; @@ -154,6 +155,7 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3BE: case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0; default: return -EINVAL; @@ -232,6 +234,7 @@ int snd_pcm_format_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U32_BE: case SNDRV_PCM_FORMAT_FLOAT_LE: case SNDRV_PCM_FORMAT_FLOAT_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 32; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -292,6 +295,7 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_FLOAT_BE: case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE: case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 32; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -348,6 +352,7 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) case SNDRV_PCM_FORMAT_U32_BE: case SNDRV_PCM_FORMAT_FLOAT_LE: case SNDRV_PCM_FORMAT_FLOAT_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return samples * 4; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -394,6 +399,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) return 0x8080808080808080ULL; case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0x6969696969696969ULL; #ifdef SNDRV_LITTLE_ENDIAN case SNDRV_PCM_FORMAT_U16_LE: -- 1.9.3

3 7

[alsa-devel] [PATCH 0/1] Add big-endian DSD sample formats and fix XMOS DSD sample format
by Jussi Laako 21 Nov '14

21 Nov '14

Resend, previous had non matching subject for cover letter vs patch. This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds DSD_U16_BE and DSD_U32_BE sample formats. Jussi Laako (1): Add big-endian DSD sample formats and fix XMOS DSD sample format include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ sound/core/pcm_misc.c | 8 ++++++++ sound/usb/quirks.c | 4 ++-- 5 files changed, 17 insertions(+), 3 deletions(-) -- 1.9.1

1 1

[alsa-devel] [PATCH 0/1] Fix DSD sample format endianess and add big-endian format
by Jussi Laako 21 Nov '14

21 Nov '14

This patch adds big-endian DSD sample format and fixes return value of DSD formats for snd_pcm_format_little_endian(). Jussi Laako (1): Fix DSD sample format endianess and add big-endian format include/pcm.h | 6 +++++- include/sound/asound.h | 4 +++- src/pcm/pcm.c | 4 ++++ src/pcm/pcm_misc.c | 17 ++++++++++++++--- 4 files changed, 26 insertions(+), 5 deletions(-) -- 1.9.1

1 1

[alsa-devel] [PATCH 0/1] Fix XMOS DSD suppport and add big-endian DSD sample formats
by Jussi Laako 21 Nov '14

21 Nov '14

This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds DSD_U16_BE and DSD_U32_BE sample formats. Jussi Laako (1): Add big-endian DSD sample formats and fix XMOS DSD sample format include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ sound/core/pcm_misc.c | 8 ++++++++ sound/usb/quirks.c | 4 ++-- 5 files changed, 17 insertions(+), 3 deletions(-) -- 1.9.1

1 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: cleanup runtime_pm initialization
by Vinod Koul 21 Nov '14

21 Nov '14

For ACPI we missed to pm_runtime_enable() call which is required to tell PM core that runtime on this device is enabled now. Since this is common to both PCI and APCI move it out. Also for ACPI we do not require pm_runtime_allow() call, so remove that Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/sst/sst.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index b2b5604..9e68a7c 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -378,13 +378,13 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ - if (acpi_disabled) { + pm_runtime_enable(ctx->dev); + + if (acpi_disabled) pm_runtime_set_active(ctx->dev); - pm_runtime_enable(ctx->dev); - } else { - pm_runtime_allow(ctx->dev); + else pm_runtime_put_noidle(ctx->dev); - } + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); } EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); -- 1.7.0.4

3 7

[alsa-devel] [PATCH] ALSA: usb-audio: Use snd_usb_ctl_msg() for Native Instruments quirk
by Takashi Iwai 20 Nov '14

20 Nov '14

snd_nativeinstruments_control_get() uses a stack as a buffer for usb_control_msg(), but it's basically not allowed. Replace the call with a safer helper, snd_usb_ctl_msg(), instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 03405b93b438..8b55c0667f60 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -702,10 +702,10 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, if (mixer->chip->shutdown) ret = -ENODEV; else - ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0, wIndex, - &tmp, sizeof(tmp), 1000); + &tmp, sizeof(tmp)); up_read(&mixer->chip->shutdown_rwsem); if (ret < 0) { -- 2.1.3

1 0

[alsa-devel] [PATCH RFC 00/10] Add proper resume support to USB-audio quirks
by Takashi Iwai 20 Nov '14

20 Nov '14

Hi, here is a patch series to add proper resume support to USB-audio quirks. So far, the control elements created directly by quirk functions aren't resumed restore_mixer_value(). This patch series try to address it. [PATCH RFC 01/10] ALSA: usb-audio: Refactor ignore_ctl_error checks [PATCH RFC 02/10] ALSA: usb-audio: Allow quirks to handle own resume [PATCH RFC 03/10] ALSA: usb-audio: Add audigy2nx resume support [PATCH RFC 04/10] ALSA: usb-audio: Add Emu0204 channel switch resume [PATCH RFC 05/10] ALSA: usb-audio: Add Xonar U1 resume support [PATCH RFC 06/10] ALSA: usb-audio: Add Digidesign Mbox 1 resume [PATCH RFC 07/10] ALSA: usb-audio: Add resume support for Native [PATCH RFC 08/10] ALSA: usb-audio: Add resume support for FTU [PATCH RFC 09/10] ALSA: usb-audio: Add resume support for MicroII [PATCH RFC 10/10] ALSA: usb-audio: Add resume support for Scarlett I have only compile-tested this, so far. It'd be appreciated if anyone with such devices can test with real devices. thanks, Takashi

1 10

[alsa-devel] [PATCH] ASoC: samsung: Fix IISMOD setting in i2s_set_sysclk()
by Sylwester Nawrocki 20 Nov '14

20 Nov '14

In the i2s_set_sysclk() callback we are currently clearing all bits of the IISMOD register when clk_id is SAMSUNG_I2S_CDCLK and dir is SND_SOC_CLOCK_OUT. It's due to an incorrect mask used for the AND operation and breaks the I2S interface operation when we attempt to use the CDCLK output clock. It fixes commit a5a56871f804edac93a53b5e871c0e9818fb9033 ("ASoC: samsung: add support for exynos7 I2S controller") and restores sound support for Odroid X2/U3. Cc: Padmavathi Venna <padma.v(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 947352d..8db8c66 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -494,7 +494,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) mod |= 1 << i2s_regs->cdclkcon_off; else - mod &= 0 << i2s_regs->cdclkcon_off; + mod &= ~(1 << i2s_regs->cdclkcon_off); i2s->rfs = rfs; break; -- 1.7.9.5

2 3

[alsa-devel] [PATCH v2 1/2] ASoC: wm8960: Add device tree support
by Zidan Wang 20 Nov '14

20 Nov '14

Document the device tree binding for the WM8960 codec, and modify the driver to extract the platform data from device tree, if present. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 27 ++++++++++++++ sound/soc/codecs/wm8960.c | 41 ++++++++++++++++------ 2 files changed, 58 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt new file mode 100644 index 0000000..1a9ad07 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -0,0 +1,27 @@ +WM8960 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8960" + + - reg : the I2C address of the device. + +Optional properties: + - shared_lrclk: This is a boolean property. If present, the LRCM bit of + R24 (Additional control 2 ) gets set, indicating that ADCLRC and DACLRC + will be disabled only when ADC (Left and Right) and DAC (Left and Right) + are disabled. + + - capless: This is a boolean property. If present, OUT3 will be enabled + and disabled together with HP_L and HP_R in response to jack detect events. + +Example: + +codec: wm8960@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + + shared_lrclk; +}; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4dc4e85..e5d89a2 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -125,6 +125,7 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + struct wm8960_data pdata; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) @@ -440,8 +441,8 @@ static const struct snd_soc_dapm_route audio_paths_capless[] = { static int wm8960_add_widgets(struct snd_soc_codec *codec) { - struct wm8960_data *pdata = codec->dev->platform_data; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + struct wm8960_data *pdata = &wm8960->pdata; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; @@ -961,17 +962,13 @@ static int wm8960_resume(struct snd_soc_codec *codec) static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - struct wm8960_data *pdata = dev_get_platdata(codec->dev); + struct wm8960_data *pdata = &wm8960->pdata; int ret; - wm8960->set_bias_level = wm8960_set_bias_level_out3; - - if (!pdata) { - dev_warn(codec->dev, "No platform data supplied\n"); - } else { - if (pdata->capless) - wm8960->set_bias_level = wm8960_set_bias_level_capless; - } + if (pdata->capless) + wm8960->set_bias_level = wm8960_set_bias_level_capless; + else + wm8960->set_bias_level = wm8960_set_bias_level_out3; ret = wm8960_reset(codec); if (ret < 0) { @@ -1029,6 +1026,18 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; +static void wm8960_set_pdata_from_of(struct i2c_client *i2c, + struct wm8960_data *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + + if (of_property_read_bool(np, "capless")) + pdata->capless = true; + + if (of_property_read_bool(np, "shared_lrclk")) + pdata->shared_lrclk = true; +} + static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1045,6 +1054,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); + if (pdata) + memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); + else if (i2c->dev.of_node) + wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (pdata && pdata->shared_lrclk) { ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 0x4, 0x4); @@ -1075,10 +1089,17 @@ static const struct i2c_device_id wm8960_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); +static const struct of_device_id wm8960_of_match[] = { + { .compatible = "wlf,wm8960", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8960_of_match); + static struct i2c_driver wm8960_i2c_driver = { .driver = { .name = "wm8960", .owner = THIS_MODULE, + .of_match_table = wm8960_of_match, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

3 3

[alsa-devel] AKAI EIE PRO
by Martin Svensson 20 Nov '14

20 Nov '14

Dont know if its ok to ask a question here so please let me know if its not ok then i wont ask again ;) i just bought an external usb AKAI EIE PRO sound card, is there any future plans to implement it in Alsa? Sincerally Martin aka "ztealmax"

4 7

[alsa-devel] Baytrail init file for alsa-utils
by Bastien Nocera 20 Nov '14

20 Nov '14

Hey, I've done some work on the community generated "alsa state" file for Baytrail tablets, and shrunk it massively: https://github.com/AdamWill/baytrail-m/blob/master/alsa/t100_B.state Are there any tools, or hints as to how I might convert that to an "init" file that alsactl could run to set the default state of the mixer? Or have any Intel folks already done that work? I'll have a few more sound-related questions for that tablet, for which I could do with a few hints as to where to direct the questions/bug reports: - no jack sensing support (driver problem?) - left/right headphones output is inverted (another driver bug?) - HDMI audio is not detected (Intel video driver bug?) Cheers

3 5

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Minor coding style and typo fix
by Oder Chiou 20 Nov '14

20 Nov '14

Minor coding style and typo fix Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677-spi.h | 4 ++-- sound/soc/codecs/rt5677.c | 14 +++++++------- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h index 7528bfd..ec41b2b 100644 --- a/sound/soc/codecs/rt5677-spi.h +++ b/sound/soc/codecs/rt5677-spi.h @@ -9,8 +9,8 @@ * published by the Free Software Foundation. */ -#ifndef __RT5671_SPI_H__ -#define __RT5671_SPI_H__ +#ifndef __RT5677_SPI_H__ +#define __RT5677_SPI_H__ #define RT5677_SPI_BUF_LEN 240 #define RT5677_SPI_CMD_BURST_WRITE 0x05 diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ca264f8..0d24dc4 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1353,7 +1353,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux = SOC_DAPM_ENUM("IB45 Bypass Source", rt5677_ib45_bypass_src_enum); -/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ +/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ static const char * const rt5677_stereo_adc2_src[] = { "DD MIX1", "DMIC", "Stereo DAC MIX" }; @@ -1438,7 +1438,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux = SOC_DAPM_ENUM("Stereo2 ADC LR Source", rt5677_stereo2_adc_lr_enum); -/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ +/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ static const char * const rt5677_stereo_adc1_src[] = { "DD MIX1", "ADC1/2", "Stereo DAC MIX" }; @@ -1710,7 +1710,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_pdm2_r_mux = SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_r_enum); -/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/ +/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0] */ static const char * const rt5677_if12_adc1_src[] = { "STO1 ADC MIX", "OB01", "VAD ADC" }; @@ -1788,7 +1788,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_slb_adc3_mux = SOC_DAPM_ENUM("SLB ADC3 Source", rt5677_slb_adc3_enum); -/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ +/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ static const char * const rt5677_if12_adc4_src[] = { "STO4 ADC MIX", "OB67", "OB01" }; @@ -1814,7 +1814,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_slb_adc4_mux = SOC_DAPM_ENUM("SLB ADC4 Source", rt5677_slb_adc4_enum); -/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/ +/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4] */ static const char * const rt5677_if34_adc_src[] = { "STO1 ADC MIX", "STO2 ADC MIX", "STO3 ADC MIX", "STO4 ADC MIX", "MONO ADC MIX", "OB01", "OB23", "VAD ADC" @@ -1895,7 +1895,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if2_adc4_swap_mux = SOC_DAPM_ENUM("IF2 ADC4 Swap Source", rt5677_if2_adc4_swap_enum); -/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ +/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ static const char * const rt5677_if1_adc_tdm_swap_src[] = { "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", "3/1/2/4", "3/4/1/2" @@ -2442,7 +2442,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { rt5677_ob_7_mix, ARRAY_SIZE(rt5677_ob_7_mix)), /* Output Side */ - /* DAC mixer before sound effect */ + /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, rt5677_dac_l_mix, ARRAY_SIZE(rt5677_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, -- 1.8.1.1.439.g50a6b54

3 4

[alsa-devel] [PATCH] ASoC: wm8960: Add shared_lrclk and capless to device tree
by Zidan Wang 19 Nov '14

19 Nov '14

wm8960 codec has a wm8960_data struct which has two fields shared_lrclk and capless. The wm8960_data is get from platform_data and it is reasonable to set it from device tree when platform_data is null. And when shared_lrclk is set, LRCM will be enabled. But the following software reset in wm8960_probe will reset it to the default state. So LRCM operation should after software reset. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 49 ++++++++++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4dc4e85..1d56c90 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -125,6 +125,7 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + struct wm8960_data pdata; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) @@ -961,17 +962,13 @@ static int wm8960_resume(struct snd_soc_codec *codec) static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - struct wm8960_data *pdata = dev_get_platdata(codec->dev); + struct wm8960_data *pdata = &wm8960->pdata; int ret; - wm8960->set_bias_level = wm8960_set_bias_level_out3; - - if (!pdata) { - dev_warn(codec->dev, "No platform data supplied\n"); - } else { - if (pdata->capless) - wm8960->set_bias_level = wm8960_set_bias_level_capless; - } + if (pdata->capless) + wm8960->set_bias_level = wm8960_set_bias_level_capless; + else + wm8960->set_bias_level = wm8960_set_bias_level_out3; ret = wm8960_reset(codec); if (ret < 0) { @@ -979,6 +976,14 @@ static int wm8960_probe(struct snd_soc_codec *codec) return ret; } + if (pdata->shared_lrclk) { + ret = snd_soc_update_bits(codec, WM8960_ADDCTL2, 0x4, 0x4); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable LRCM: %d\n", ret); + return ret; + } + } + wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ @@ -1029,6 +1034,18 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; +static void wm8960_set_pdata_from_of(struct i2c_client *i2c, + struct wm8960_data *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + + if (of_property_read_bool(np, "capless")) + pdata->capless = true; + + if (of_property_read_bool(np, "shared_lrclk")) + pdata->shared_lrclk = true; +} + static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1038,6 +1055,7 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, wm8960 = devm_kzalloc(&i2c->dev, sizeof(struct wm8960_priv), GFP_KERNEL); + if (wm8960 == NULL) return -ENOMEM; @@ -1045,15 +1063,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); - if (pdata && pdata->shared_lrclk) { - ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, - 0x4, 0x4); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", - ret); - return ret; - } - } + if (pdata) + memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); + else if (i2c->dev.of_node) + wm8960_set_pdata_from_of(i2c, &wm8960->pdata); i2c_set_clientdata(i2c, wm8960); -- 1.9.1

4 3

[alsa-devel] [PATCH 0/3] Driver for Cirrus Logic CS47L24 codec
by Richard Fitzgerald 19 Nov '14

19 Nov '14

This set of patches add support for the Cirrus Logic CS47L24 codec. The patches depend on two open patches from Charles Keepax which must be applied first: [PATCH] mfd: wm5110: Add missing registers for AIF2 channels 3-6 [PATCH] mfd: arizona: Document HP_CTRL_1L and HP_CTRL_1R registers Richard Fitzgerald (3): mfd: arizona: Add support for CS47L24 gpio: arizona: Add support for CS47L24 ASoC: arizona: Add initial CS47L24 codec driver drivers/gpio/gpio-arizona.c | 5 + drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 91 ++- drivers/mfd/arizona-irq.c | 42 +- drivers/mfd/arizona-spi.c | 9 + drivers/mfd/arizona.h | 5 + drivers/mfd/cs47l24-tables.c | 1632 ++++++++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/core.h | 4 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 5 + sound/soc/codecs/cs47l24.c | 1133 ++++++++++++++++++++++++++ sound/soc/codecs/cs47l24.h | 23 + 14 files changed, 2941 insertions(+), 27 deletions(-) create mode 100644 drivers/mfd/cs47l24-tables.c create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h -- 1.7.2.5

5 22

[alsa-devel] [PATCH 0/4] ASoC: AC'97 regmap support and conversion
by Lars-Peter Clausen 19 Nov '14

19 Nov '14

This series contains the final pieces for the AC'97 conversion to regmap. The first patch adds a regmap backend for AC'97 devices that will use the AC'97 bus of the device to perform reads and writes. The second patch adds a bit of plumbing to the ASoC framework that it makes it possible to assign a regmap instance to ASoC CODEC (or component) after it has been registered. This is necessary since for AC'97 drivers we allocate the AC'97 device in the CODEC drivers probe callback. For now the series only converts the ad1980 driver, others might follow later. - Lars Lars-Peter Clausen (3): ASoC: Add helper functions for deferred regmap setup ASoC: ad1980: Convert to regmap ASoC: ad1980: Remove ac97_read/ac97_write wrappers Mark Brown (1): regmap: ac97: Add generic AC'97 callbacks drivers/base/regmap/Kconfig | 5 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-ac97.c | 114 +++++++++++++++++++++++++ include/linux/regmap.h | 7 ++ include/sound/soc.h | 35 ++++++++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/ad1980.c | 173 +++++++++++++++++++++----------------- sound/soc/soc-core.c | 58 ++++++++++--- 8 files changed, 306 insertions(+), 88 deletions(-) create mode 100644 drivers/base/regmap/regmap-ac97.c -- 1.8.0

2 9

[alsa-devel] [PATCH 0/2] ASoC: rockchip: i2s: add support for grabbing output clock to codec
by Jianqun Xu 19 Nov '14

19 Nov '14

Reference to Chapter 3 Clock & Reset Unit (CRU) of RK3288 TRM, i2s0_clkout has two source clock, XIN24M_DIV2 for 12M and clk_i2s0 for more kinds of frequencies, I2S0(i2s_clk) has only single source clock clk_i2s0. i2s0_clkout: generate to output to outside of chip, generally for codec I2S0(i2s_clk): generate for i2s controller inside of chip Jianqun Xu (2): ASoC: rockchip-i2s: dt: add i2s_clkout to list of clocks ASoC: rockchip: i2s: add support for grabbing output clock to codec Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 5 +++-- sound/soc/rockchip/rockchip_i2s.c | 11 +++++++++++ 2 files changed, 14 insertions(+), 2 deletions(-) -- 1.9.1

6 9

[alsa-devel] [PATCH 1/3] ASoC: rt5677: Align the reg_default table with tab character
by Oder Chiou 19 Nov '14

19 Nov '14

Align the reg_default table with tab character Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 9ae2e84..b2d88bb 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,13 +55,13 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_ASRC_12, 0x0018}, - {RT5677_PR_BASE + 0x3d, 0x364d}, - {RT5677_PR_BASE + 0x17, 0x4fc0}, - {RT5677_PR_BASE + 0x13, 0x0312}, - {RT5677_PR_BASE + 0x1e, 0x0000}, - {RT5677_PR_BASE + 0x12, 0x0eaa}, - {RT5677_PR_BASE + 0x14, 0x018a}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_PR_BASE + 0x17, 0x4fc0}, + {RT5677_PR_BASE + 0x13, 0x0312}, + {RT5677_PR_BASE + 0x1e, 0x0000}, + {RT5677_PR_BASE + 0x12, 0x0eaa}, + {RT5677_PR_BASE + 0x14, 0x018a}, }; #define RT5677_INIT_REG_LEN ARRAY_SIZE(init_list) -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v2 0/6] media token resource framework
by Shuah Khan 18 Nov '14

18 Nov '14

Add media token device resource framework to allow sharing resources such as tuner, dma, audio etc. across media drivers and non-media sound drivers that control media hardware. The Media token resource is created at the main struct device that is common to all drivers that claim various pieces of the main media device, which allows them to find the resource using the main struct device. As an example, digital, analog, and snd-usb-audio drivers can use the media token resource API using the main struct device for the interface the media device is attached to. This patch series consists of media token resource framework and changes to use it in dvb-core, v4l2-core, au0828 driver, and snd-usb-audio driver. With these changes dvb and v4l2 can share the tuner without disrupting each other. Used tvtime, xawtv, kaffeine, and vlc, vlc audio capture option, arecord/aplay during development to identify v4l2 vb2 and vb1 ioctls and file operations that disrupt the digital stream and would require changes to check tuner ownership prior to changing the tuner configuration. vb2 changes are made in the v4l2-core and vb1 changes are made in the au0828 driver to encourage porting drivers to vb2 to advantage of the new media token resource framework with changes in the core. In this patch v2 series, fixed problems identified in the patch v1 series. Important ones are changing snd-usb-audio to use media tokens, holding tuner lock in VIDIOC_ENUMINPUT, and VIDIOC_QUERYSTD. Shuah Khan (6): media: add media token device resource framework media: v4l2-core changes to use media token api media: au0828-video changes to use media token api media: dvb-core changes to use media token api sound/usb: pcm changes to use media token api media: au0828-core changes to create and destroy media MAINTAINERS | 2 + drivers/media/dvb-core/dvb_frontend.c | 14 +- drivers/media/usb/au0828/au0828-core.c | 23 +++ drivers/media/usb/au0828/au0828-video.c | 42 +++++- drivers/media/v4l2-core/v4l2-fh.c | 7 + drivers/media/v4l2-core/v4l2-ioctl.c | 61 ++++++++ include/linux/media_tknres.h | 50 +++++++ lib/Makefile | 2 + lib/media_tknres.c | 237 +++++++++++++++++++++++++++++++ sound/usb/pcm.c | 9 ++ 10 files changed, 445 insertions(+), 2 deletions(-) create mode 100644 include/linux/media_tknres.h create mode 100644 lib/media_tknres.c -- 1.7.10.4

11 56

[alsa-devel] [PATCH] ASoC: wm_adsp: Move core_ena to be co-located with start bit
by Charles Keepax 18 Nov '14

18 Nov '14

Many firmwares do not wait for the start bit before they begin processing audio, whilst this is a bug on the firmware side there are too many such firmwares in the wild to ignore the situation. This patch moves the core enable to happen at same time as the start, the firmware looses the ability to overlap its own startup with the audio path bring up but we ensure that all firmwares behave. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 11 ++--------- 1 files changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e54d7b5..720d6e8 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1593,13 +1593,6 @@ static void wm_adsp2_boot_work(struct work_struct *work) if (ret != 0) goto err; - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA, - ADSP2_CORE_ENA); - if (ret != 0) - goto err; - dsp->running = true; return; @@ -1649,8 +1642,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_START, - ADSP2_START); + ADSP2_CORE_ENA | ADSP2_START, + ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; break; -- 1.7.2.5

2 1

[alsa-devel] [PATCH v2] ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs
by Charles Keepax 18 Nov '14

18 Nov '14

From: JS Park <aitdark.park(a)samsung.com> If we fail to allocate memory for one of the memory regions whilst parsing the DSP firmware, we will leak the buffer holding the algorithm information. This patch ensures that this buffer is freed on all error paths. Signed-off-by: JS Park <aitdark.park(a)samsung.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 30 ++++++++++++++++++++---------- 1 files changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 08d8ec3..e54d7b5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1053,8 +1053,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); @@ -1071,8 +1073,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); @@ -1101,8 +1105,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); @@ -1119,8 +1125,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); @@ -1137,8 +1145,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: fix kerneldoc errors
by Takashi Sakamoto 18 Nov '14

18 Nov '14

Complete missing parameters, correct wrong reference, and add an explaination about the differences between the latest specification and our implementation. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.h | 24 ++++++++++++++++++++++-- sound/firewire/cmp.c | 2 ++ 2 files changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 4823c08..e6e8926 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -23,7 +23,7 @@ * corresponds to the end of event in the packet. Out of IEC 61883. * @CIP_WRONG_DBS: Only for in-stream. The value of dbs is wrong in in-packets. * The value of data_block_quadlets is used instead of reported value. - * @SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is + * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is * skipped for detecting discontinuity. * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first * packet is not continuous from an initial value. @@ -43,7 +43,27 @@ enum cip_flags { }; /** - * enum cip_sfc - a stream's sample rate + * enum cip_sfc - supported Sampling Frequency Codes (SFCs) + * @CIP_SFC_32000: 32,000 data blocks + * @CIP_SFC_44100: 44,100 data blocks + * @CIP_SFC_48000: 48,000 data blocks + * @CIP_SFC_88200: 88,200 data blocks + * @CIP_SFC_96000: 96,000 data blocks + * @CIP_SFC_176400: 176,400 data blocks + * @CIP_SFC_192000: 192,000 data blocks + * @CIP_SFC_COUNT: the number of supported SFCs + * + * These values are used to show nominal Sampling Frequency Code in + * Format Dependent Field (FDF) of AMDTP packet header. In IEC 61883-6:2002, + * this code means the number of events per second. Actually the code + * represents the number of data blocks transferred per second in an AMDTP + * stream. + * + * In IEC 61883-6:2005, some extensions were added to support more types of + * data such as 'One Bit LInear Audio', therefore the meaning of SFC became + * different depending on the types. + * + * Currently our implementation is compatible with IEC 61883-6:2002. */ enum cip_sfc { CIP_SFC_32000 = 0, diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c index ba8df5a..ae3bc19 100644 --- a/sound/firewire/cmp.c +++ b/sound/firewire/cmp.c @@ -114,6 +114,7 @@ static int pcr_modify(struct cmp_connection *c, * cmp_connection_init - initializes a connection manager * @c: the connection manager to initialize * @unit: a unit of the target device + * @direction: input or output * @pcr_index: the index of the iPCR/oPCR on the target device */ int cmp_connection_init(struct cmp_connection *c, @@ -154,6 +155,7 @@ EXPORT_SYMBOL(cmp_connection_init); /** * cmp_connection_check_used - check connection is already esablished or not * @c: the connection manager to be checked + * @used: the pointer to store the result of checking the connection */ int cmp_connection_check_used(struct cmp_connection *c, bool *used) { -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs
by Charles Keepax 18 Nov '14

18 Nov '14

From: JS Park <aitdark.park(a)samsung.com> Signed-off-by: JS Park <aitdark.park(a)samsung.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 30 ++++++++++++++++++++---------- 1 files changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 08d8ec3..e54d7b5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1053,8 +1053,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); @@ -1071,8 +1073,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); @@ -1101,8 +1105,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); @@ -1119,8 +1125,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); @@ -1137,8 +1145,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); -- 1.7.2.5

2 1

[alsa-devel] [PATCH 0/6] Add support for the WM8280 and WM8281 codecs
by Richard Fitzgerald 18 Nov '14

18 Nov '14

This set of patches adds support for the WM8280 and WM8281 codecs to the Wolfson Arizona drivers. Richard Fitzgerald (6): mfd: arizona: add support for WM8280/WM8281 Documentation: devicetree: arizona: Add bindings for WM8280 regulator: arizona-micsupp: Add support for WM8280/WM8281 gpio: arizona: Add support for WM8280/WM8281 extcon: arizona: Add support for WM8280/WM8281 ASoC: arizona: Add support for WM8280/WM8281 Documentation/devicetree/bindings/mfd/arizona.txt | 10 ++++++---- drivers/extcon/extcon-arizona.c | 1 + drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 5 +++-- drivers/mfd/arizona-core.c | 14 ++++++++++++-- drivers/mfd/arizona-i2c.c | 2 ++ drivers/mfd/arizona-irq.c | 1 + drivers/mfd/arizona-spi.c | 2 ++ drivers/regulator/arizona-micsupp.c | 1 + include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 2 ++ 11 files changed, 32 insertions(+), 8 deletions(-) -- 1.7.2.5

5 13

[alsa-devel] [PATCH] ASoC: rt286: build warning of section mismatch
by Sudip Mukherjee 18 Nov '14

18 Nov '14

From: Sudip Mukherjee <sudip(a)vectorindia.org> while building we were getting the following build warning: Section mismatch in reference from the function rt286_i2c_probe() to the variable .init.data:force_combo_jack_table The function rt286_i2c_probe() references the variable __initdata force_combo_jack_table. This is often because rt286_i2c_probe lacks a __initdata annotation or the annotation of force_combo_jack_table is wrong. we were getting the warning as force_combo_jack_table was marked with __initdata Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2e818aa..2cd4fe4 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); -static struct dmi_system_id force_combo_jack_table[] __initdata = { +static struct dmi_system_id force_combo_jack_table[] = { { .ident = "Intel Wilson Beach", .matches = { -- 1.8.1.2

2 1

[alsa-devel] [PATCH 00/11] ASoC: Untangle AC'97 support form the core
by Lars-Peter Clausen 18 Nov '14

18 Nov '14

The AC'97 support has had a special role within the ASoC framework with it being (in contrast to other audio buses) tightly integrated into the core. This patch series creates a clean separation between the core and the AC'97 support making AC'97 from the point of view of the core just another bus. Some of the changes in this series are required for the upcoming conversion of the AC'97 drivers to regmap, the other changes just fit nicely with the idea of componentization and having a generic ASoC core. The series starts with moving AC'97 support to its own file and doing a few general cleanups. After that it drops support of two features which were mainly responsible for the tight integration but are seemingly unused. This is a) Delaying the initialization and registration of the AC'97 device until after the card has been registered. There doesn't seem to be a good reason to delay to registration, all the information we need to register the device is available at the time when the device is created. Furthermore this will be needed for regmap. b) Setting platform data for the AC'97 via the CPU DAI. This hasn't been used in the last 5 years and doesn't seem to be that useful in general. If we should ever have a need for those features again they can still be implemented on top of the current code while keeping the clean separation. But for the time being it is simpler to just drop unused code rather than restructuring it. Finally the series pushes the pointer to the AC'97 device from the snd_soc_codec struct out to the driver private structs. Depends on asoc/topic/ad1980, asoc/topic/wm9705 and asoc/topic/wm971x - Lars Lars-Peter Clausen (11): ASoC: mpc5200_psc_ac97: Remove unused on-stack snd_ac97 device ASoC: mpc5200_dma: Don't overwrite ac97 device private_data ASoC: Properly handle AC'97 device lifetime management ASoC: Move AC'97 support to its own file ASoC: ac97: Use static ac97_bus ASoC: ac97: Merge soc_ac97_dev_{un,}register()/soc_{un,}register_ac97_codec() ASoC: ac97: Drop support for setting platform data via the CPU DAI ASoC: ac97: Drop delayed device registration ASoC: Drop ac97_control initialization from CODEC driver DAIs ASoC: Rename snd_soc_dai_driver struct ac97_control field to bus_control ASoC: ac97: Push snd_ac97 pointer to the driver level include/sound/soc-dai.h | 4 +- include/sound/soc.h | 25 ++- sound/soc/Makefile | 4 + sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/codecs/ac97.c | 18 +- sound/soc/codecs/ad1980.c | 28 +-- sound/soc/codecs/stac9766.c | 40 +++-- sound/soc/codecs/wm9705.c | 32 ++-- sound/soc/codecs/wm9712.c | 33 ++-- sound/soc/codecs/wm9713.c | 32 ++-- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 3 - sound/soc/fsl/mpc5200_psc_ac97.c | 6 +- sound/soc/nuc900/nuc900-ac97.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 6 +- sound/soc/samsung/ac97.c | 4 +- sound/soc/sh/hac.c | 2 +- sound/soc/soc-ac97.c | 256 +++++++++++++++++++++++++++ sound/soc/soc-core.c | 372 ++------------------------------------- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- 25 files changed, 428 insertions(+), 455 deletions(-) create mode 100644 sound/soc/soc-ac97.c -- 1.8.0

3 13

[alsa-devel] [PATCH 1/2] ALSA: hda - move DELL_WMI_MIC_MUTE_LED to the tail in the quirk chain
by Hui Wang 18 Nov '14

18 Nov '14

We have one more Dell machine needs DELL_WMI_MIC_MUTE_LED quirk, but the machine uses alc293 instead of alc255. So if DELL_WMI_MIC_MUTE_LED still chain ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, the machine can't use this quirk. To change this situation, let the DELL_WMI_MIC_MUTE_LED to be a standalone quirk, and let other quirks chain it. After this change, this quirk can be chained to any existing quirks, and as a result, it is possible that this quirk is applied to a non-Dell machine or a Dell machine without mic mute led on it, but it is still safe since alc_fixup_dell_wmi() will return an error in these situations. And remove the quirk for machine with subsystem id 0x6010 and 0x601f, these two machines will fall back to the quirk ALC255_FIXUP_DELL1_MIC_NO_PRESENCE-->ALC255_FIXUP_HEADSET_MODE--> ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED through pin_fixup_tbl[]. BugLink: https://bugs.launchpad.net/bugs/1381856 Reported-and-Tested-by: Po-Hsu Lin <po-hsu.lin(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1723954..9da5798 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4709,6 +4709,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC255_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, + .chained = true, + .chain_id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED }, [ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC] = { .type = HDA_FIXUP_FUNC, @@ -4744,8 +4746,6 @@ static const struct hda_fixup alc269_fixups[] = { [ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_wmi, - .chained_before = true, - .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, [ALC282_FIXUP_ASPIRE_V5_PINS] = { .type = HDA_FIXUP_PINS, @@ -4783,10 +4783,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), - SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: hda: separate stream tag for playback and capture
by Vinod Koul 18 Nov '14

18 Nov '14

From: Rafal Redzimski <rafal.f.redzimski(a)intel.com> According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. Newer Intel HW has the capability to provides a total of more than 15 stream DMA engines which with the current implementation causes an overflow on STRM field (and the whole register) which results in usage of reserved value 0 in the STRM field which confuses HDA controller. So make stream_tag assignment separate for input and output streams Signed-off-by: Rafal Redzimski <rafal.f.redzimski(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_controller.c | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8337645..703ac6e 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1921,11 +1921,17 @@ int azx_mixer_create(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return index < chip->playback_index_offset; +} /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1944,13 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout the stream direction + * group, valid values 1...15 + */ + azx_dev->stream_tag = is_input_stream(chip, i) ? + ++in_stream_tag : ++out_stream_tag; } return 0; -- 1.7.0.4

2 1

[alsa-devel] ASoC: omap-mcbsp: Deletion of an unnecessary check before the function call "kfree"
by SF Markus Elfring 18 Nov '14

18 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 14:05:27 +0100 The kfree() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/omap/mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 86c7538..68a1252 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -621,8 +621,7 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = NULL; spin_unlock(&mcbsp->lock); - if (reg_cache) - kfree(reg_cache); + kfree(reg_cache); } /* -- 2.1.3

3 2

[alsa-devel] [PATCH] ASoC: fsl_ssi: Remove comment about SSI running only in slave mode
by Fabio Estevam 18 Nov '14

18 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Current driver can also run in I2S master mode, so remove the old comment. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ab05262..1ef0c84 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,8 +67,6 @@ /** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI * - * This driver currently only supports the SSI running in I2S slave mode. - * * The SSI has a limitation in that the samples must be in the same byte * order as the host CPU. This is because when multiple bytes are written * to the STX register, the bytes and bits must be written in the same -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rt5677: Modify the default value of the MX-8E[4] for ASRC function
by Oder Chiou 18 Nov '14

18 Nov '14

Modify the default value of the MX-8E[4] to 1 for ASRC function. It could prevent the pop noise with ASRC function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index d659237..49ac9ba 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,7 +55,8 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, {RT5677_PR_BASE + 0x13, 0x0312}, {RT5677_PR_BASE + 0x1e, 0x0000}, @@ -173,7 +174,7 @@ static const struct reg_default rt5677_reg[] = { {RT5677_ASRC_9 , 0x0000}, {RT5677_ASRC_10 , 0x0000}, {RT5677_ASRC_11 , 0x0000}, - {RT5677_ASRC_12 , 0x0008}, + {RT5677_ASRC_12 , 0x0018}, {RT5677_ASRC_13 , 0x0000}, {RT5677_ASRC_14 , 0x0000}, {RT5677_ASRC_15 , 0x0000}, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] ASoC fixes for v3.18
by Mark Brown 17 Nov '14

17 Nov '14

The following changes since commit fc14f9c1272f62c3e8d01300f52467c0d9af50f9: Linux 3.18-rc5 (2014-11-16 16:36:20 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18-rc5 for you to fetch changes up to cf9a7f7823c67243da44da2ac47ca944a3108282: Merge remote-tracking branches 'asoc/fix/rt5670', 'asoc/fix/samsung' and 'asoc/fix/sgtl5000' into asoc-linus (2014-11-17 16:41:11 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.18 As well as the usual driver fixes there's a few other things here: One is a fix for a race in DPCM which is unfortuantely a rather large diffstat, this is the result of growing usage of the mainline code and hence more detailed testing so I'm relatively happy. The other is a fix for non-DT machine driver matching following some of the componentization work which is much more focused. Both have had a while to cook in -next. ---------------------------------------------------------------- Andreas Färber (1): ASoC: samsung: Add MODULE_DEVICE_TABLE for Snow Axel Lin (1): ASoC: es8328-i2c: Fix i2c_device_id name field in es8328_id Bard Liao (2): ASoC: rt5670: correct the incorrect default values ASoC: rt5670: change dapm routes of PLL connection Charles Keepax (1): ASoC: wm_adsp: Avoid attempt to free buffers that might still be in use Dylan Reid (1): ASoC: max98090: Correct pclk divisor settings Fabio Estevam (1): ASoC: sgtl5000: Fix SMALL_POP bit definition Fang, Yang A (1): ASoC: rt5645: Mark RT5645_TDM_CTRL_3 as readable Jianqun (1): ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_rxctrl Kuninori Morimoto (2): ASoC: fsi: remove unsupported PAUSE flag ASoC: rsnd: remove unsupported PAUSE flag Lars-Peter Clausen (1): ASoC: Fix snd_soc_find_dai() matching component by name Mark Brown (3): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/cs41l51', 'asoc/fix/dpcm', 'asoc/fix/es8328', 'asoc/fix/fsl-asrc', 'asoc/fix/max98090', 'asoc/fix/rcar', 'asoc/fix/rockchip' and 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5670', 'asoc/fix/samsung' and 'asoc/fix/sgtl5000' into asoc-linus Nicolin Chen (1): ASoC: fsl_asrc: Add reg_defaults for regmap to fix kernel dump Takashi Iwai (1): ASoC: dpcm: Fix race between FE/BE updates and trigger Thomas Petazzoni (1): ASoC: cs42l51: re-hook of_match_table pointer include/sound/soc-dpcm.h | 2 ++ sound/soc/codecs/cs42l51-i2c.c | 1 + sound/soc/codecs/cs42l51.c | 4 ++- sound/soc/codecs/cs42l51.h | 1 + sound/soc/codecs/es8328-i2c.c | 2 +- sound/soc/codecs/max98090.c | 6 ++-- sound/soc/codecs/rt5645.c | 2 ++ sound/soc/codecs/rt5670.c | 36 ++++++++++---------- sound/soc/codecs/sgtl5000.c | 3 +- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/wm_adsp.c | 1 + sound/soc/fsl/fsl_asrc.c | 26 ++++++++++++++ sound/soc/rockchip/rockchip_i2s.c | 4 ++- sound/soc/samsung/snow.c | 1 + sound/soc/sh/fsi.c | 3 +- sound/soc/sh/rcar/core.c | 3 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-pcm.c | 72 ++++++++++++++++++++++++++++++--------- 18 files changed, 123 insertions(+), 48 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: samsung-i2s: Add 192kHz config option
by Richard Fitzgerald 17 Nov '14

17 Nov '14

Adds QUIRK_SUPPORTS_192KHZ to allow 192kHz rate to be selected for hardware that supports it. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- .../devicetree/bindings/sound/samsung-i2s.txt | 2 ++ include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s.c | 8 ++++++++ 3 files changed, 11 insertions(+), 0 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index d188296..c3f3950 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -40,6 +40,8 @@ Optional SoC Specific Properties: sub system(used in secondary sound source). - pinctrl-0: Should specify pin control groups used for this controller. - pinctrl-names: Should contain only one value - "default". +- samsung,supports-192khz: specify this (without a value) if you want to allow + 192000 sample rate over the I2S link. Example: diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 5e0bc77..6d0702d 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -28,6 +28,7 @@ struct samsung_i2s { #define QUIRK_NEED_RSTCLR (1 << 3) #define QUIRK_SUPPORTS_TDM (1 << 4) #define QUIRK_SUPPORTS_IDMA (1 << 5) +#define QUIRK_SUPPORTS_192KHZ (1 << 6) /* Quirks of the I2S controller */ u32 quirks; dma_addr_t idma_addr; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 947352d..f688e89 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1219,6 +1219,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) "specified"); } } + + if (of_find_property(np, "samsung,supports-192khz", NULL)) + quirks |= QUIRK_SUPPORTS_192KHZ; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1247,6 +1250,11 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (quirks & QUIRK_PRI_6CHAN) pri_dai->i2s_dai_drv.playback.channels_max = 6; + if (quirks & QUIRK_SUPPORTS_192KHZ) { + pri_dai->i2s_dai_drv.playback.rates |= SNDRV_PCM_RATE_192000; + pri_dai->i2s_dai_drv.capture.rates |= SNDRV_PCM_RATE_192000; + } + if (quirks & QUIRK_SEC_DAI) { sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Avoid attempt to free buffers that might still be in use
by Charles Keepax 17 Nov '14

17 Nov '14

We should not free any buffers associated with writing out coefficients to the DSP until all the async writes have completed. This patch updates the out of memory path when allocating a new buffer to include a call to regmap_async_complete. Reported-by: JS Park <aitdark.park(a)samsung.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0a08ef5e..08d8ec3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1343,6 +1343,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) file, blocks, pos - firmware->size); out_fw: + regmap_async_complete(regmap); release_firmware(firmware); wm_adsp_buf_free(&buf_list); out: -- 1.7.2.5

2 1

[alsa-devel] [PATCH] usb-audio: Set the Control Selector to SU_SELECTOR_CONTROL for UAC2
by Johan Rastén 17 Nov '14

17 Nov '14

Specified in section 5.2.5.6.1 of the USB Audio Class 2.0 definition. Solves the following error for C-Media 6632A (Asus Xonar U7): [ 8219.676164] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 Signed-off-by: Johan Rastén <johan(a)oljud.se> --- sound/usb/mixer.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 14e1455..c6209ce 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2032,10 +2032,11 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, cval->res = 1; cval->initialized = 1; - if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) - cval->control = UAC2_CX_CLOCK_SELECTOR; - else + if (state->mixer->protocol == UAC_VERSION_1) cval->control = 0; + else /* UAC_VERSION_2 */ + cval->control = (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) ? + UAC2_CX_CLOCK_SELECTOR : UAC2_SU_SELECTOR; namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); if (!namelist) { -- 2.1.0

2 1

[alsa-devel] [PATCH (alsa-lib)] USB-audio: Add five more cards to IEC958 blacklist
by David Henningsson 17 Nov '14

17 Nov '14

These devices do not have any IEC958 outputs, so prevent them from being opened. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index a6754d9..05e5d09 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -38,10 +38,14 @@ USB-Audio.pcm.iec958_device { # "NoiseBlaster 3000" 42 # The below don't have digital in/out, so prevent them from being opened. + "Andrea PureAudio USB-SA Headset" 999 "Blue Snowball" 999 "HP Digital Stereo Headset" 999 + "GN 9330" 999 "Logitech Speaker Lapdesk N700" 999 + "Logitech G35 Headset" 999 "Logitech USB Headset" 999 + "Logitech USB Headset H540" 999 "Logitech Wireless Headset" 999 "Plantronics GameCom 780" 999 "Plantronics USB Headset" 999 @@ -50,6 +54,7 @@ USB-Audio.pcm.iec958_device { "Scarlett 2i2 USB" 999 "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 + "SWTOR Gaming Headset by Razer" 999 "USB Device 0x46d:0x992" 999 } -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] ALSA: powermac: Deletion of an unnecessary check before the function call "pci_dev_put"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 13:35:54 +0100 The pci_dev_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/ppc/pmac.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 8a431bc..5a13b22 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -887,8 +887,7 @@ static int snd_pmac_free(struct snd_pmac *chip) } } - if (chip->pdev) - pci_dev_put(chip->pdev); + pci_dev_put(chip->pdev); of_node_put(chip->node); kfree(chip); return 0; -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: hdsp: Deletion of an unnecessary check before the function call "release_firmware"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 13:04:14 +0100 The release_firmware() function tests whether its argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/rme9652/hdsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7646ba1..0ae568d 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5368,8 +5368,7 @@ static int snd_hdsp_free(struct hdsp *hdsp) snd_hdsp_free_buffers(hdsp); - if (hdsp->firmware) - release_firmware(hdsp->firmware); + release_firmware(hdsp->firmware); vfree(hdsp->fw_uploaded); if (hdsp->iobase) -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: lola: Deletion of an unnecessary check before the function call "vfree"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 12:42:16 +0100 The vfree() function performs also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/lola/lola_mixer.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index 782f4d8..e7fe15d 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -108,8 +108,7 @@ int lola_init_pins(struct lola *chip, int dir, int *nidp) void lola_free_mixer(struct lola *chip) { - if (chip->mixer.array_saved) - vfree(chip->mixer.array_saved); + vfree(chip->mixer.array_saved); } int lola_init_mixer_widget(struct lola *chip, int nid) -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: ice17xx: Deletion of unnecessary checks before the function call "snd_ac97_resume"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 11:28:02 +0100 The snd_ac97_resume() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ice1712/ice1712.c | 3 +-- sound/pci/ice1712/ice1724.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 87f7fc4..e1e18b5 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2905,8 +2905,7 @@ static int snd_ice1712_resume(struct device *dev) outw(ice->pm_saved_spdif_ctrl, ICEMT(ice, ROUTE_SPDOUT)); outw(ice->pm_saved_route, ICEMT(ice, ROUTE_PSDOUT03)); - if (ice->ac97) - snd_ac97_resume(ice->ac97); + snd_ac97_resume(ice->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 08cb08a..0e56835 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2884,8 +2884,7 @@ static int snd_vt1724_resume(struct device *dev) outb(ice->pm_saved_spdif_cfg, ICEREG1724(ice, SPDIF_CFG)); outl(ice->pm_saved_route, ICEMT1724(ice, ROUTE_PLAYBACK)); - if (ice->ac97) - snd_ac97_resume(ice->ac97); + snd_ac97_resume(ice->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda: Deletion of unnecessary checks before two function calls
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 10:44:33 +0100 The functions kfree() and release_firmware() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/hda_intel.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ec6a7d0..3fe8859 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -827,8 +827,7 @@ static void snd_hda_bus_free(struct hda_bus *bus) WARN_ON(!list_empty(&bus->codec_list)); if (bus->workq) flush_workqueue(bus->workq); - if (bus->unsol) - kfree(bus->unsol); + kfree(bus->unsol); if (bus->ops.private_free) bus->ops.private_free(bus); if (bus->workq) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a7f848..b4ec4e1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1129,8 +1129,7 @@ static int azx_free(struct azx *chip) pci_disable_device(chip->pci); kfree(chip->azx_dev); #ifdef CONFIG_SND_HDA_PATCH_LOADER - if (chip->fw) - release_firmware(chip->fw); + release_firmware(chip->fw); #endif if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(false); -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: pxa: prepare/unprepare clocks in pxa-ssp
by Dmitry Eremin-Solenikov 17 Nov '14

17 Nov '14

Change clk_enable/disable() calls to clk_prepare_enable() and clk_disable_unrepapre(). Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a8e0974..cbba063 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -97,7 +97,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); pxa_ssp_disable(ssp); } @@ -121,7 +121,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); } kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); @@ -136,7 +136,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1); @@ -144,7 +144,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->psp = __raw_readl(ssp->mmio_base + SSPSP); pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -154,7 +154,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); __raw_writel(sssr, ssp->mmio_base + SSSR); __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); @@ -165,7 +165,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) if (cpu_dai->active) pxa_ssp_enable(ssp); else - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -256,11 +256,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (ssp->type != PXA3xx_SSP) - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); if (ssp->type != PXA3xx_SSP) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); return 0; } -- 2.1.1

3 2

[alsa-devel] [PATCH] ASoC: Intel: add missing ACPI device table
by mengdong.lin＠intel.com 17 Nov '14

17 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The ACPI device table will generate the driver module alias for Intel audio devices enumerated from ACPI. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 2b1c5d9..b261821 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -342,6 +342,8 @@ static const struct acpi_device_id sst_acpi_ids[] = { { }, }; +MODULE_DEVICE_TABLE(acpi, sst_acpi_ids); + static struct platform_driver sst_acpi_driver = { .driver = { .name = "intel_sst_acpi", -- 1.9.1

3 2

[alsa-devel] [PATCH 1/3] ASoC: adav80x: Replace w->codec with snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 17 Nov '14

17 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adav80x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index ce3cdca..b67480f 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -212,7 +212,7 @@ static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); const char *clk; @@ -236,7 +236,7 @@ static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL; -- 1.8.0

2 5

[alsa-devel] TLV320AIC3106 + DM3730 soc doens't capture audio (no BLCK, no WLCK)
by Sergio Serrano 17 Nov '14

17 Nov '14

Dear members, Please, let me explain my setup: I'm using DM3730 soc <https://www.isee.biz/products/igep-processor-boards/igepv2-dm3730> with McBSP3 and I2C2 to drive audio and control to the TLV320AIC3106 codec (which it is acting as master). Currently I'm using 2.6.37 kernel version <http://git.isee.biz/?p=pub/scm/linux-omap-2.6.git;a=tree>. For reference I've used BBB Audio Cape <http://elinux.org/CircuitCo:Audio_Cape_RevA> as base design. Right now I've got playback working as expected (codec generates BCLK and WCLK). Whenever I try to record using "arecor,d -f cd -Dhw:1,0" it's only showing wave header but no audio is captured at all. I can not see any clocks running on the codec. How can I debug or go further with this issue? Any thoughts will be appreciated!! BTW this is the init code: static int TLV320AIC3106_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; printk(KERN_ERR "+tlv_hw_params(%p,%p) %p, %p, %p\n", substream, params, rtd, codec_dai, cpu_dai); // Set codec DAI configuration 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) { printk(KERN_ERR "tlv_hw_params: can't set codec DAI configuration\n"); return ret; } // Set cpu DAI configuration 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) { printk(KERN_ERR "tlv_hw_params: can't set cpu DAI configuration\n"); return ret; } // Set the codec system clock for DAC and ADC ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set codec system clock\n"); return ret; } printk(KERN_ERR "-tlv_hw_params\n"); return 0; } BR, Sergio

1 0

[alsa-devel] Pop and offsets at start of audio playback/record for SGTL5000 on i.MX28
by Craig McQueen 17 Nov '14

17 Nov '14

I'm testing the 3.14.19 kernel for i.MX28 EVK, which has an SGTL5000 CODEC. I've also tested on the 3.18-rc4 kernel and confirmed this issue still occurs. Playback When doing audio playback, I notice that the audio "fades in" over the first (approximately) 500 ms of playback. This is very noticeable and not ideal in certain applications (e.g. audio notifications that are of short duration). However, if audio playback starts while audio recording is already in-progress, then there is no fade-in; instead there is a loud pop. I assume the fade-in is there to mask the pop. The pop is very unpleasant through headphones. (However, if I do audio playback a _second_ time while audio recording is still in-progress, then there is no loud pop the second time.) I didn't think either the pop or the fade-in occurred on the 2.6.35 kernel. How feasible is it to improve the driver to avoid or reduce the pop, and remove the necessity for a fade-in? Record When recording, I notice there is a low-frequency ramping offset at the start of a recording (after the 400-500 ms of previous audio) of approx 600-700 ms duration. This also reduces the audio quality at the start of the recording. It could reduce the quality of captured audio depending on the application usage. I am getting distortion (clipping I assume) of my audio input during this time. It could cause a pop on playback (local playback or e.g. VoIP transmission to a remote device). It could reduce the audio quality through an audio CODEC. How feasible is it to improve the driver to avoid a ramping offset at the start of recording? -- Craig McQueen

3 6

[alsa-devel] What tree to base patches on?
by Johan Aires Rastén 16 Nov '14

16 Nov '14

If I want to send you a patch for snd_usb_audio, is there a git repo + branch or something else which I should base it on?

2 1

[alsa-devel] [PATCH v2] ASoC: Fix kerneldoc errors
by Takashi Iwai 16 Nov '14

16 Nov '14

Lots of typos and missing parameter descriptions. Some private struct fields are moved to the end of struct and put markers to make kerneldoc happy (and it make even easier for human readers). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 16 +++++++++++----- sound/soc/soc-core.c | 21 ++++++++++++++------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-devres.c | 2 +- sound/soc/soc-jack.c | 2 +- 5 files changed, 28 insertions(+), 15 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130037a0..10129be3cbb1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -588,10 +588,11 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, * @invert: if non-zero then pin is enabled when status is not reported */ struct snd_soc_jack_pin { - struct list_head list; const char *pin; int mask; bool invert; + /* private: */ + struct list_head list; }; /** @@ -602,13 +603,13 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container */ struct snd_soc_jack_zone { unsigned int min_mv; unsigned int max_mv; unsigned int jack_type; unsigned int debounce_time; + /* private: */ struct list_head list; }; @@ -618,16 +619,19 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debouce time in ms * @wake: enable as wake source + * @jack: assigned jack + * @desc: GPIO descriptor * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an * ADC). + * @data: arbitrary data passed to jack_status_check callback */ struct snd_soc_jack_gpio { unsigned int gpio; @@ -640,11 +644,13 @@ struct snd_soc_jack_gpio { bool wake; struct snd_soc_jack *jack; - struct delayed_work work; struct gpio_desc *desc; void *data; int (*jack_status_check)(void *data); + + /* private: */ + struct delayed_work work; }; struct snd_soc_jack { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a23a0e9..74a8a80e2d9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2113,6 +2113,8 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); /** * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * @ops: AC97 opts to set + * @pdev: platform to deal with * * This function sets the reset and warm_reset properties of ops and parses * the device node of pdev to get pinctrl states and gpio numbers to use. @@ -2685,7 +2687,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * @@ -3458,7 +3460,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -4082,6 +4084,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core + * @dev: device to unregister * */ void snd_soc_unregister_component(struct device *dev) @@ -4119,7 +4122,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -4158,7 +4161,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: platform to register + * @platform_drv: the driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -4215,7 +4219,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -4306,7 +4310,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: codec to register + * @codec_drv: the codec driver to register + * @dai_drv: the array of DAI drivers to register + * @num_dai: the number of DAIs to register */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -4406,7 +4413,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..7050ec1d5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2680,7 +2680,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card object containing widgets * * Checks the codec for any new dapm widgets and creates them if found. * diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 057e5ef7dcce..a57921eeee81 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res) /** * devm_snd_soc_register_platform - resource managed platform registration * @dev: Device used to manage platform - * @platform: platform to register + * @platform_drv: platform to register * * Register a platform driver with automatic unregistration when the device is * unregistered. diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..ef1d42d7c6f6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -116,7 +116,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * * @jack: ASoC jack * @count: Number of zones - * @zone: Array of zones + * @zones: Array of zones * * After this function has been called the zones specified in the * array will be associated with the jack. -- 2.1.3

2 7

[alsa-devel] [PATCH] usb-audio: Add ctrl message delay quirk for Marantz/Denon devices
by Jurgen Kramer 16 Nov '14

16 Nov '14

This patch adds a USB control message delay quirk for a few specific Marantz/Denon devices. Without the delay the DACs will not work properly and produces the following type of messages: Nov 15 10:09:21 orwell kernel: [ 91.342880] usb 3-13: clock source 41 is not valid, cannot use Nov 15 10:09:21 orwell kernel: [ 91.343775] usb 3-13: clock source 41 is not valid, cannot use There are likely other Marantz/Denon devices using the same USB module which exhibit the same problems. But as this cannot be verified I limited the patch to the devices I could test. The following two devices are covered by this path: - Marantz SA-14S1 - Marantz HD-DAC1 Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d2aa45a..a5941f8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1146,6 +1146,20 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); + + /* Marantz/Denon devices with USB DAC functionality need a delay + * after each class compliant request + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { + + switch (le16_to_cpu(dev->descriptor.idProduct)) { + case 0x3005: /* Marantz HD-DAC1 */ + case 0x3006: /* Marantz SA-14S1 */ + mdelay(20); + break; + } + } } /* -- 1.9.3

2 1

[alsa-devel] [PATCH] snd_dlsym: only do versioning checks if built --with-versioning
by John Spencer 16 Nov '14

16 Nov '14

From 291553a0132936611d48f41e924601dff8941537 Mon Sep 17 00:00:00 2001 From: John Spencer <maillist-alsa(a)barfooze.de> Date: Sat, 15 Nov 2014 13:41:00 +0100 Subject: [PATCH] snd_dlsym: only do versioning checks if built --with-versioning a combination of some of the following garbage collecting LD/CFLAGS -Os -g0 -fdata-sections -ffunction-sections -s -Wl,--gc-sections -fno-unwind-tables -fno-asynchronous-unwind-tables causes the symbol versioning marker symbols to be removed from BSS since they're otherwise unreferenced. this causes dlsym failing to find them which results in runtime breakage: $ alsamixer ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load ALSA lib conf.c:3328:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null) ALSA lib conf.c:3788:(snd_config_update_r) hooks failed, removing configuration cannot open mixer: No such device or address correct DSO: $ objdump -T libasound.so.2.strip | grep _snd_config_hook_load_dlsym_config_hook_001 001196bc g DO .bss 00000001 _snd_config_hook_load_dlsym_config_hook_001 incorrect DSO: $ objdump -T libs/libasound.so.2 | grep _snd_config_hook_load_dlsym_config_hook_001 00000000 g DO *ABS* 00000001 _snd_config_hook_load_dlsym_config_hook_001 since alsa was built with --without-versioning, doing those versioning checks at all is unnecessary and harmful and thus now disabled in this case. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/dlmisc.c | 2 ++ 1 file changed, 2 insertions(+) From 291553a0132936611d48f41e924601dff8941537 Mon Sep 17 00:00:00 2001 From: John Spencer <maillist-alsa(a)barfooze.de> Date: Sat, 15 Nov 2014 13:41:00 +0100 Subject: [PATCH] snd_dlsym: only do versioning checks if built --with-versioning a combination of some of the following garbage collecting LD/CFLAGS -Os -g0 -fdata-sections -ffunction-sections -s -Wl,--gc-sections -fno-unwind-tables -fno-asynchronous-unwind-tables causes the symbol versioning marker symbols to be removed from BSS since they're otherwise unreferenced. this causes dlsym failing to find them which results in runtime breakage: $ alsamixer ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load ALSA lib conf.c:3328:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null) ALSA lib conf.c:3788:(snd_config_update_r) hooks failed, removing configuration cannot open mixer: No such device or address correct DSO: $ objdump -T libasound.so.2.strip | grep _snd_config_hook_load_dlsym_config_hook_001 001196bc g DO .bss 00000001 _snd_config_hook_load_dlsym_config_hook_001 incorrect DSO: $ objdump -T libs/libasound.so.2 | grep _snd_config_hook_load_dlsym_config_hook_001 00000000 g DO *ABS* 00000001 _snd_config_hook_load_dlsym_config_hook_001 since alsa was built with --without-versioning, doing those versioning checks at all is unnecessary and harmful and thus now disabled in this case. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/dlmisc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/dlmisc.c b/src/dlmisc.c index 4b8a02c..62f7047 100644 --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -177,11 +177,13 @@ void *snd_dlsym(void *handle, const char *name, const char *version) } #endif #ifdef HAVE_LIBDL +#ifdef VERSIONED_SYMBOLS if (version) { err = snd_dlsym_verify(handle, name, version); if (err < 0) return NULL; } +#endif return dlsym(handle, name); #else return NULL; -- 1.8.4

2 1

[alsa-devel] DTS-HD over HDMI
by Kip Warner 15 Nov '14

15 Nov '14

Hey list, I apologize if this ought to have gone to alsa-user. I wasn't sure as a developer using API, but not developing ALSA itself, if it have been better there than here. I'd like to be able to stream DTS-HD Master Audio bits out the HDMI port in their compressed form, and not decompressed as LPCM. From what I can gather, this may be possible with the ALSA API, provided the HDMI port is >= 1.3 and the audio card supports high bit-rate (HBR). Is my thinking correct? -- Kip Warner -- Senior Software Engineer OpenPGP encrypted/signed mail preferred http://www.thevertigo.com

2 2

[alsa-devel] usb-audio: check_input_term (mixer.c) iterating to clock selectors?
by Johan Aires Rastén 15 Nov '14

15 Nov '14

Hi everybody! I've just started looking at ALSA because I want to get the mic working with my Asus Xonar U7. Some of you might have seen me on IRC as johanar. I'm wondering about check_input_term(...) in sound/usb/mixer.c, which for case UAC_INPUT_TERMINAL and UAC_VERSION_2 will recursively go into the clock source. Is this correct behaviour? For my card both mic and line will show up as "Unit 22" when I call "amixer contents" since the two inputs have the same unnamed clock source (they also share physical connector). However, if I comment out the recursive call they show up as "mic" and "line" as they're supposed to. Btw, if this is correct, couldn't we set id = d->bCSourceID and break; in the same way that case UAC_FEATURE_UNIT does? //Johan Rastén

1 0

[alsa-devel] [PATCH 1/2] spi: Fix mapping from vmalloc-ed buffer to scatter list
by Charles Keepax 14 Nov '14

14 Nov '14

We can only use page_address on memory that has been mapped using kmap, when the buffer passed to the SPI has been allocated by vmalloc the page has not necessarily been mapped through kmap. This means sometimes page_address will return NULL causing the pointer we pass to sg_set_buf to be invalid. As we only call page_address such that we can pass a virtual address to sg_set_buf which will then immediately call virt_to_page on it, fix this by calling sg_set_page directly rather then relying on the sg_set_buf helper. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/spi/spi.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/spi/spi.c b/drivers/spi/spi.c index ebcb33d..50f20f2 100644 --- a/drivers/spi/spi.c +++ b/drivers/spi/spi.c @@ -615,13 +615,13 @@ static int spi_map_buf(struct spi_master *master, struct device *dev, sg_free_table(sgt); return -ENOMEM; } - sg_buf = page_address(vm_page) + - ((size_t)buf & ~PAGE_MASK); + sg_set_page(&sgt->sgl[i], vm_page, + min, offset_in_page(buf)); } else { sg_buf = buf; + sg_set_buf(&sgt->sgl[i], sg_buf, min); } - sg_set_buf(&sgt->sgl[i], sg_buf, min); buf += min; len -= min; -- 1.7.2.5

2 7

[alsa-devel] Question on Compressed offload session
by gsantosh＠codeaurora.org 14 Nov '14

14 Nov '14

Hi All, The Question is for the compressed offload session. For a generic codec driver during the startup function it will set some of the hw_constraints rule similarly like this. snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_12_24); pcm_lib.c will try to add the rule to the runtime structure by accessing the pointers which will be initialized during opening of the session, as The Constraints added by the codec driver will be updated in the struct snd_pcm_hw_constraints of runtime structure which will be part of substream handle. But for the compressed offload I do not see the initialization done for HW constraints, as done in pcm session 2092int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, 2093 struct file *file, 2094 struct snd_pcm_substream **rsubstream) most of the existing drivers which has the hw_constraint_list code will not be applicable for compress offload session, how to solve this? Regards, Santosh M G. Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,a Linux Foundation Collaborative Project

3 4

[alsa-devel] (no subject)
by Angelo Dureghello 14 Nov '14

14 Nov '14

unsubscribe alsa-devel angelo70(a)gmail.com -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH 00/15] new locomo driver
by Dmitry Eremin-Solenikov 14 Nov '14

14 Nov '14

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. Current drivers present in Linux kernel has some problems: * It uses custom bus instead of platform bus/mfd core. * Device drivers are not well layered/separated. * It uses custom gpio accessors instead of using GPIOLIB. With this patchset I tried to modernise and restructure the LoCoMo driver. Also I added an experimental SPI driver that will be used by mmc-spi on Sharp SL-5500 collie devices. ---------------------------------------------------------------- Dmitry Eremin-Solenikov (15): mfd: add new driver for Sharp LoCoMo GPIO: port LoCoMo gpio support from old driver leds: port locomo leds driver to new locomo core input: convert LoCoMo keyboard driver to use new locomo core video: backlight: add new locomo backlight driver video: lcd: add LoCoMo LCD driver video: backlight: drop old locomo bl/lcd driver ARM: sa1100: make collie use new locomo drivers ARM: sa1100: don't preallocate IRQ space for locomo ARM: pxa: poodle: use new LoCoMo driver sound: soc: poodle: make use of new locomo GPIO interface ARM: pxa: poodle: don't preallocate IRQ space for locomo ARM: drop old LoCoMo driver gpio: locomo: implement per-pin irq handling spi: add locomo SPI driver arch/arm/common/Kconfig | 3 - arch/arm/common/Makefile | 1 - arch/arm/common/locomo.c | 914 --------------------- arch/arm/mach-pxa/Kconfig | 1 - arch/arm/mach-pxa/include/mach/poodle.h | 16 +- arch/arm/mach-pxa/poodle.c | 23 +- arch/arm/mach-sa1100/Kconfig | 1 - arch/arm/mach-sa1100/collie.c | 112 +-- arch/arm/mach-sa1100/include/mach/collie.h | 25 +- arch/arm/mach-sa1100/include/mach/irqs.h | 19 +- drivers/gpio/Kconfig | 7 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-locomo.c | 386 +++++++++ drivers/input/keyboard/Kconfig | 1 - drivers/input/keyboard/locomokbd.c | 165 ++-- drivers/leds/Kconfig | 1 - drivers/leds/leds-locomo.c | 104 ++- drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/locomo.c | 644 +++++++++++++++ drivers/spi/Kconfig | 8 + drivers/spi/Makefile | 1 + drivers/spi/spi-locomo.c | 370 +++++++++ drivers/video/backlight/Kconfig | 14 +- drivers/video/backlight/Makefile | 3 +- drivers/video/backlight/locomo_bl.c | 171 ++++ drivers/video/backlight/locomo_lcd.c | 224 +++++ drivers/video/backlight/locomolcd.c | 255 ------ .../asm/hardware => include/linux/mfd}/locomo.h | 148 ++-- sound/soc/pxa/poodle.c | 51 +- 30 files changed, 2185 insertions(+), 1493 deletions(-) delete mode 100644 arch/arm/common/locomo.c create mode 100644 drivers/gpio/gpio-locomo.c create mode 100644 drivers/mfd/locomo.c create mode 100644 drivers/spi/spi-locomo.c create mode 100644 drivers/video/backlight/locomo_bl.c create mode 100644 drivers/video/backlight/locomo_lcd.c delete mode 100644 drivers/video/backlight/locomolcd.c rename {arch/arm/include/asm/hardware => include/linux/mfd}/locomo.h (55%)

9 53

[alsa-devel] [PATCH 1/4] ASoC: rt5670: make bias level more reasonable
by bardliao＠realtek.com 14 Nov '14

14 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index c4c42d5..85dc210 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2309,6 +2309,8 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2330,16 +2332,27 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); - snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); - snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); - snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); - snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); snd_soc_update_bits(codec, RT5670_PWR_ANLG1, RT5670_LDO_SEL_MASK, 0x1); break; + case SND_SOC_BIAS_OFF: + if (rt5670->pdata.jd_mode) + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_MB | RT5670_PWR_BG); + else + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + break; default: break; -- 1.8.1.1.439.g50a6b54

4 7

[alsa-devel] [PATCH v4] ALSA: ice1712: consider error value
by Sudip Mukherjee 14 Nov '14

14 Nov '14

earlier we were ignoring the return value of snd_ak4114_create and always returning 0. now we are returning the actual status. revo_init is calling this function, and revo_init is checking the return value. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v3: spec->ak4114 is now dereferenced if we dont get an error value from snd_ak4114_create. change in v4: error value returned according to coding style. sound/pci/ice1712/revo.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1112ec1..1d81ae6 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -494,11 +494,13 @@ static int ap192_ak4114_init(struct snd_ice1712 *ice) ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &spec->ak4114); + if (err < 0) + return err; /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; - return 0; /* error ignored; it's no fatal error */ + return 0; } static int revo_init(struct snd_ice1712 *ice) -- 1.8.1.2

2 4

[alsa-devel] [PATCH v3] ALSA: ice1712: consider error value
by Sudip Mukherjee 14 Nov '14

14 Nov '14

earlier we were ignoring the return value of snd_ak4114_create and always returning 0. now we are returning the actual status. revo_init is calling this function, and revo_init is checking the return value. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v3: spec->ak4114 is now dereferenced if we dont get an error value from snd_ak4114_create. sound/pci/ice1712/revo.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1112ec1..24a7c6a 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -496,9 +496,10 @@ static int ap192_ak4114_init(struct snd_ice1712 *ice) ice, &spec->ak4114); /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ - spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; + if (err == 0) + spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; - return 0; /* error ignored; it's no fatal error */ + return err; } static int revo_init(struct snd_ice1712 *ice) -- 1.8.1.2

2 1

[alsa-devel] [PATCH v4] ASoC: Intel: mrfld- add ACPI module
by Vinod Koul 14 Nov '14

14 Nov '14

Add the last ACPI module support which also uses core module like the PCI part Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 5 + sound/soc/intel/sst/Makefile | 4 +- sound/soc/intel/sst/sst.c | 22 ++- sound/soc/intel/sst/sst.h | 1 + sound/soc/intel/sst/sst_acpi.c | 362 ++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 6 files changed, 391 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/sst/sst_acpi.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c963a5d..a992e85 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -20,6 +20,11 @@ config SND_SST_IPC_PCI tristate select SND_SST_IPC +config SND_SST_IPC_ACPI + tristate + select SND_SST_IPC + depends on ACPI + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI diff --git a/sound/soc/intel/sst/Makefile b/sound/soc/intel/sst/Makefile index b8aa1d3..fd21726 100644 --- a/sound/soc/intel/sst/Makefile +++ b/sound/soc/intel/sst/Makefile @@ -1,7 +1,7 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o snd-intel-sst-pci-objs += sst_pci.o - +snd-intel-sst-acpi-objs += sst_acpi.o obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o - +obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index b97c231..b2b5604 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -26,6 +26,7 @@ #include <linux/pm_runtime.h> #include <linux/pm_qos.h> #include <linux/async.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/soc.h> #include <asm/platform_sst_audio.h> @@ -181,6 +182,7 @@ int sst_driver_ops(struct intel_sst_drv *sst) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: sst->tstamp = SST_TIME_STAMP_MRFLD; sst->ops = &mrfld_ops; return 0; @@ -323,7 +325,7 @@ EXPORT_SYMBOL_GPL(sst_context_init); void sst_context_cleanup(struct intel_sst_drv *ctx) { pm_runtime_get_noresume(ctx->dev); - pm_runtime_forbid(ctx->dev); + pm_runtime_disable(ctx->dev); sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); @@ -371,8 +373,19 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { pm_runtime_set_autosuspend_delay(ctx->dev, SST_SUSPEND_DELAY); pm_runtime_use_autosuspend(ctx->dev); - pm_runtime_allow(ctx->dev); - pm_runtime_put_noidle(ctx->dev); + /* + * For acpi devices, the actual physical device state is + * initially active. So change the state to active before + * enabling the pm + */ + if (acpi_disabled) { + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); + } else { + pm_runtime_allow(ctx->dev); + pm_runtime_put_noidle(ctx->dev); + } + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); } EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); @@ -395,6 +408,9 @@ static int intel_sst_runtime_suspend(struct device *dev) synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); + /* save the shim registers because PMC doesn't save state */ + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); + return ret; } diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 2dcbf47..683dc71 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -29,6 +29,7 @@ /* driver names */ #define SST_DRV_NAME "intel_sst_driver" #define SST_MRFLD_PCI_ID 0x119A +#define SST_BYT_ACPI_ID 0x80860F28 #define SST_SUSPEND_DELAY 2000 #define FW_CONTEXT_MEM (64*1024) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c new file mode 100644 index 0000000..2b1c5d9 --- /dev/null +++ b/sound/soc/intel/sst/sst_acpi.c @@ -0,0 +1,362 @@ +/* + * sst_acpi.c - SST (LPE) driver init file for ACPI enumeration. + * + * Copyright (c) 2013, Intel Corporation. + * + * Authors: Ramesh Babu K V <Ramesh.Babu(a)intel.com> + * Authors: Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * + */ + +#include <linux/module.h> +#include <linux/fs.h> +#include <linux/interrupt.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/miscdevice.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/pm_runtime.h> +#include <linux/pm_qos.h> +#include <linux/acpi.h> +#include <asm/platform_sst_audio.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/compress_driver.h> +#include <acpi/acbuffer.h> +#include <acpi/platform/acenv.h> +#include <acpi/platform/aclinux.h> +#include <acpi/actypes.h> +#include <acpi/acpi_bus.h> +#include "../sst-mfld-platform.h" +#include "../sst-dsp.h" +#include "sst.h" + +struct sst_machines { + char codec_id[32]; + char board[32]; + char machine[32]; + void (*machine_quirk)(void); + char firmware[32]; + struct sst_platform_info *pdata; + +}; + +/* LPE viewpoint addresses */ +#define SST_BYT_IRAM_PHY_START 0xff2c0000 +#define SST_BYT_IRAM_PHY_END 0xff2d4000 +#define SST_BYT_DRAM_PHY_START 0xff300000 +#define SST_BYT_DRAM_PHY_END 0xff320000 +#define SST_BYT_IMR_VIRT_START 0xc0000000 /* virtual addr in LPE */ +#define SST_BYT_IMR_VIRT_END 0xc01fffff +#define SST_BYT_SHIM_PHY_ADDR 0xff340000 +#define SST_BYT_MBOX_PHY_ADDR 0xff344000 +#define SST_BYT_DMA0_PHY_ADDR 0xff298000 +#define SST_BYT_DMA1_PHY_ADDR 0xff29c000 +#define SST_BYT_SSP0_PHY_ADDR 0xff2a0000 +#define SST_BYT_SSP2_PHY_ADDR 0xff2a2000 + +#define BYT_FW_MOD_TABLE_OFFSET 0x80000 +#define BYT_FW_MOD_TABLE_SIZE 0x100 +#define BYT_FW_MOD_OFFSET (BYT_FW_MOD_TABLE_OFFSET + BYT_FW_MOD_TABLE_SIZE) + +static const struct sst_info byt_fwparse_info = { + .use_elf = false, + .max_streams = 25, + .iram_start = SST_BYT_IRAM_PHY_START, + .iram_end = SST_BYT_IRAM_PHY_END, + .iram_use = true, + .dram_start = SST_BYT_DRAM_PHY_START, + .dram_end = SST_BYT_DRAM_PHY_END, + .dram_use = true, + .imr_start = SST_BYT_IMR_VIRT_START, + .imr_end = SST_BYT_IMR_VIRT_END, + .imr_use = true, + .mailbox_start = SST_BYT_MBOX_PHY_ADDR, + .num_probes = 0, + .lpe_viewpt_rqd = true, +}; + +static const struct sst_ipc_info byt_ipc_info = { + .ipc_offset = 0, + .mbox_recv_off = 0x400, +}; + +static const struct sst_lib_dnld_info byt_lib_dnld_info = { + .mod_base = SST_BYT_IMR_VIRT_START, + .mod_end = SST_BYT_IMR_VIRT_END, + .mod_table_offset = BYT_FW_MOD_TABLE_OFFSET, + .mod_table_size = BYT_FW_MOD_TABLE_SIZE, + .mod_ddr_dnld = false, +}; + +static const struct sst_res_info byt_rvp_res_info = { + .shim_offset = 0x140000, + .shim_size = 0x000100, + .shim_phy_addr = SST_BYT_SHIM_PHY_ADDR, + .ssp0_offset = 0xa0000, + .ssp0_size = 0x1000, + .dma0_offset = 0x98000, + .dma0_size = 0x4000, + .dma1_offset = 0x9c000, + .dma1_size = 0x4000, + .iram_offset = 0x0c0000, + .iram_size = 0x14000, + .dram_offset = 0x100000, + .dram_size = 0x28000, + .mbox_offset = 0x144000, + .mbox_size = 0x1000, + .acpi_lpe_res_index = 0, + .acpi_ddr_index = 2, + .acpi_ipc_irq_index = 5, +}; + +struct sst_platform_info byt_rvp_platform_data = { + .probe_data = &byt_fwparse_info, + .ipc_info = &byt_ipc_info, + .lib_info = &byt_lib_dnld_info, + .res_info = &byt_rvp_res_info, + .platform = "sst-mfld-platform", +}; + +static int sst_platform_get_resources(struct intel_sst_drv *ctx) +{ + struct resource *rsrc; + struct platform_device *pdev = to_platform_device(ctx->dev); + + /* All ACPI resource request here */ + /* Get Shim addr */ + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + ctx->pdata->res_info->acpi_lpe_res_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid SHIM base from IFWI"); + return -EIO; + } + dev_info(ctx->dev, "LPE base: %#x size:%#x", (unsigned int) rsrc->start, + (unsigned int)resource_size(rsrc)); + + ctx->iram_base = rsrc->start + ctx->pdata->res_info->iram_offset; + ctx->iram_end = ctx->iram_base + ctx->pdata->res_info->iram_size - 1; + dev_info(ctx->dev, "IRAM base: %#x", ctx->iram_base); + ctx->iram = devm_ioremap_nocache(ctx->dev, ctx->iram_base, + ctx->pdata->res_info->iram_size); + if (!ctx->iram) { + dev_err(ctx->dev, "unable to map IRAM"); + return -EIO; + } + + ctx->dram_base = rsrc->start + ctx->pdata->res_info->dram_offset; + ctx->dram_end = ctx->dram_base + ctx->pdata->res_info->dram_size - 1; + dev_info(ctx->dev, "DRAM base: %#x", ctx->dram_base); + ctx->dram = devm_ioremap_nocache(ctx->dev, ctx->dram_base, + ctx->pdata->res_info->dram_size); + if (!ctx->dram) { + dev_err(ctx->dev, "unable to map DRAM"); + return -EIO; + } + + ctx->shim_phy_add = rsrc->start + ctx->pdata->res_info->shim_offset; + dev_info(ctx->dev, "SHIM base: %#x", ctx->shim_phy_add); + ctx->shim = devm_ioremap_nocache(ctx->dev, ctx->shim_phy_add, + ctx->pdata->res_info->shim_size); + if (!ctx->shim) { + dev_err(ctx->dev, "unable to map SHIM"); + return -EIO; + } + + /* reassign physical address to LPE viewpoint address */ + ctx->shim_phy_add = ctx->pdata->res_info->shim_phy_addr; + + /* Get mailbox addr */ + ctx->mailbox_add = rsrc->start + ctx->pdata->res_info->mbox_offset; + dev_info(ctx->dev, "Mailbox base: %#x", ctx->mailbox_add); + ctx->mailbox = devm_ioremap_nocache(ctx->dev, ctx->mailbox_add, + ctx->pdata->res_info->mbox_size); + if (!ctx->mailbox) { + dev_err(ctx->dev, "unable to map mailbox"); + return -EIO; + } + + /* reassign physical address to LPE viewpoint address */ + ctx->mailbox_add = ctx->info.mailbox_start; + + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + ctx->pdata->res_info->acpi_ddr_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid DDR base from IFWI"); + return -EIO; + } + ctx->ddr_base = rsrc->start; + ctx->ddr_end = rsrc->end; + dev_info(ctx->dev, "DDR base: %#x", ctx->ddr_base); + ctx->ddr = devm_ioremap_nocache(ctx->dev, ctx->ddr_base, + resource_size(rsrc)); + if (!ctx->ddr) { + dev_err(ctx->dev, "unable to map DDR"); + return -EIO; + } + + /* Find the IRQ */ + ctx->irq_num = platform_get_irq(pdev, + ctx->pdata->res_info->acpi_ipc_irq_index); + return 0; +} + +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_machines *sst_acpi_find_machine( + struct sst_machines *machines) +{ + struct sst_machines *mach; + bool found = false; + + for (mach = machines; mach->codec_id; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->codec_id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + +int sst_acpi_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + int ret = 0; + struct intel_sst_drv *ctx; + const struct acpi_device_id *id; + struct sst_machines *mach; + struct platform_device *mdev; + struct platform_device *plat_dev; + unsigned int dev_id; + + id = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!id) + return -ENODEV; + dev_dbg(dev, "for %s", id->id); + + mach = (struct sst_machines *)id->driver_data; + mach = sst_acpi_find_machine(mach); + if (mach == NULL) { + dev_err(dev, "No matching machine driver found\n"); + return -ENODEV; + } + + ret = kstrtouint(id->id, 16, &dev_id); + if (ret < 0) { + dev_err(dev, "Unique device id conversion error: %d\n", ret); + return ret; + } + + dev_dbg(dev, "ACPI device id: %x\n", dev_id); + + plat_dev = platform_device_register_data(dev, mach->pdata->platform, -1, NULL, 0); + if (plat_dev == NULL) { + dev_err(dev, "Failed to create machine device: %s\n", mach->pdata->platform); + return -ENODEV; + } + + /* Create platform device for sst machine driver */ + mdev = platform_device_register_data(dev, mach->machine, -1, NULL, 0); + if (mdev == NULL) { + dev_err(dev, "Failed to create machine device: %s\n", mach->machine); + return -ENODEV; + } + + ret = sst_alloc_drv_context(&ctx, dev, dev_id); + if (ret < 0) + return ret; + + /* Fill sst platform data */ + ctx->pdata = mach->pdata; + strcpy(ctx->firmware_name, mach->firmware); + + ret = sst_platform_get_resources(ctx); + if (ret) + return ret; + + ret = sst_context_init(ctx); + if (ret < 0) + return ret; + + /* need to save shim registers in BYT */ + ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), + GFP_KERNEL); + if (!ctx->shim_regs64) { + return -ENOMEM; + goto do_sst_cleanup; + } + + sst_configure_runtime_pm(ctx); + platform_set_drvdata(pdev, ctx); + return ret; + +do_sst_cleanup: + sst_context_cleanup(ctx); + platform_set_drvdata(pdev, NULL); + dev_err(ctx->dev, "failed with %d\n", ret); + return ret; +} + +/** +* intel_sst_remove - remove function +* +* @pdev: platform device structure +* +* This function is called by OS when a device is unloaded +* This frees the interrupt etc +*/ +int sst_acpi_remove(struct platform_device *pdev) +{ + struct intel_sst_drv *ctx; + + ctx = platform_get_drvdata(pdev); + sst_context_cleanup(ctx); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct sst_machines sst_acpi_bytcr[] = { + {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + &byt_rvp_platform_data }, + {}, +}; + +static const struct acpi_device_id sst_acpi_ids[] = { + { "80860F28", (unsigned long)&sst_acpi_bytcr}, + { }, +}; + +static struct platform_driver sst_acpi_driver = { + .driver = { + .name = "intel_sst_acpi", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(sst_acpi_ids), + .pm = &intel_sst_pm, + }, + .probe = sst_acpi_probe, + .remove = sst_acpi_remove, +}; + +module_platform_driver(sst_acpi_driver); + +MODULE_DESCRIPTION("Intel (R) SST(R) Audio Engine ACPI Driver"); +MODULE_AUTHOR("Ramesh Babu K V"); +MODULE_AUTHOR("Omair Mohammed Abdullah"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("sst"); diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/sst/sst_pvt.c index 9a5df19..4b77208 100644 --- a/sound/soc/intel/sst/sst_pvt.c +++ b/sound/soc/intel/sst/sst_pvt.c @@ -117,6 +117,7 @@ unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: val = sst_shim_read64(sst->shim, addr); break; } @@ -128,6 +129,7 @@ void write_shim_data(struct intel_sst_drv *sst, int addr, { switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: sst_shim_write64(sst->shim, addr, (u64) data); break; } -- 1.7.0.4

2 1

[alsa-devel] [PATCH v2 1/3] ALSA: ice1712: remove unneeded return statement
by Sudip Mukherjee 14 Nov '14

14 Nov '14

the functions: snd_ice1712_akm4xxx_build_controls snd_ice1712_build_pro_mixer snd_ctl_add snd_ak4114_build prodigy192_ak4114_init snd_ak4113_build are all returning either 0 or a negetive error value. so we can easily remove the check for a negative value and return the value instead. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v2: returning the return value of the function directly. sound/pci/ice1712/hoontech.c | 6 +----- sound/pci/ice1712/ice1712.c | 19 ++++++------------- sound/pci/ice1712/ice1724.c | 7 ++----- sound/pci/ice1712/juli.c | 5 +---- sound/pci/ice1712/prodigy192.c | 4 +--- sound/pci/ice1712/quartet.c | 5 +---- 6 files changed, 12 insertions(+), 34 deletions(-) diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index 59e37c5..a40001c 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -309,11 +309,7 @@ static int snd_ice1712_value_init(struct snd_ice1712 *ice) return err; /* ak4524 controls */ - err = snd_ice1712_akm4xxx_build_controls(ice); - if (err < 0) - return err; - - return 0; + return snd_ice1712_akm4xxx_build_controls(ice); } static int snd_ice1712_ez8_init(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 48a0c33..5975334 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1295,10 +1295,7 @@ static int snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd return err; } - err = snd_ice1712_build_pro_mixer(ice); - if (err < 0) - return err; - return 0; + return snd_ice1712_build_pro_mixer(ice); } /* @@ -1545,10 +1542,9 @@ static int snd_ice1712_ac97_mixer(struct snd_ice1712 *ice) dev_warn(ice->card->dev, "cannot initialize ac97 for consumer, skipped\n"); else { - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice)); - if (err < 0) - return err; - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, + ice)); } } @@ -2497,11 +2493,8 @@ static int snd_ice1712_build_controls(struct snd_ice1712 *ice) err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_volume_rate, ice)); if (err < 0) return err; - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); } static int snd_ice1712_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index f633e3b..ea53167 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2497,11 +2497,8 @@ static int snd_vt1724_build_controls(struct snd_ice1712 *ice) return err; } - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); } static int snd_vt1724_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 7a6c078..a1536c1 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -475,11 +475,8 @@ static int juli_add_controls(struct snd_ice1712 *ice) return err; /* only capture SPDIF over AK4114 */ - err = snd_ak4114_build(spec->ak4114, NULL, + return snd_ak4114_build(spec->ak4114, NULL, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; } /* diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 1eb151aa..3919aed 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -758,10 +758,8 @@ static int prodigy192_init(struct snd_ice1712 *ice) "AK4114 initialized with status %d\n", err); } else dev_dbg(ice->card->dev, "AK4114 not found\n"); - if (err < 0) - return err; - return 0; + return err; } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d4caf9d..6f55e02 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -833,11 +833,8 @@ static int qtet_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; /* only capture SPDIF over AK4113 */ - err = snd_ak4113_build(spec->ak4113, + return snd_ak4113_build(spec->ak4113, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; } static inline int qtet_is_spdif_master(struct snd_ice1712 *ice) -- 1.8.1.2

2 7

[alsa-devel] [PATCH v3] ASoC: add TI ts3a227e headset chip driver
by Dylan Reid 14 Nov '14

14 Nov '14

The TS3A227E is an autonomous audio accessory detection and configuration switch that detects 3-pole or 4-pole audio accessories and configures internal switches to route the signals accordingly. This chip also has built-in support for the new button standard described in the Android "Wired audio headset specification" v1.0. These buttons will be reported on the jack as buttons 0-3 mapped to KEY_MEDIA, KEY_VOLUMEUP, KEY_VOLUMEDOWN, and KEY_VOICE_COMMAND. This will be added as an aux_dev and have the jack passed in from the machine driver. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v2: Switch from codec to component driver Changes since v1: Use button to key mapping instead of separate input device --- .../devicetree/bindings/sound/ts3a227e.txt | 26 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ts3a227e.c | 314 +++++++++++++++++++++ sound/soc/codecs/ts3a227e.h | 17 ++ 5 files changed, 364 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt new file mode 100644 index 0000000..e8bf23e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -0,0 +1,26 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupt-parent: The parent interrupt controller + - interrupts: Interrupt number for /INT pin from the 227e + + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..243ec86 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C + select SND_SOC_TS3A227E if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X @@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate +config SND_SOC_TS3A227E + tristate "TI Headset/Mic detect and keypress chip" + depends on I2C + config SND_SOC_TWL4030 select MFD_TWL4030_AUDIO tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..a1eb7ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o @@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c new file mode 100644 index 0000000..1d12057 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.c @@ -0,0 +1,314 @@ +/* + * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +struct ts3a227e { + struct regmap *regmap; + struct snd_soc_jack *jack; + bool plugged; + bool mic_present; + unsigned int buttons_held; +}; + +/* Button values to be reported on the jack */ +static const int ts3a227e_buttons[] = { + SND_JACK_BTN_0, + SND_JACK_BTN_1, + SND_JACK_BTN_2, + SND_JACK_BTN_3, +}; + +#define TS3A227E_NUM_BUTTONS 4 +#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_MICROPHONE | \ + SND_JACK_BTN_0 | \ + SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | \ + SND_JACK_BTN_3) + +/* TS3A227E registers */ +#define TS3A227E_REG_DEVICE_ID 0x00 +#define TS3A227E_REG_INTERRUPT 0x01 +#define TS3A227E_REG_KP_INTERRUPT 0x02 +#define TS3A227E_REG_INTERRUPT_DISABLE 0x03 +#define TS3A227E_REG_SETTING_1 0x04 +#define TS3A227E_REG_SETTING_2 0x05 +#define TS3A227E_REG_SETTING_3 0x06 +#define TS3A227E_REG_SWITCH_CONTROL_1 0x07 +#define TS3A227E_REG_SWITCH_CONTROL_2 0x08 +#define TS3A227E_REG_SWITCH_STATUS_1 0x09 +#define TS3A227E_REG_SWITCH_STATUS_2 0x0a +#define TS3A227E_REG_ACCESSORY_STATUS 0x0b +#define TS3A227E_REG_ADC_OUTPUT 0x0c +#define TS3A227E_REG_KP_THRESHOLD_1 0x0d +#define TS3A227E_REG_KP_THRESHOLD_2 0x0e +#define TS3A227E_REG_KP_THRESHOLD_3 0x0f + +/* TS3A227E_REG_INTERRUPT 0x01 */ +#define INS_REM_EVENT 0x01 +#define DETECTION_COMPLETE_EVENT 0x02 + +/* TS3A227E_REG_KP_INTERRUPT 0x02 */ +#define PRESS_MASK(idx) (0x01 << (2 * (idx))) +#define RELEASE_MASK(idx) (0x02 << (2 * (idx))) + +/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */ +#define INS_REM_INT_DISABLE 0x01 +#define DETECTION_COMPLETE_INT_DISABLE 0x02 +#define ADC_COMPLETE_INT_DISABLE 0x04 +#define INTB_DISABLE 0x08 + +/* TS3A227E_REG_SETTING_2 0x05 */ +#define KP_ENABLE 0x04 + +/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ +#define TYPE_3_POLE 0x01 +#define TYPE_4_POLE_OMTP 0x02 +#define TYPE_4_POLE_STANDARD 0x04 +#define JACK_INSERTED 0x08 +#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD) + +static const struct reg_default ts3a227e_reg_defaults[] = { + { TS3A227E_REG_DEVICE_ID, 0x10 }, + { TS3A227E_REG_INTERRUPT, 0x00 }, + { TS3A227E_REG_KP_INTERRUPT, 0x00 }, + { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 }, + { TS3A227E_REG_SETTING_1, 0x23 }, + { TS3A227E_REG_SETTING_2, 0x00 }, + { TS3A227E_REG_SETTING_3, 0x0e }, + { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 }, + { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 }, + { TS3A227E_REG_SWITCH_STATUS_1, 0x0c }, + { TS3A227E_REG_SWITCH_STATUS_2, 0x00 }, + { TS3A227E_REG_ACCESSORY_STATUS, 0x00 }, + { TS3A227E_REG_ADC_OUTPUT, 0x00 }, + { TS3A227E_REG_KP_THRESHOLD_1, 0x20 }, + { TS3A227E_REG_KP_THRESHOLD_2, 0x40 }, + { TS3A227E_REG_KP_THRESHOLD_3, 0x68 }, +}; + +static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2: + case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: + case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: + return true; + default: + return false; + } +} + +static void ts3a227e_jack_report(struct ts3a227e *ts3a227e) +{ + unsigned int i; + int report = 0; + + if (!ts3a227e->jack) + return; + + if (ts3a227e->plugged) + report = SND_JACK_HEADPHONE; + if (ts3a227e->mic_present) + report |= SND_JACK_MICROPHONE; + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (ts3a227e->buttons_held & (1 << i)) + report |= ts3a227e_buttons[i]; + } + snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK); +} + +static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg) +{ + bool plugged, mic_present; + + plugged = !!(acc_reg & JACK_INSERTED); + mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK); + + ts3a227e->plugged = plugged; + + if (mic_present != ts3a227e->mic_present) { + ts3a227e->mic_present = mic_present; + ts3a227e->buttons_held = 0; + if (mic_present) { + /* Enable key press detection. */ + regmap_update_bits(ts3a227e->regmap, + TS3A227E_REG_SETTING_2, + KP_ENABLE, KP_ENABLE); + } + } +} + +static irqreturn_t ts3a227e_interrupt(int irq, void *data) +{ + struct ts3a227e *ts3a227e = (struct ts3a227e *)data; + struct regmap *regmap = ts3a227e->regmap; + unsigned int int_reg, kp_int_reg, acc_reg, i; + + /* Check for plug/unplug. */ + regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { + regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + } + + /* Report any key events. */ + regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (kp_int_reg & PRESS_MASK(i)) + ts3a227e->buttons_held |= (1 << i); + if (kp_int_reg & RELEASE_MASK(i)) + ts3a227e->buttons_held &= ~(1 << i); + } + + ts3a227e_jack_report(ts3a227e); + + return IRQ_HANDLED; +} + +/** + * ts3a227e_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events 0-3 will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int ts3a227e_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ts3a227e->jack = jack; + ts3a227e_jack_report(ts3a227e); + + return 0; +} +EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); + +static struct snd_soc_component_driver ts3a227e_soc_driver; + +static const struct regmap_config ts3a227e_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = TS3A227E_REG_KP_THRESHOLD_3, + .readable_reg = ts3a227e_readable_reg, + .writeable_reg = ts3a227e_writeable_reg, + .volatile_reg = ts3a227e_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ts3a227e_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), +}; + +static int ts3a227e_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ts3a227e *ts3a227e; + struct device *dev = &i2c->dev; + int ret; + + ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); + if (ts3a227e == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ts3a227e); + + ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); + if (IS_ERR(ts3a227e->regmap)) + return PTR_ERR(ts3a227e->regmap); + + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "TS3A227E", ts3a227e); + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret); + return ret; + } + + ret = devm_snd_soc_register_component(&i2c->dev, &ts3a227e_soc_driver, + NULL, 0); + if (ret) + return ret; + + /* Enable interrupts except for ADC complete. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE, + INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, + ADC_COMPLETE_INT_DISABLE); + + return 0; +} + +static const struct i2c_device_id ts3a227e_i2c_ids[] = { + { "ts3a227e", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); + +static const struct of_device_id ts3a227e_of_match[] = { + { .compatible = "ti,ts3a227e", }, + { } +}; +MODULE_DEVICE_TABLE(of, ts3a227e_of_match); + +static struct i2c_driver ts3a227e_driver = { + .driver = { + .name = "ts3a227e", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ts3a227e_of_match), + }, + .probe = ts3a227e_i2c_probe, + .id_table = ts3a227e_i2c_ids, +}; +module_i2c_driver(ts3a227e_driver); + +MODULE_DESCRIPTION("ASoC ts3a227e driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h new file mode 100644 index 0000000..e2acf9c --- /dev/null +++ b/sound/soc/codecs/ts3a227e.h @@ -0,0 +1,17 @@ +/* + * TS3A227E Autonous Audio Accessory Detection and Configureation Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TS3A227E_H +#define _TS3A227E_H + +int ts3a227e_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + +#endif -- 2.1.2.330.g565301e

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Modify the default value and add it to patch table
by Oder Chiou 14 Nov '14

14 Nov '14

Modify the default value and add it to patch table Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d317c68..9ae2e84 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,7 +55,8 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, {RT5677_PR_BASE + 0x13, 0x0312}, {RT5677_PR_BASE + 0x1e, 0x0000}, @@ -173,7 +174,7 @@ static const struct reg_default rt5677_reg[] = { {RT5677_ASRC_9 , 0x0000}, {RT5677_ASRC_10 , 0x0000}, {RT5677_ASRC_11 , 0x0000}, - {RT5677_ASRC_12 , 0x0008}, + {RT5677_ASRC_12 , 0x0018}, {RT5677_ASRC_13 , 0x0000}, {RT5677_ASRC_14 , 0x0000}, {RT5677_ASRC_15 , 0x0000}, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5645: two jacks for hp and mic
by Bard Liao 14 Nov '14

14 Nov '14

Some OS need headphone and microphone to be separated. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 17 ++++++++--------- sound/soc/codecs/rt5645.h | 5 +++-- 2 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 82d705a..31b5b1a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2203,8 +2203,7 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int rt5645_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack) +static int rt5645_jack_detect(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); int gpio_state, jack_type = 0; @@ -2247,19 +2246,19 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, snd_soc_dapm_sync(&codec->dapm); } - snd_soc_jack_report(rt5645->jack, jack_type, SND_JACK_HEADSET); - + snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, jack_type, SND_JACK_MICROPHONE); return 0; } int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack) + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - rt5645->jack = jack; - - rt5645_jack_detect(codec, rt5645->jack); + rt5645->hp_jack = hp_jack; + rt5645->mic_jack = mic_jack; + rt5645_jack_detect(codec); return 0; } @@ -2270,7 +2269,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) struct rt5645_priv *rt5645 = container_of(work, struct rt5645_priv, jack_detect_work.work); - rt5645_jack_detect(rt5645->codec, rt5645->jack); + rt5645_jack_detect(rt5645->codec); } static irqreturn_t rt5645_irq(int irq, void *data) diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 196daf0..c72220a 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2173,7 +2173,8 @@ struct rt5645_priv { struct rt5645_platform_data pdata; struct regmap *regmap; struct i2c_client *i2c; - struct snd_soc_jack *jack; + struct snd_soc_jack *hp_jack; + struct snd_soc_jack *mic_jack; struct delayed_work jack_detect_work; int sysclk; @@ -2188,6 +2189,6 @@ struct rt5645_priv { }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack); + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); #endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix SMALL_POP bit definition
by Fabio Estevam 14 Nov '14

14 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> On a mx28evk with a sgtl5000 codec we notice a loud 'click' sound to happen 5 seconds after the end of a playback. The SMALL_POP bit should fix this, but its definition is incorrect: according to the sgtl5000 manual it is bit 0 of CHIP_REF_CTRL register, not bit 1. Fix the definition accordingly and enable the bit as intended per the code comment. After applying this change, no loud 'click' sound is heard after playback Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 3 +-- sound/soc/codecs/sgtl5000.h | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..dab9b15 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1299,8 +1299,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) /* enable small pop, introduce 400ms delay in turning off */ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - SGTL5000_SMALL_POP, - SGTL5000_SMALL_POP); + SGTL5000_SMALL_POP, 1); /* disable short cut detector */ snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 2f8c889..bd7a344 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -275,7 +275,7 @@ #define SGTL5000_BIAS_CTRL_MASK 0x000e #define SGTL5000_BIAS_CTRL_SHIFT 1 #define SGTL5000_BIAS_CTRL_WIDTH 3 -#define SGTL5000_SMALL_POP 0x0001 +#define SGTL5000_SMALL_POP 0 /* * SGTL5000_CHIP_MIC_CTRL -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5670 : Add ACPI match ID for Intel CHT/BSW platforms
by mengdong.lin＠intel.com 14 Nov '14

14 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds the ACPI match ID for rt5670/5672 codec. So on Intel CherryTrail/Braswell platforms, the codec can be enumerated from ACPI and depends on ACPI to get platform-specific info and power saving. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index c4c42d5..fd32501 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -16,6 +16,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/acpi.h> #include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -2503,6 +2504,14 @@ static const struct i2c_device_id rt5670_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5670_acpi_match[] = { + { "10EC5670", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); +#endif + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2691,6 +2700,7 @@ static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5670_acpi_match), }, .probe = rt5670_i2c_probe, .remove = rt5670_i2c_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: ice1712: remove unneeded return statement
by Sudip Mukherjee 14 Nov '14

14 Nov '14

the functions: snd_ice1712_akm4xxx_build_controls snd_ice1712_build_pro_mixer snd_ctl_add snd_ak4114_build prodigy192_ak4114_init snd_ak4113_build are all returning either 0 or a negetive error value. so we can easily remove the check for a negative value and return the value instead. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- i have not given Jaroslav Kysela <perex(a)perex.cz> in the To list, as in my last mail, all mails bounced with the error: "cant create user output file" sound/pci/ice1712/hoontech.c | 5 +---- sound/pci/ice1712/ice1712.c | 13 +++---------- sound/pci/ice1712/ice1724.c | 5 +---- sound/pci/ice1712/juli.c | 4 +--- sound/pci/ice1712/prodigy192.c | 4 +--- sound/pci/ice1712/quartet.c | 4 +--- 6 files changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index 59e37c5..2c53154 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -310,10 +310,7 @@ static int snd_ice1712_value_init(struct snd_ice1712 *ice) /* ak4524 controls */ err = snd_ice1712_akm4xxx_build_controls(ice); - if (err < 0) - return err; - - return 0; + return err; } static int snd_ice1712_ez8_init(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 48a0c33..b97a0f9 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1296,9 +1296,7 @@ static int snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd } err = snd_ice1712_build_pro_mixer(ice); - if (err < 0) - return err; - return 0; + return err; } /* @@ -1546,9 +1544,7 @@ static int snd_ice1712_ac97_mixer(struct snd_ice1712 *ice) "cannot initialize ac97 for consumer, skipped\n"); else { err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice)); - if (err < 0) - return err; - return 0; + return err; } } @@ -2498,10 +2494,7 @@ static int snd_ice1712_build_controls(struct snd_ice1712 *ice) if (err < 0) return err; err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return err; } static int snd_ice1712_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index f633e3b..fa88485 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2498,10 +2498,7 @@ static int snd_vt1724_build_controls(struct snd_ice1712 *ice) } err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return err; } static int snd_vt1724_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 7a6c078..d6f2fb0 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -477,9 +477,7 @@ static int juli_add_controls(struct snd_ice1712 *ice) /* only capture SPDIF over AK4114 */ err = snd_ak4114_build(spec->ak4114, NULL, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; + return err; } /* diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 1eb151aa..3919aed 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -758,10 +758,8 @@ static int prodigy192_init(struct snd_ice1712 *ice) "AK4114 initialized with status %d\n", err); } else dev_dbg(ice->card->dev, "AK4114 not found\n"); - if (err < 0) - return err; - return 0; + return err; } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d4caf9d..3903e05 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -835,9 +835,7 @@ static int qtet_add_controls(struct snd_ice1712 *ice) /* only capture SPDIF over AK4113 */ err = snd_ak4113_build(spec->ak4113, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; + return err; } static inline int qtet_is_spdif_master(struct snd_ice1712 *ice) -- 1.8.1.2

2 4

[alsa-devel] [PATCH] 6fire: Convert byte_rev_table uses to bitrev8
by Wang, Yalin 14 Nov '14

14 Nov '14

Use the inline function instead of directly indexing the array. This allows some architectures with hardware instructions for bit reversals to eliminate the array. Signed-off-by: Joe Perches <joe(a)perches.com> Signed-off-by: Yalin Wang <yalin.wang(a)sonymobile.com> --- sound/usb/6fire/firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 3b02e54..2e39c3f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -316,7 +316,7 @@ static int usb6fire_fw_fpga_upload( while (c != end) { for (i = 0; c != end && i < FPGA_BUFSIZE; i++, c++) - buffer[i] = byte_rev_table[(u8) *c]; + buffer[i] = bitrev8((u8) *c); ret = usb6fire_fw_fpga_write(device, buffer, i); if (ret < 0) { -- 2.1.1

2 1

[alsa-devel] Low sample rates for SGTL5000 on i.MX28
by Craig McQueen 14 Nov '14

14 Nov '14

I'm testing the 3.18 kernel for i.MX28 EVK, which has an SGTL5000 CODEC. I notice that audio cannot be output at 8 kHz or 11.025 kHz, although the SGTL5000 data sheet table 8 indicates this should be possible. If I try to play or record at 8 kHz or 11.025 kHz, I get an error message: # aplay test_8k.wav [ 6168.910201] mxs_evk_startup: Skipped programming wolfson codec Playing WAVE 'test_8k.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono [ 6168.941211] sgtl5000 0-000a: Invalid mclk frequency: 4.096MHz [ 6168.947047] mxs-sgtl5000 sound.11: ASoC: machine hw_params failed: -22 aplay: set_params:1145: Unable to install hw params: ... For an 8 kHz sample rate, I get Invalid mclk frequency: 4.096MHz For 11.025 kHz sample rate, I get Invalid mclk frequency: 5.644MHz Looking in the code (/sound/soc/mxs/mxs-sgtl5000c), I see: /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ if (mclk < 8000000 || mclk > 27000000) { dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n", mclk / 1000000, mclk / 1000 % 1000); return -EINVAL; } But a sysclk constraint of 8 to 27 MHz only applies if the supplied clock is asynchronous to the frame clock, and the internal clock is being generated by PLL (which it's not in this case). So I tried removing this constraint. But I found that from codecs/sgtl5000.c, I would get the error "PLL not supported in slave mode". I tried changing the switch (sgtl5000->sysclk / sys_fs) to switch (sgtl5000->sysclk / frame_rate), on the basis of the SGTL5000 data sheet table 8. But the resulting audio sounds funny (frequency shifted). Looking in the SGTL5000 data sheet table 18, it seems unclear whether the MCLK_FREQ field is a multiple of the sample rate (Fs) or the internal system sample rate SYS_FS. It seems that there is some inconsistency in the interpretation of this in codecs/sgtl5000.c compared to mxs/mxs-sgtl5000.c. What is the correct interpretation of the SGTL5000 data sheet MCLK_FREQ field, and can the mxs-sgtl5000 driver be fixed to support 8 kHz and 11.025 kHz sample rates? -- Craig McQueen

2 6

[alsa-devel] Stale data at start of audio recording on Freescale i.MX28 EVK with SGTL5000
by Craig McQueen 13 Nov '14

13 Nov '14

I was testing the 3.14.19 kernel on Freescale i.MX28 EVK. Since then, I've retested and confirmed this also on 3.18-rc4 kernel. I am doing audio recordings through the SGTL5000 driver as such: arecord -D hw:0,1 -f s16 -c 2 -r 16000 -d 5 test.wav I'm finding that the first 400-500 ms of the saved data is actually stale data from the end of the previous recording. To see this, do the following: 1. Provide some test audio input in the LINE INPUT jack. E.g. a 100 Hz sine wave, or some music. 2. Record audio with e.g. arecord -D hw:0,1 -f s16 -c 2 -r 16000 -d 5 test1.wav 3. Change the test audio input, e.g. a 150 Hz sine wave, or some different music. 4. Record a new audio file with e.g. arecord -D hw:0,1 -f s16 -c 2 -r 16000 -d 5 test2.wav Observe that the first 400-500 ms of test2.wav contains stale data from the end of the previous recording. -- Craig McQueen

1 0

[alsa-devel] [PATCH v2] ASoC: add TI ts3a227e headset chip driver
by Dylan Reid 13 Nov '14

13 Nov '14

The TS3A227E is an autonomous audio accessory detection and configuration switch that detects 3-pole or 4-pole audio accessories and configures internal switches to route the signals accordingly. This chip also has built-in support for the new button standard described in the Android "Wired audio headset specification" v1.0. These buttons will be reported on the jack as buttons 0-3 mapped to KEY_MEDIA, KEY_VOLUMEUP, KEY_VOLUMEDOWN, and KEY_VOICE_COMMAND. This will be added as an aux_dev and have the jack passed in from the machine driver. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Use button to key mapping instead of separate input device --- .../devicetree/bindings/sound/ts3a227e.txt | 26 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ts3a227e.c | 321 +++++++++++++++++++++ sound/soc/codecs/ts3a227e.h | 17 ++ 5 files changed, 371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt new file mode 100644 index 0000000..e8bf23e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -0,0 +1,26 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupt-parent: The parent interrupt controller + - interrupts: Interrupt number for /INT pin from the 227e + + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..243ec86 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C + select SND_SOC_TS3A227E if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X @@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate +config SND_SOC_TS3A227E + tristate "TI Headset/Mic detect and keypress chip" + depends on I2C + config SND_SOC_TWL4030 select MFD_TWL4030_AUDIO tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..a1eb7ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o @@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c new file mode 100644 index 0000000..4ca6ddc --- /dev/null +++ b/sound/soc/codecs/ts3a227e.c @@ -0,0 +1,321 @@ +/* + * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +struct ts3a227e { + struct regmap *regmap; + struct snd_soc_jack *jack; + bool plugged; + bool mic_present; + unsigned int buttons_held; +}; + +/* Button values to be reported on the jack */ +static const int ts3a227e_buttons[] = { + SND_JACK_BTN_0, + SND_JACK_BTN_1, + SND_JACK_BTN_2, + SND_JACK_BTN_3, +}; + +#define TS3A227E_NUM_BUTTONS 4 +#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_MICROPHONE | \ + SND_JACK_BTN_0 | \ + SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | \ + SND_JACK_BTN_3) + +/* TS3A227E registers */ +#define TS3A227E_REG_DEVICE_ID 0x00 +#define TS3A227E_REG_INTERRUPT 0x01 +#define TS3A227E_REG_KP_INTERRUPT 0x02 +#define TS3A227E_REG_INTERRUPT_DISABLE 0x03 +#define TS3A227E_REG_SETTING_1 0x04 +#define TS3A227E_REG_SETTING_2 0x05 +#define TS3A227E_REG_SETTING_3 0x06 +#define TS3A227E_REG_SWITCH_CONTROL_1 0x07 +#define TS3A227E_REG_SWITCH_CONTROL_2 0x08 +#define TS3A227E_REG_SWITCH_STATUS_1 0x09 +#define TS3A227E_REG_SWITCH_STATUS_2 0x0a +#define TS3A227E_REG_ACCESSORY_STATUS 0x0b +#define TS3A227E_REG_ADC_OUTPUT 0x0c +#define TS3A227E_REG_KP_THRESHOLD_1 0x0d +#define TS3A227E_REG_KP_THRESHOLD_2 0x0e +#define TS3A227E_REG_KP_THRESHOLD_3 0x0f + +/* TS3A227E_REG_INTERRUPT 0x01 */ +#define INS_REM_EVENT 0x01 +#define DETECTION_COMPLETE_EVENT 0x02 + +/* TS3A227E_REG_KP_INTERRUPT 0x02 */ +#define PRESS_MASK(idx) (0x01 << (2 * (idx))) +#define RELEASE_MASK(idx) (0x02 << (2 * (idx))) + +/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */ +#define INS_REM_INT_DISABLE 0x01 +#define DETECTION_COMPLETE_INT_DISABLE 0x02 +#define ADC_COMPLETE_INT_DISABLE 0x04 +#define INTB_DISABLE 0x08 + +/* TS3A227E_REG_SETTING_2 0x05 */ +#define KP_ENABLE 0x04 + +/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ +#define TYPE_3_POLE 0x01 +#define TYPE_4_POLE_OMTP 0x02 +#define TYPE_4_POLE_STANDARD 0x04 +#define JACK_INSERTED 0x08 +#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD) + +static const struct reg_default ts3a227e_reg_defaults[] = { + { TS3A227E_REG_DEVICE_ID, 0x10 }, + { TS3A227E_REG_INTERRUPT, 0x00 }, + { TS3A227E_REG_KP_INTERRUPT, 0x00 }, + { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 }, + { TS3A227E_REG_SETTING_1, 0x23 }, + { TS3A227E_REG_SETTING_2, 0x00 }, + { TS3A227E_REG_SETTING_3, 0x0e }, + { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 }, + { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 }, + { TS3A227E_REG_SWITCH_STATUS_1, 0x0c }, + { TS3A227E_REG_SWITCH_STATUS_2, 0x00 }, + { TS3A227E_REG_ACCESSORY_STATUS, 0x00 }, + { TS3A227E_REG_ADC_OUTPUT, 0x00 }, + { TS3A227E_REG_KP_THRESHOLD_1, 0x20 }, + { TS3A227E_REG_KP_THRESHOLD_2, 0x40 }, + { TS3A227E_REG_KP_THRESHOLD_3, 0x68 }, +}; + +static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2: + case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: + case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: + return true; + default: + return false; + } +} + +static void ts3a227e_jack_report(struct ts3a227e *ts3a227e) +{ + unsigned int i; + int report = 0; + + if (!ts3a227e->jack) + return; + + if (ts3a227e->plugged) + report = SND_JACK_HEADPHONE; + if (ts3a227e->mic_present) + report |= SND_JACK_MICROPHONE; + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (ts3a227e->buttons_held & (1 << i)) + report |= ts3a227e_buttons[i]; + } + snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK); +} + +static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg) +{ + bool plugged, mic_present; + + plugged = !!(acc_reg & JACK_INSERTED); + mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK); + + ts3a227e->plugged = plugged; + + if (mic_present != ts3a227e->mic_present) { + ts3a227e->mic_present = mic_present; + ts3a227e->buttons_held = 0; + if (mic_present) { + /* Enable key press detection. */ + regmap_update_bits(ts3a227e->regmap, + TS3A227E_REG_SETTING_2, + KP_ENABLE, KP_ENABLE); + } + } +} + +static irqreturn_t ts3a227e_interrupt(int irq, void *data) +{ + struct ts3a227e *ts3a227e = (struct ts3a227e *)data; + struct regmap *regmap = ts3a227e->regmap; + unsigned int int_reg, kp_int_reg, acc_reg, i; + + /* Check for plug/unplug. */ + regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { + regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + } + + /* Report any key events. */ + regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (kp_int_reg & PRESS_MASK(i)) + ts3a227e->buttons_held |= (1 << i); + if (kp_int_reg & RELEASE_MASK(i)) + ts3a227e->buttons_held &= ~(1 << i); + } + + ts3a227e_jack_report(ts3a227e); + + return IRQ_HANDLED; +} + +/** + * ts3a227e_enable_jack_detect - Specify a jack for event reporting + * + * @codec: codec to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events 0-3 will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int ts3a227e_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct ts3a227e *ts3a227e = snd_soc_codec_get_drvdata(codec); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ts3a227e->jack = jack; + ts3a227e_jack_report(ts3a227e); + + return 0; +} +EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); + +static struct snd_soc_codec_driver ts3a227e_codec_driver; + +static const struct regmap_config ts3a227e_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = TS3A227E_REG_KP_THRESHOLD_3, + .readable_reg = ts3a227e_readable_reg, + .writeable_reg = ts3a227e_writeable_reg, + .volatile_reg = ts3a227e_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ts3a227e_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), +}; + +static int ts3a227e_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ts3a227e *ts3a227e; + struct device *dev = &i2c->dev; + int ret; + + ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); + if (ts3a227e == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ts3a227e); + + ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); + if (IS_ERR(ts3a227e->regmap)) + return PTR_ERR(ts3a227e->regmap); + + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "TS3A227E", ts3a227e); + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &ts3a227e_codec_driver, + NULL, 0); + if (ret) + return ret; + + /* Enable interrupts except for ADC complete. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE, + INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, + ADC_COMPLETE_INT_DISABLE); + + return 0; +} + +static int ts3a227e_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ts3a227e_i2c_ids[] = { + { "ts3a227e", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); + +static const struct of_device_id ts3a227e_of_match[] = { + { .compatible = "ti,ts3a227e", }, + { } +}; +MODULE_DEVICE_TABLE(of, ts3a227e_of_match); + +static struct i2c_driver ts3a227e_driver = { + .driver = { + .name = "ts3a227e", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ts3a227e_of_match), + }, + .probe = ts3a227e_i2c_probe, + .remove = ts3a227e_i2c_remove, + .id_table = ts3a227e_i2c_ids, +}; +module_i2c_driver(ts3a227e_driver); + +MODULE_DESCRIPTION("ASoC ts3a227e driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h new file mode 100644 index 0000000..8a58886 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.h @@ -0,0 +1,17 @@ +/* + * TS3A227E Autonous Audio Accessory Detection and Configureation Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TS3A227E_H +#define _TS3A227E_H + +int ts3a227e_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- 2.1.2.330.g565301e

2 2

[alsa-devel] Problems with usb soundcard in kernels > 3.3
by Bogdan Veringioiu 13 Nov '14

13 Nov '14

Hello all, After spending a couple of days searching the web for a solution, I write here to ask for help. I am using an usb soundcard (Phoenix Audio Technologies Phnx MT201solo) on Debian Wheezy, and started to experience problems when using kernels > 3.3. The issue I am experiencing is a ~4 seconds delay in playing sound. I am reproducing the problem using aplay. The sound play is being delayed by more or less 4 seconds from the moment the file is sent to the card. I have tested with kernels 3.2 and 3.3, which work properly. I have noticed this delay/wait when using kernels 3.15, 3.16. Below are some details: WORKING PROPERLY # uname -rv 3.2.0-4-686-pae #1 SMP Debian 3.2.63-2 # dpkg -l | grep alsa ii alsa-base 1.0.25+3~deb7u1 all ALSA driver configuration files ii alsa-utils 1.0.25-4 i386 Utilities for configuring and using ALSA # dmesg .... usb 3-1: New USB device found, idVendor=1de7, idProduct=0011 usb 3-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 3-1: Product: Phnx MT201solo usb 3-1: Manufacturer: Phoenix Audio Technologies .... # time aplay /usr/share/sounds/alsa/Rear_Left.wav Playing WAVE '/usr/share/sounds/alsa/Rear_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono real 0m2.498s <--- this is OK user 0m0.032s sys 0m0.024s DELAY INTRODUCED # uname -rv 3.16-0.bpo.2-686-pae #1 SMP Debian 3.16.3-2~bpo70+1 (2014-09-21) # dpkg -l | grep alsa ii alsa-base 1.0.25+3~deb7u1 all ALSA driver configuration files ii alsa-utils 1.0.25-4 i386 Utilities for configuring and using ALSA # dmesg .... usb 2-1: New USB device found, idVendor=1de7, idProduct=0011 usb 2-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 2-1: Product: Phnx MT201solo usb 2-1: Manufacturer: Phoenix Audio Technologies .... # time aplay /usr/share/sounds/alsa/Rear_Left.wav Playing WAVE '/usr/share/sounds/alsa/Rear_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono real 0m6.387s <----4 more seconds delay, NOK user 0m0.016s sys 0m0.024s I have done a strace, see below where the 4 seconds are lost, maybe it helps: open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_CLOEXEC) = 4 fcntl64(4, F_SETFD, FD_CLOEXEC) = 0 ioctl(4, AGPIOC_ACQUIRE or APM_IOC_STANDBY, 0xbf823f4c) = 0 close(3) = 0 ioctl(4, AGPIOC_ACQUIRE or APM_IOC_STANDBY, 0xbf823dac) = 0 fcntl64(4, F_GETFL) = 0x802 (flags O_RDWR|O_NONBLOCK) ioctl(4, AGPIOC_INFO, 0xbf823d14) = 0 clock_gettime(CLOCK_MONOTONIC, {5839, 700325532}) = 0 ioctl(4, AGPIOC_SETUP, 0xbf823d1c) = 0 mmap2(NULL, 4096, PROT_READ, MAP_SHARED, 4, 0x80000) = 0xb72db000 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0x81000) = 0xb72da000 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4111, 0xbf823ee0) = 0 ioctl(4, 0xc0684113, 0xbf823d94) = 0 ioctl(4, 0x80104132, 0xbf823cd0) = 0 ioctl(4, 0x80104132, 0xbf823cd0) = 0 mmap2(NULL, 65536, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0) = 0xb7293000 ----> ioctl(4, 0x4140, 0xb7718ff4) = 0 <---- !!! Here is where the 4 seconds are lost !!! ioctl(4, 0xc0684113, 0xbf823e60) = 0 ioctl(4, 0x4142, 0xb7709c16) = 0 ioctl(4, AGPIOC_INFO, 0xbf8241a0) = 0 shmget(0x56a4d6, 131072, IPC_CREAT|0660) = 163841 shmctl(163841, IPC_64|IPC_STAT, 0xbf82415c) = 0 shmctl(163841, IPC_64|IPC_SET, 0xbf82415c) = 0 .... I think this is a problem in the snd-usb-audio module introduced in kernels > 3.3. Can anyone help me with this issue? Thanks, Bogdan

2 5

[alsa-devel] [PATCH v2] Scarlett mixer interface inclusion
by Chris J Arges 13 Nov '14

13 Nov '14

This is a followup to David Henningsson's PATCH [1] to include the work of Tobias Hoffman and Robin Gareus in getting the Scarlett mixer interface working. I've read through the comments and tried to include fixes where appropriate. Overall, I'd like to get a good base commit and start working on other commits that address any major structual changes such as making functions more generic or any other cleanup. I've tested this on my Scarlett 18i8 mixer and it does work, however there are still a few issues which need to be fixed: 1) When loading a device one gets the following messages: snd-usb-audio 1-1.1:1.0: control 2:0:0:Master Playback Switch:0 is already present snd-usb-audio: probe of 1-1.1:1.3 failed with error -5 2) When unloading a device there are numerous sysfs_remove_group issues: usb 1-1.1: USB disconnect, device number 6 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 54 at /build/buildd/linux-3.16.0/fs/sysfs/group.c:219 sysfs_remove_group+0x99/0xa0() sysfs group ffffffff82cbd6e0 not found for kobject 'midi1' Modules linked in: snd_usb_audio(OE) snd_usbmidi_lib(OE) ctr ccm xt_conntrack ipt_REJECT xt_CHECKSUM iptable_mangle ipt_MASQUERADE iptable_nat nf_conntrack_ipv4 nf_defrag_ipv4 nf_nat_ipv4 nf_nat nf_conntrack xt_tcpudp bridge stp llc iptable_filter ip_tables x_tables arc4 iwldvm snd_hda_codec_hdmi mac80211 snd_hda_codec_conexant snd_hda_codec_generic iwlwifi snd_hda_intel uvcvideo snd_hda_controller videobuf2_vmalloc snd_hda_codec videobuf2_memops videobuf2_core snd_hwdep v4l2_common snd_pcm intel_rapl x86_pkg_temp_thermal videodev intel_powerclamp media coretemp kvm_intel bnep kvm btusb rfcomm cfg80211 joydev bluetooth serio_raw 6lowpan_iphc thinkpad_acpi nvram snd_seq_midi lpc_ich snd_seq_midi_event snd_rawmidi shpchp snd_seq mei_me snd_seq_device snd_timer mei snd soundcore parport_pc ppdev mac_hid lp parport binfmt_misc nls_iso8859_1 dm_crypt crct10dif_pclmul crc32_pclmul ghash_clmulni_intel aesni_intel aes_x86_64 lrw gf128mul glue_helper ablk_helper cryptd i915 psmouse ahci libahci firewire_ohci sdhci_pci firewire_core sdhci crc_itu_t i2c_algo_bit e1000e drm_kms_helper drm ptp pps_core wmi video CPU: 0 PID: 54 Comm: khubd Tainted: G OE 3.16.0-23-generic #30-Ubuntu Hardware name: LENOVO 4177CTO/4177CTO, BIOS 83ET76WW (1.46 ) 07/05/2013 0000000000000009 ffff880407b4f920 ffffffff8277fcbc ffff880407b4f968 ffff880407b4f958 ffffffff8206fd8d 0000000000000000 ffffffff82cbd6e0 ffff880405c7b410 0000000000000001 ffff8803eb059cf8 ffff880407b4f9b8 Call Trace: [<ffffffff8277fcbc>] dump_stack+0x45/0x56 [<ffffffff8206fd8d>] warn_slowpath_common+0x7d/0xa0 [<ffffffff8206fdfc>] warn_slowpath_fmt+0x4c/0x50 [<ffffffff82256538>] ? kernfs_find_and_get_ns+0x48/0x60 [<ffffffff8225a259>] sysfs_remove_group+0x99/0xa0 [<ffffffff824d6927>] dpm_sysfs_remove+0x57/0x60 [<ffffffff824cb625>] device_del+0x45/0x1d0 [<ffffffff824cb7ce>] device_unregister+0x1e/0x70 [<ffffffff824cb89c>] device_destroy+0x3c/0x50 [<ffffffffc045f34b>] sound_remove_unit+0x5b/0x90 [soundcore] [<ffffffffc045f3aa>] unregister_sound_special+0x2a/0x30 [soundcore] [<ffffffffc04cc9b0>] snd_unregister_oss_device+0x60/0x100 [snd] [<ffffffffc0550ca9>] snd_rawmidi_dev_disconnect+0x119/0x160 [snd_rawmidi] [<ffffffffc04cc14c>] __snd_device_disconnect.part.1+0x1c/0x60 [snd] [<ffffffffc04cc437>] snd_device_disconnect_all+0x47/0x60 [snd] [<ffffffffc04c59f7>] snd_card_disconnect+0x147/0x1f0 [snd] [<ffffffffc09cf5a8>] usb_audio_disconnect+0x78/0x1a0 [snd_usb_audio] [<ffffffff825867b4>] usb_unbind_interface+0x74/0x2b0 [<ffffffff824cf76f>] __device_release_driver+0x7f/0xf0 [<ffffffff824cf803>] device_release_driver+0x23/0x30 [<ffffffff824cf0f8>] bus_remove_device+0x108/0x180 [<ffffffff824cb709>] device_del+0x129/0x1d0 [<ffffffff82583f21>] usb_disable_device+0x91/0x290 [<ffffffff82578688>] usb_disconnect+0x98/0x2f0 [<ffffffff8257a6cf>] hub_port_connect+0x7f/0xac0 [<ffffffff8257b9a3>] hub_events+0x893/0xda0 [<ffffffff8257beed>] hub_thread+0x3d/0x160 [<ffffffff820b9590>] ? prepare_to_wait_event+0x100/0x100 [<ffffffff8257beb0>] ? hub_events+0xda0/0xda0 [<ffffffff82094aeb>] kthread+0xdb/0x100 [<ffffffff82094a10>] ? kthread_create_on_node+0x1c0/0x1c0 [<ffffffff82787c3c>] ret_from_fork+0x7c/0xb0 [<ffffffff82094a10>] ? kthread_create_on_node+0x1c0/0x1c0 ---[ end trace 7a8cdc4a9c65ddf5 ]--- However this issue occured even without the patch applied. Please follow up with additional comments and review and I'll be happy to incorporate them into the next patch version. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/081986.ht… Chris J Arges (1): Scarlett mixer interface for 6i6, 18i6, 18i8 and 18i20 sound/usb/Makefile | 1 + sound/usb/mixer.c | 27 +- sound/usb/quirks-table.h | 51 -- sound/usb/scarlett/scarlettmixer.c | 1264 ++++++++++++++++++++++++++++++++++++ sound/usb/scarlett/scarlettmixer.h | 6 + 5 files changed, 1293 insertions(+), 56 deletions(-) create mode 100644 sound/usb/scarlett/scarlettmixer.c create mode 100644 sound/usb/scarlett/scarlettmixer.h -- 2.1.0

6 59

[alsa-devel] [PATCH v2] ASoC: rt5645: add register setting for TDM
by Bard Liao 13 Nov '14

13 Nov '14

We need to set extra register to avoid a recording issue in TDM mode. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Change regmap_update_bits(rt5645->regmap to snd_soc_update_bits(codec since rt5645 struct is not in rt5645_set_tdm_slot --- sound/soc/codecs/rt5645.c | 5 ++++- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d6298b3..82d705a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2114,8 +2114,11 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, struct snd_soc_codec *codec = dai->codec; unsigned int val = 0; - if (rx_mask || tx_mask) + if (rx_mask || tx_mask) { val |= (1 << 14); + snd_soc_update_bits(codec, RT5645_BASS_BACK, + RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); + } switch (slots) { case 4: diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 82f681b..196daf0 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1855,6 +1855,7 @@ #define RT5645_M_BB_HPF_R_SFT 6 #define RT5645_G_BB_BST_MASK (0x3f) #define RT5645_G_BB_BST_SFT 0 +#define RT5645_G_BB_BST_25DB 0x14 /* MP3 Plus Control 1 (0xd0) */ #define RT5645_M_MP3_L_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/4 V2] Arndale: Adding Sound Machine Driver for ALC5631 Codec
by Krishna Mohan Dani 13 Nov '14

13 Nov '14

These patches add machine driver to instantiate I2S based realtek's ALC5631 based sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec based card hence support for ALC5631 based machine driver is being added. Most of the code has been pulled from insignal's git for Arndale and modified to suit the maintain syntax. Machine driver has been made DT compatible. Changes done: - Removed ifdef for of_device_id structure as commented by Mark Brown - Supporting both variants of device name:alc5631 and rt5631 as commented by Mark Brown - Added "return -EINVAL" if cpu_of_node or codec_of_node is not found in dtb, as commented by Anil Kumar Krishna Mohan Dani (4): ASoC: ALC5631/RT5631: Add device tree binding documentation ASoC: Samsung: Add arndale_rt5631 machine driver Sound: Kconfig: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631 codec driver Documentation/devicetree/bindings/sound/rt5631.txt | 48 ++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/rt5631.c | 11 ++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 166 ++++++++++++++++++++ 6 files changed, 235 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c -- 1.7.9.5

3 8

[alsa-devel] [PATCH] ALSA: hda - Add mute LED control for Lenovo Ideapad Z560
by Takashi Iwai 13 Nov '14

13 Nov '14

Lenovo Ideapad Z560 has a mute LED that is controlled via EAPD pin 0x1b on CX20585 codec. (EAPD bit on corresponds to mute LED on.) The machine doesn't need other EAPD, so the fixup concentrates on controlling EAPD 0x1b following the vmaster state (but inversely). Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=665315 Reported-by: Szymon Kowalczyk <fazerxlo(a)o2.pl> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 71e4bad06345..e9ebc7bd752c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -43,6 +43,7 @@ struct conexant_spec { unsigned int num_eapds; hda_nid_t eapds[4]; bool dynamic_eapd; + hda_nid_t mute_led_eapd; unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ @@ -163,6 +164,17 @@ static void cx_auto_vmaster_hook(void *private_data, int enabled) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, enabled); } +/* turn on/off EAPD according to Master switch (inversely!) for mute LED */ +static void cx_auto_vmaster_hook_mute_led(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct conexant_spec *spec = codec->spec; + + snd_hda_codec_write(codec, spec->mute_led_eapd, 0, + AC_VERB_SET_EAPD_BTLENABLE, + enabled ? 0x00 : 0x02); +} + static int cx_auto_build_controls(struct hda_codec *codec) { int err; @@ -223,6 +235,7 @@ enum { CXT_FIXUP_TOSHIBA_P105, CXT_FIXUP_HP_530, CXT_FIXUP_CAP_MIX_AMP_5047, + CXT_FIXUP_MUTE_LED_EAPD, }; /* for hda_fixup_thinkpad_acpi() */ @@ -557,6 +570,18 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec, } } +static void cxt_fixup_mute_led_eapd(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_eapd = 0x1b; + spec->dynamic_eapd = 1; + spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook_mute_led; + } +} + /* * Fix max input level on mixer widget to 0dB * (originally it has 0x2b steps with 0dB offset 0x14) @@ -705,6 +730,10 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_cap_mix_amp_5047, }, + [CXT_FIXUP_MUTE_LED_EAPD] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_mute_led_eapd, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -762,6 +791,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), @@ -780,6 +810,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LEMOTE_A1004, .name = "lemote-a1004" }, { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, + { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, {} }; -- 2.1.3

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[alsa-devel] possible snd_hda_intel jack-sensing issue with device 8086:1c20
by Chris Pemberton 13 Nov '14

13 Nov '14

CentOS 7 x86_64 Asus P8Z68-V motherboard Realtek ALC892 Gnome 3 desktop nVidia GTX 770 with nVidia's driver With BIOS set to HD for both rear and front jacks, I experience the problems I reported here: https://bugs.freedesktop.org/show_bug.cgi?id=86158 Basically, sound works, but the selected output keeps auto-magically flipping from Analog-Out to Headphones and back so fast you barely hear a click.. Tried all sorts of snd_hda_intel module options and it didn't change. Finally changed BIOS to set front jacks to AC97 (left rear jacks as HD) and sound works great... but no jack sensing occurs at all... I can live without the jack sensing... but if anyone wants a stab at a fix, I'm game. I've got the HDA Analyzer downloaded and running if that helps. My device has id 8086:1c20 Perhaps similar to this bug that was brought to my attention? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Output of alsa-info.sh is here: http://www.alsa-project.org/db/?f=8cf116b0bc230524bcfa7285b2adb698078d4224 Thanks.

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Re: [alsa-devel] Compressed audio support over HDMI
by Andrej Falout 13 Nov '14

13 Nov '14

> On 11/4/14, 3:40 PM, D K wrote: > > Hi, > > I recently came across the compressed audio framework used with > ALSA-SOC. > > Can the framework be used to send compressed audio to HDMI driver?.Like > > using tinycompress from usespace . > > thanks, > > HDMI already supports IEC modes when you can send AC3, DTS, etc payloads > without decoding. All you need to do is packetize the data in userspace > and send the result over the HDMI PCM device (and set some control bits > to help the receiver lock faster). > > Apologies for jumping in on this subject, I'm wondering if anyone knows if this method is of sending audio over HDMI is aplocable to DSD streams too? The reasoning being that once audio is compressed, and encoded (AC3, DTS, etc), there is not much inherently "PCM" about it. It only becomes PCM again after it is decoded and uncompressed, right? In short, it is simply a bitstream, packetized and with "some control bits set". Which is exactly what I see in HDMI specs in regards to DSD. So, does anyone know of a reason why this same method wont work for sending DSD bitstream over HDMI? Any advice very much appreciated! Kind regards, Andrej Falout

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[alsa-devel] [PATCH] ASoC: rt5645: add register setting for TDM
by Bard Liao 12 Nov '14

12 Nov '14

We need to set extra register to avoid a recording issue in TDM mode. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 5 ++++- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d6298b3..ac7a52a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2114,8 +2114,11 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, struct snd_soc_codec *codec = dai->codec; unsigned int val = 0; - if (rx_mask || tx_mask) + if (rx_mask || tx_mask) { val |= (1 << 14); + regmap_update_bits(rt5645->regmap, RT5645_BASS_BACK, + RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); + } switch (slots) { case 4: diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 82f681b..196daf0 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1855,6 +1855,7 @@ #define RT5645_M_BB_HPF_R_SFT 6 #define RT5645_G_BB_BST_MASK (0x3f) #define RT5645_G_BB_BST_SFT 0 +#define RT5645_G_BB_BST_25DB 0x14 /* MP3 Plus Control 1 (0xd0) */ #define RT5645_M_MP3_L_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/5] ARM: shmobile: Sound binding updates
by Geert Uytterhoeven 12 Nov '14

12 Nov '14

Hi Mark, Liam, Simon, Magnus, Morimoto-san, THis patch series add SoC-specific binding to the shmobile sound bindings, and updates/cleans up the affected dtsi files. Patches 1 and 3 are meant for the ASoC tree. Patches 2, 4, 5 and are meant for the shmobile tree. Thanks! Geert Uytterhoeven (5): ASoC: sh: fsi: Document SoC-specific bindings ARM: shmobile: sh73a0 dtsi: Add SoC-specific FSI2 compatible property ASoC: rsnd: Document SoC-specific bindings ARM: shmobile: r8a7790 dtsi: Drop "renesas,rcar_sound" compatible value ARM: shmobile: r8a7791 dtsi: Drop "renesas,rcar_sound" compatible value Documentation/devicetree/bindings/sound/renesas,fsi.txt | 9 +++++++-- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 10 +++++++--- arch/arm/boot/dts/r8a7790.dtsi | 2 +- arch/arm/boot/dts/r8a7791.dtsi | 2 +- arch/arm/boot/dts/sh73a0.dtsi | 2 +- 5 files changed, 17 insertions(+), 8 deletions(-) -- 1.9.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [PATCH] ASoC: add TI ts3a227e headset chip driver
by Dylan Reid 12 Nov '14

12 Nov '14

The TS3A227E is an autonomous audio accessory detection and configuration switch that detects 3-pole or 4-pole audio accessories and configures internal switches to route the signals accordingly. This chip also has built-in support for the new button standard described in the Android "Wired audio headset specification" v1.0. These buttons will be reported on a separate input device. The CODEC will be added as an aux_dev and have the jack passed in from the machine driver. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- .../devicetree/bindings/sound/ts3a227e.txt | 26 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ts3a227e.c | 330 +++++++++++++++++++++ sound/soc/codecs/ts3a227e.h | 17 ++ 5 files changed, 380 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt new file mode 100644 index 0000000..e8bf23e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -0,0 +1,26 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupt-parent: The parent interrupt controller + - interrupts: Interrupt number for /INT pin from the 227e + + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..243ec86 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C + select SND_SOC_TS3A227E if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X @@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate +config SND_SOC_TS3A227E + tristate "TI Headset/Mic detect and keypress chip" + depends on I2C + config SND_SOC_TWL4030 select MFD_TWL4030_AUDIO tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..a1eb7ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o @@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c new file mode 100644 index 0000000..c1aa2f5 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.c @@ -0,0 +1,330 @@ +/* + * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +struct ts3a227e { + struct regmap *regmap; + struct snd_soc_jack *jack; + struct input_dev *button_dev; + bool plugged; + bool mic_present; +}; + +/* TS3A227E registers */ +#define TS3A227E_REG_DEVICE_ID 0x00 +#define TS3A227E_REG_INTERRUPT 0x01 +#define TS3A227E_REG_KP_INTERRUPT 0x02 +#define TS3A227E_REG_INTERRUPT_DISABLE 0x03 +#define TS3A227E_REG_SETTING_1 0x04 +#define TS3A227E_REG_SETTING_2 0x05 +#define TS3A227E_REG_SETTING_3 0x06 +#define TS3A227E_REG_SWITCH_CONTROL_1 0x07 +#define TS3A227E_REG_SWITCH_CONTROL_2 0x08 +#define TS3A227E_REG_SWITCH_STATUS_1 0x09 +#define TS3A227E_REG_SWITCH_STATUS_2 0x0a +#define TS3A227E_REG_ACCESSORY_STATUS 0x0b +#define TS3A227E_REG_ADC_OUTPUT 0x0c +#define TS3A227E_REG_KP_THRESHOLD_1 0x0d +#define TS3A227E_REG_KP_THRESHOLD_2 0x0e +#define TS3A227E_REG_KP_THRESHOLD_3 0x0f + +/* TS3A227E_REG_INTERRUPT 0x01 */ +#define INS_REM_EVENT 0x01 +#define DETECTION_COMPLETE_EVENT 0x02 + +/* TS3A227E_REG_KP_INTERRUPT 0x02 */ +#define PRESS_MASK(idx) (0x01 << (2 * (idx))) +#define RELEASE_MASK(idx) (0x02 << (2 * (idx))) + +/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */ +#define INS_REM_INT_DISABLE 0x01 +#define DETECTION_COMPLETE_INT_DISABLE 0x02 +#define ADC_COMPLETE_INT_DISABLE 0x04 +#define INTB_DISABLE 0x08 + +/* TS3A227E_REG_SETTING_2 0x05 */ +#define KP_ENABLE 0x04 + +/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ +#define TYPE_3_POLE 0x01 +#define TYPE_4_POLE_OMTP 0x02 +#define TYPE_4_POLE_STANDARD 0x04 +#define JACK_INSERTED 0x08 +#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD) + +/* Button values to be reported on the jack */ +static const int ts3a227e_keycodes[] = { + KEY_MEDIA, + KEY_VOLUMEUP, + KEY_VOLUMEDOWN, + KEY_VOICECOMMAND, +}; + +#define TS3A227E_NUM_KEYS ARRAY_SIZE(ts3a227e_keycodes) +#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | SND_JACK_MICROPHONE) + +static const struct reg_default ts3a227e_reg_defaults[] = { + { TS3A227E_REG_DEVICE_ID, 0x10 }, + { TS3A227E_REG_INTERRUPT, 0x00 }, + { TS3A227E_REG_KP_INTERRUPT, 0x00 }, + { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 }, + { TS3A227E_REG_SETTING_1, 0x23 }, + { TS3A227E_REG_SETTING_2, 0x00 }, + { TS3A227E_REG_SETTING_3, 0x0e }, + { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 }, + { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 }, + { TS3A227E_REG_SWITCH_STATUS_1, 0x0c }, + { TS3A227E_REG_SWITCH_STATUS_2, 0x00 }, + { TS3A227E_REG_ACCESSORY_STATUS, 0x00 }, + { TS3A227E_REG_ADC_OUTPUT, 0x00 }, + { TS3A227E_REG_KP_THRESHOLD_1, 0x20 }, + { TS3A227E_REG_KP_THRESHOLD_2, 0x40 }, + { TS3A227E_REG_KP_THRESHOLD_3, 0x68 }, +}; + +static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2: + case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: + case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: + return true; + default: + return false; + } +} + +static void ts3a227e_jack_report(struct ts3a227e *ts3a227e) +{ + int report = 0; + + if (!ts3a227e->jack) + return; + + if (ts3a227e->plugged) + report = SND_JACK_HEADPHONE; + if (ts3a227e->mic_present) + report |= SND_JACK_MICROPHONE; + snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK); +} + +static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg) +{ + bool plugged, mic_present; + unsigned int i; + + plugged = !!(acc_reg & JACK_INSERTED); + mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK); + + ts3a227e->plugged = plugged; + + if (mic_present != ts3a227e->mic_present) { + ts3a227e->mic_present = mic_present; + for (i = 0; i < TS3A227E_NUM_KEYS; i++) + input_report_key(ts3a227e->button_dev, + ts3a227e_keycodes[i], 0); + /* Enable key press detection. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_2, + KP_ENABLE, mic_present ? KP_ENABLE : 0); + } +} + +static irqreturn_t ts3a227e_interrupt(int irq, void *data) +{ + struct ts3a227e *ts3a227e = (struct ts3a227e *)data; + struct regmap *regmap = ts3a227e->regmap; + unsigned int int_reg, kp_int_reg, acc_reg, i; + + /* Check for plug/unplug. */ + regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { + regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + } + + /* Report any key events. */ + regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + for (i = 0; i < TS3A227E_NUM_KEYS; i++) { + if (kp_int_reg & PRESS_MASK(i)) + input_report_key(ts3a227e->button_dev, + ts3a227e_keycodes[i], 1); + if (kp_int_reg & RELEASE_MASK(i)) + input_report_key(ts3a227e->button_dev, + ts3a227e_keycodes[i], 0); + } + + input_sync(ts3a227e->button_dev); + ts3a227e_jack_report(ts3a227e); + + return IRQ_HANDLED; +} + +/** + * ts3a227e_enable_jack_detect - Specify a jack for event reporting + * + * @codec: codec to register the jack with + * @jack: jack to use to report headphone/mic plug/unplug on + * + * After this function has been called the headset insert/remove will be routed + * to the given jack. Jack can be null to stop reporting. + */ +int ts3a227e_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct ts3a227e *ts3a227e = snd_soc_codec_get_drvdata(codec); + + ts3a227e->jack = jack; + ts3a227e_jack_report(ts3a227e); + + return 0; +} +EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); + +static struct snd_soc_codec_driver ts3a227e_codec_driver; + +static const struct regmap_config ts3a227e_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = TS3A227E_REG_KP_THRESHOLD_3, + .readable_reg = ts3a227e_readable_reg, + .writeable_reg = ts3a227e_writeable_reg, + .volatile_reg = ts3a227e_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ts3a227e_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), +}; + +static int ts3a227e_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ts3a227e *ts3a227e; + struct device *dev = &i2c->dev; + unsigned int i; + int ret; + + ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); + if (!ts3a227e) + return -ENOMEM; + + i2c_set_clientdata(i2c, ts3a227e); + + ts3a227e->button_dev = devm_input_allocate_device(dev); + if (!ts3a227e->button_dev) { + dev_err(dev, "Failed to allocate button_dev\n"); + return -ENOMEM; + } + + ts3a227e->button_dev->evbit[0] = BIT_MASK(EV_KEY); + for (i = 0; i < TS3A227E_NUM_KEYS; i++) + ts3a227e->button_dev->keybit[BIT_WORD(ts3a227e_keycodes[i])] |= + BIT_MASK(ts3a227e_keycodes[i]); + + ts3a227e->button_dev->name = "ts3a227e headset buttons"; + ret = input_register_device(ts3a227e->button_dev); + if (ret) { + dev_err(dev, "Failed to register input dev %d\n", ret); + return ret; + } + + ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); + if (IS_ERR(ts3a227e->regmap)) + return PTR_ERR(ts3a227e->regmap); + + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "TS3A227E", ts3a227e); + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &ts3a227e_codec_driver, + NULL, 0); + if (ret) + return ret; + + /* Enable interrupts except for ADC complete. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE, + INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, + ADC_COMPLETE_INT_DISABLE); + + return 0; +} + +static int ts3a227e_i2c_remove(struct i2c_client *i2c) +{ + struct ts3a227e *ts3a227e = i2c_get_clientdata(i2c); + + input_unregister_device(ts3a227e->button_dev); + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static const struct i2c_device_id ts3a227e_i2c_ids[] = { + { "ts3a227e", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); + +static const struct of_device_id ts3a227e_of_match[] = { + { .compatible = "ti,ts3a227e", }, + { } +}; +MODULE_DEVICE_TABLE(of, ts3a227e_of_match); + +static struct i2c_driver ts3a227e_driver = { + .driver = { + .name = "ts3a227e", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ts3a227e_of_match), + }, + .probe = ts3a227e_i2c_probe, + .remove = ts3a227e_i2c_remove, + .id_table = ts3a227e_i2c_ids, +}; +module_i2c_driver(ts3a227e_driver); + +MODULE_DESCRIPTION("ASoC ts3a227e driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h new file mode 100644 index 0000000..8a58886 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.h @@ -0,0 +1,17 @@ +/* + * TS3A227E Autonous Audio Accessory Detection and Configureation Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TS3A227E_H +#define _TS3A227E_H + +int ts3a227e_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- 2.1.2.330.g565301e

2 2

[alsa-devel] [PATCH] ASoC: arizona: Add DSP_B and LEFT_J mode support
by Charles Keepax 12 Nov '14

12 Nov '14

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 28 +++++++++++++++++++++++++--- 1 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 1e352d8..9550d74 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -61,6 +61,11 @@ #define ARIZONA_FLL_MIN_OUTDIV 2 #define ARIZONA_FLL_MAX_OUTDIV 7 +#define ARIZONA_FMT_DSP_MODE_A 0 +#define ARIZONA_FMT_DSP_MODE_B 1 +#define ARIZONA_FMT_I2S_MODE 2 +#define ARIZONA_FMT_LEFT_JUSTIFIED_MODE 3 + #define arizona_fll_err(_fll, fmt, ...) \ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_warn(_fll, fmt, ...) \ @@ -946,10 +951,26 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - mode = 0; + mode = ARIZONA_FMT_DSP_MODE_A; + break; + case SND_SOC_DAIFMT_DSP_B: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "DSP_B not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_DSP_MODE_B; break; case SND_SOC_DAIFMT_I2S: - mode = 2; + mode = ARIZONA_FMT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "LEFT_J not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_LEFT_JUSTIFIED_MODE; break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", @@ -1298,7 +1319,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { + val &= ARIZONA_AIF1_FMT_MASK; + if ((channels & 1) && (val == ARIZONA_FMT_I2S_MODE)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); bclk_target /= channels; bclk_target *= channels + 1; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: add missing description for kerneldoc
by Takashi Sakamoto 12 Nov '14

12 Nov '14

A description for second paramter for cmp_connection_check_used() is missing. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/cmp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c index ba8df5a..dcccb01 100644 --- a/sound/firewire/cmp.c +++ b/sound/firewire/cmp.c @@ -154,6 +154,7 @@ EXPORT_SYMBOL(cmp_connection_init); /** * cmp_connection_check_used - check connection is already esablished or not * @c: the connection manager to be checked + * @used: the pointer to store result to check the connection */ int cmp_connection_check_used(struct cmp_connection *c, bool *used) { -- 2.1.0

2 2

[alsa-devel] [PATCHv2] ASoC: cs42l51: re-hook of_match_table pointer
by Thomas Petazzoni 12 Nov '14

12 Nov '14

In commit a1253ef6d3fa ("ASoC: cs42l51: split i2c from codec driver"), the I2C part of the CS42L51 was moved to a separate file, but the definition of the of_device_id array was left in the driver file itself, no longer connected to the platform_driver structure using the .of_match_table pointer. This commit exports the of_device_id array in cs42l51, and uses it as .of_match_able in cs42l51-i2c.c. This solution was suggested by Brian Austin. Fixes: a1253ef6d3fa ("ASoC: cs42l51: split i2c from codec driver") Cc: <stable(a)vger.kernel.org> # v3.16+ Cc: Arnaud Patard (Rtp) <arnaud.patard(a)rtp-net.org> Cc: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> --- sound/soc/codecs/cs42l51-i2c.c | 1 + sound/soc/codecs/cs42l51.c | 4 +++- sound/soc/codecs/cs42l51.h | 1 + 3 files changed, 5 insertions(+), 1 deletion(-) --- Changes since v1: - Instead of moving the of_device_id array definition to the I2C specific file, keep it in the core file, and export it so it can be used by the I2C specific file. Suggested by Brian Austin. diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index cee51ae..c40428f 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -46,6 +46,7 @@ static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51", .owner = THIS_MODULE, + .of_match_table = cs42l51_of_match, }, .probe = cs42l51_i2c_probe, .remove = cs42l51_i2c_remove, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 09488d9..669c38f 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -558,11 +558,13 @@ error: } EXPORT_SYMBOL_GPL(cs42l51_probe); -static const struct of_device_id cs42l51_of_match[] = { +const struct of_device_id cs42l51_of_match[] = { { .compatible = "cirrus,cs42l51", }, { } }; MODULE_DEVICE_TABLE(of, cs42l51_of_match); +EXPORT_SYMBOL_GPL(cs42l51_of_match); + MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 8c55bf3..0ca8054 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -22,6 +22,7 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); +extern const struct of_device_id cs42l51_of_match[]; #define CS42L51_CHIP_ID 0x1B #define CS42L51_CHIP_REV_A 0x00 -- 2.0.0

3 2

[alsa-devel] [PATCH] ASoC: arizona: Add support for 768kHz DMIC operation
by Charles Keepax 12 Nov '14

12 Nov '14

The new IPs supports a new lower frequency 768kHz DMIC operation add support for this into the OSR control. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 730636c..1e352d8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -648,7 +648,7 @@ SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum, EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); static const char * const arizona_in_dmic_osr_text[] = { - "1.536MHz", "3.072MHz", "6.144MHz", + "1.536MHz", "3.072MHz", "6.144MHz", "768kHz", }; const struct soc_enum arizona_in_dmic_osr[] = { -- 1.7.2.5

2 1

[alsa-devel] [PATCH] a52: Ignore start request if we're already running
by David Henningsson 12 Nov '14

12 Nov '14

When trying to start a PCM that's already running, the result is EBADFD. We might have implicitly started the buffer by filling it up, so just ignore this request if we're already running. Reported-by: Chow Loong Jin <hyperair(a)ubuntu.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- a52/pcm_a52.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index b467ec8..59e25cb 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -482,6 +482,12 @@ static int a52_start(snd_pcm_ioplug_t *io) { struct a52_ctx *rec = io->private_data; + /* When trying to start a PCM that's already running, the result is + EBADFD. We might have implicitly started the buffer by filling it + up, so just ignore this request if we're already running. */ + if (snd_pcm_state(rec->slave) == SND_PCM_STATE_RUNNING) + return 0; + return snd_pcm_start(rec->slave); } -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: Add overrun/underrun event handling
by Peter Ujfalusi 12 Nov '14

12 Nov '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> An underrun (playback) event occurs when the serializer transfer data from the XRBUF buffer to the XRSR shift register, but the XRBUF hasn't been filled. Similarly, the overrun (capture) event occurs when data from the XRSR shift register is transferred to the XRBUF but it hasn't been read yet. These events are handled as XRUN events that cause the pcm to stop. The stream has to be explicitly restarted by the userspace which ensures that after stopping/starting McASP the data transfer is aligned with DMA. The other possibility was to internally stop and start McASP without DMA even knowing about it. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Chanhes since v2: - From interrupt handler return IRQ_HANDLED or IRQ_NONE depending on if we have handled event or not. Changes since v1: - only ack handled interrupt (over/underflow at the moment) - Use variable to keep track of requested irq sources for future use - print warning in case of not handled interrupt. Regards, Peter .../bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 124 +++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 11 ++ 3 files changed, 136 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 60ca07996458..46bc9829c71a 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -32,7 +32,7 @@ Optional properties: - rx-num-evt : FIFO levels. - sram-size-playback : size of sram to be allocated during playback - sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP, currently not used by the driver +- interrupts : Interrupt numbers for McASP - interrupt-names : Known interrupt names are "tx" and "rx" - pinctrl-0: Should specify pin control group used for this controller. - pinctrl-names: Should contain only one value - "default", for more details diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6b15a974f585..9de954a24160 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -70,6 +70,7 @@ struct davinci_mcasp { void __iomem *base; u32 fifo_base; struct device *dev; + struct snd_pcm_substream *substreams[2]; /* McASP specific data */ int tdm_slots; @@ -80,6 +81,7 @@ struct davinci_mcasp { u8 bclk_div; u16 bclk_lrclk_ratio; int streams; + u32 irq_request[2]; int sysclk_freq; bool bclk_master; @@ -185,6 +187,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); if (mcasp_is_synchronous(mcasp)) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable receive IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -214,6 +220,10 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); /* Release Frame Sync generator */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable transmit IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); } static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) @@ -228,6 +238,10 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); + /* * In synchronous mode stop the TX clocks if no other stream is * running @@ -249,6 +263,10 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { u32 val = 0; + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); + /* * In synchronous mode keep TX clocks running if the capture stream is * still running. @@ -276,6 +294,76 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) mcasp_stop_rx(mcasp); } +static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG); + if (stat & XUNDRN & irq_mask) { + dev_warn(mcasp->dev, "Transmit buffer underflow\n"); + handled_mask |= XUNDRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled tx event. txstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, handled_mask); + + return IRQ_RETVAL(handled_mask); +} + +static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG); + if (stat & ROVRN & irq_mask) { + dev_warn(mcasp->dev, "Receive buffer overflow\n"); + handled_mask |= ROVRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled rx event. rxstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, handled_mask); + + return IRQ_RETVAL(handled_mask); +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -878,6 +966,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir; + mcasp->substreams[substream->stream] = substream; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; @@ -916,6 +1006,8 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + mcasp->substreams[substream->stream] = NULL; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return; @@ -1266,6 +1358,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; + char *irq_name; + int irq; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -1346,6 +1440,36 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "rx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_rx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "RX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + + irq = platform_get_irq_byname(pdev, "tx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_tx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "TX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + } + dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); if (dat) mcasp->dat_port = true; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9737108f0305..79dc511180bf 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -256,6 +256,7 @@ * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits */ +#define XRERR BIT(8) /* Transmit/Receive error */ #define XRDATA BIT(5) /* Transmit/Receive data ready */ /* @@ -285,6 +286,16 @@ #define TXDATADMADIS BIT(0) /* + * DAVINCI_MCASP_EVTCTLR_REG - Receiver Interrupt Control Register Bits + */ +#define ROVRN BIT(0) + +/* + * DAVINCI_MCASP_EVTCTLX_REG - Transmitter Interrupt Control Register Bits + */ +#define XUNDRN BIT(0) + +/* * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits */ #define FIFO_ENABLE BIT(16) -- 2.1.3

2 1

[alsa-devel] [PATCH v7 00/20] Rework OMAP4+ HDMI audio support
by Jyri Sarha 12 Nov '14

12 Nov '14

The patches are based on: git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git for-next The base, the patches, and couple of additional not-to-be-merged omap2plus_defconfig patches can be found here: https://github.com/jsarha/linux.git omap-hdmi-audio It would make the most sense to get these in trough fbdev tree. So it would be nice to get acked-bys (if the patches are Ok) for ASoC side changes from appropriate maintainers. Changes since v6: - "ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support" - Fix Kconfig help text - Remove #include <sound/simple_card.h> - Use snd-soc-dummy codec instead of hdmi-audio-codec - Add: OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() - Add: OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter - Add: "OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device" - "OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - "OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - Add: "OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback" Jyri Sarha (20): OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi: Remove most of OMAP[45]_DSS_HDMI_AUDIO ifdefs OMAPDSS: hdmi4_core: Remove unused hdmi4_audio_get_dma_port() OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device OMAPDSS: hdmi_wp: Add function for getting audio dma address OMAPDSS: hdmi: Make hdmi structure public OMAPDSS: hdmi: Add pdev pointer for audio_pdev in HDMI DRV data ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support OMAPDSS: Kconfig: Remove HDMI audio booleans from Kconfig OMAPDSS: hdmi: Make hdmi_mode_has_audio() more user friedly OMAPDSS: hdmi.h: Add members to hdmi drvdata for audio implementation OMAPDSS: hdmi4: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio OMAPDSS: hdmi5: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove all references to obsolete HDMI audio callbacks OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ----- .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 7 - drivers/video/fbdev/omap2/dss/hdmi.h | 38 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 269 +++++++------ drivers/video/fbdev/omap2/dss/hdmi4_core.c | 14 - drivers/video/fbdev/omap2/dss/hdmi4_core.h | 4 - drivers/video/fbdev/omap2/dss/hdmi5.c | 265 ++++++------- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 9 +- drivers/video/fbdev/omap2/dss/hdmi5_core.h | 2 - drivers/video/fbdev/omap2/dss/hdmi_common.c | 2 - drivers/video/fbdev/omap2/dss/hdmi_wp.c | 16 +- include/sound/omap-hdmi-audio.h | 43 +++ include/video/omapdss.h | 40 -- sound/soc/omap/Kconfig | 26 +- sound/soc/omap/Makefile | 6 +- sound/soc/omap/omap-hdmi-audio.c | 407 ++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 ----------------- sound/soc/omap/omap-hdmi.h | 38 -- 20 files changed, 781 insertions(+), 1011 deletions(-) create mode 100644 include/sound/omap-hdmi-audio.h create mode 100644 sound/soc/omap/omap-hdmi-audio.c delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h -- 1.7.9.5

2 23

[alsa-devel] [PATCH 4/4] ASoC: rt5670: add codec irq support
by bardliao＠realtek.com 12 Nov '14

12 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch provide the jack function of rt5670. We use a irq pin to notice the jack event. rt5670_irq_detection will determine which status should be reported. We can get gpio information from gpio number or gpiod_dev. That's why we define both on platform data. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5670.h | 2 + sound/soc/codecs/rt5670.c | 176 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5670.h | 2 + 3 files changed, 178 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h index bd31119..bb5cf6d 100644 --- a/include/sound/rt5670.h +++ b/include/sound/rt5670.h @@ -14,6 +14,8 @@ struct rt5670_platform_data { int jd_mode; bool in2_diff; + int codec_gpio; + bool dev_gpio; bool dmic_en; unsigned int dmic1_data_pin; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 89fa507..cb5a91c 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -401,6 +401,144 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg) } } +/** + * rt5670_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int val; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_DET_MODE | RT5670_CBJ_MN_JD, + RT5670_CBJ_MN_JD); + snd_soc_write(codec, RT5670_GPIO_CTRL2, 0x0004); + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + snd_soc_update_bits(codec, RT5670_CJ_CTRL1, + RT5670_CBJ_BST1_EN, RT5670_CBJ_BST1_EN); + snd_soc_write(codec, RT5670_JD_CTRL3, 0x00f0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, RT5670_CBJ_MN_JD); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, 0); + msleep(300); + val = snd_soc_read(codec, RT5670_CJ_CTRL3) & 0x7; + if (val == 0x1 || val == 0x2) { + rt5670->jack_type = SND_JACK_HEADSET; + /* for push button */ + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5670_IL_CMD, 0x40, 0x40); + snd_soc_read(codec, RT5670_IL_CMD); + } else { + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = SND_JACK_HEADPHONE; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + } else { + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x0); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = 0; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + return rt5670->jack_type; +} + +static int rt5670_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5670_IL_CMD); + btn_type = val & 0xff80; + snd_soc_write(codec, RT5670_IL_CMD, val); + if (btn_type != 0) { + msleep(20); + val = snd_soc_read(codec, RT5670_IL_CMD); + snd_soc_write(codec, RT5670_IL_CMD, val); + } + + return btn_type; +} + +static int rt5670_irq_detection(void *data) +{ + struct rt5670_priv *rt5670 = (struct rt5670_priv *)data; + struct snd_soc_jack_gpio *gpio = &rt5670->hp_gpio; + struct snd_soc_jack *jack = &rt5670->jack; + int val, btn_type, report = jack->status; + + if (rt5670->pdata.jd_mode == 1) /* 2 port */ + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0070; + else + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0020; + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5670->jack_type == 0) { + report = rt5670_headset_detect(rt5670->codec, 1); + /* for push button and jack out */ + gpio->debounce_time = 25; + break; + } + btn_type = 0; + if (snd_soc_read(rt5670->codec, RT5670_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5670_button_detect(rt5670->codec); + switch (btn_type) { + case 0x2000: /* up */ + report |= SND_JACK_BTN_1; + break; + case 0x0400: /* center */ + report |= SND_JACK_BTN_0; + break; + case 0x0080: /* down */ + report |= SND_JACK_BTN_2; + break; + default: + dev_err(rt5670->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5670->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5670->codec, RT5670_INT_IRQ_ST, 0x1, 0x0); + rt5670_headset_detect(rt5670->codec, 0); + gpio->debounce_time = 150; /* for jack in */ + break; + default: + break; + } + + return report; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -2413,6 +2551,34 @@ static int rt5670_probe(struct snd_soc_codec *codec) return -ENODEV; } rt5670->codec = codec; + if ((rt5670->pdata.codec_gpio != -1) || (rt5670->pdata.dev_gpio)) { + if (rt5670->pdata.dev_gpio) + rt5670->hp_gpio.gpiod_dev = codec->dev; + else + rt5670->hp_gpio.gpio = rt5670->pdata.codec_gpio; + + rt5670->hp_gpio.name = "headphone detect"; + rt5670->hp_gpio.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2; + rt5670->hp_gpio.debounce_time = 150, + rt5670->hp_gpio.wake = true, + rt5670->hp_gpio.data = (struct rt5670_priv *)rt5670, + rt5670->hp_gpio.jack_status_check = rt5670_irq_detection, + ret = snd_soc_jack_new(codec, rt5670->hp_gpio.name, + rt5670->hp_gpio.report, + &rt5670->jack); + if (ret) { + dev_err(codec->dev, "Jack creation failed\n"); + return ret; + } + + ret = snd_soc_jack_add_gpios(&rt5670->jack, 1, + &rt5670->hp_gpio); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + } return 0; } @@ -2422,6 +2588,7 @@ static int rt5670_remove(struct snd_soc_codec *codec) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5670->regmap, RT5670_RESET, 0); + snd_soc_jack_free_gpios(&rt5670->jack, 1, &rt5670->hp_gpio); return 0; } @@ -2590,12 +2757,17 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5670->regmap, RT5670_IN2, RT5670_IN_DF2, RT5670_IN_DF2); - if (i2c->irq) { + if ((rt5670->pdata.codec_gpio != -1) || (rt5670->pdata.dev_gpio)) { + /* for push button */ + regmap_write(rt5670->regmap, RT5670_IL_CMD, 0x0000); + regmap_write(rt5670->regmap, RT5670_IL_CMD2, 0x0010); + regmap_write(rt5670->regmap, RT5670_IL_CMD3, 0x0014); + /* for irq */ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); - + regmap_update_bits(rt5670->regmap, RT5670_DIG_MISC, 0x8, 0x8); } if (rt5670->pdata.jd_mode) { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index d11b9c2..54933f3 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1987,6 +1987,8 @@ struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; struct regmap *regmap; + struct snd_soc_jack jack; + struct snd_soc_jack_gpio hp_gpio; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

3 2

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Add overrun/underrun event handling
by Peter Ujfalusi 12 Nov '14

12 Nov '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> An underrun (playback) event occurs when the serializer transfer data from the XRBUF buffer to the XRSR shift register, but the XRBUF hasn't been filled. Similarly, the overrun (capture) event occurs when data from the XRSR shift register is transferred to the XRBUF but it hasn't been read yet. These events are handled as XRUN events that cause the pcm to stop. The stream has to be explicitly restarted by the userspace which ensures that after stopping/starting McASP the data transfer is aligned with DMA. The other possibility was to internally stop and start McASP without DMA even knowing about it. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - only ack handled interrupt (over/underflow at the moment) - Use variable to keep track of requested irq sources for future use - print warning in case of not handled interrupt. Regards, Peter .../bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 124 +++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 11 ++ 3 files changed, 136 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 60ca07996458..46bc9829c71a 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -32,7 +32,7 @@ Optional properties: - rx-num-evt : FIFO levels. - sram-size-playback : size of sram to be allocated during playback - sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP, currently not used by the driver +- interrupts : Interrupt numbers for McASP - interrupt-names : Known interrupt names are "tx" and "rx" - pinctrl-0: Should specify pin control group used for this controller. - pinctrl-names: Should contain only one value - "default", for more details diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6b15a974f585..601268350c58 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -70,6 +70,7 @@ struct davinci_mcasp { void __iomem *base; u32 fifo_base; struct device *dev; + struct snd_pcm_substream *substreams[2]; /* McASP specific data */ int tdm_slots; @@ -80,6 +81,7 @@ struct davinci_mcasp { u8 bclk_div; u16 bclk_lrclk_ratio; int streams; + u32 irq_request[2]; int sysclk_freq; bool bclk_master; @@ -185,6 +187,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); if (mcasp_is_synchronous(mcasp)) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable receive IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -214,6 +220,10 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); /* Release Frame Sync generator */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable transmit IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); } static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) @@ -228,6 +238,10 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); + /* * In synchronous mode stop the TX clocks if no other stream is * running @@ -249,6 +263,10 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { u32 val = 0; + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); + /* * In synchronous mode keep TX clocks running if the capture stream is * still running. @@ -276,6 +294,76 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) mcasp_stop_rx(mcasp); } +static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG); + if (stat & XUNDRN & irq_mask) { + dev_warn(mcasp->dev, "Transmit buffer underflow\n"); + handled_mask |= XUNDRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled tx event. txstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, handled_mask); + + return IRQ_HANDLED; +} + +static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG); + if (stat & ROVRN & irq_mask) { + dev_warn(mcasp->dev, "Receive buffer overflow\n"); + handled_mask |= ROVRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled rx event. rxstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, handled_mask); + + return IRQ_HANDLED; +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -878,6 +966,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir; + mcasp->substreams[substream->stream] = substream; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; @@ -916,6 +1006,8 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + mcasp->substreams[substream->stream] = NULL; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return; @@ -1266,6 +1358,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; + char *irq_name; + int irq; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -1346,6 +1440,36 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "rx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_rx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "RX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + + irq = platform_get_irq_byname(pdev, "tx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_tx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "TX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + } + dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); if (dat) mcasp->dat_port = true; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9737108f0305..79dc511180bf 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -256,6 +256,7 @@ * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits */ +#define XRERR BIT(8) /* Transmit/Receive error */ #define XRDATA BIT(5) /* Transmit/Receive data ready */ /* @@ -285,6 +286,16 @@ #define TXDATADMADIS BIT(0) /* + * DAVINCI_MCASP_EVTCTLR_REG - Receiver Interrupt Control Register Bits + */ +#define ROVRN BIT(0) + +/* + * DAVINCI_MCASP_EVTCTLX_REG - Transmitter Interrupt Control Register Bits + */ +#define XUNDRN BIT(0) + +/* * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits */ #define FIFO_ENABLE BIT(16) -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Add TDM channel mux in DAC side of IF1 and IF2
by Oder Chiou 12 Nov '14

12 Nov '14

It is the slot selection in DAC side of IF1 and IF2. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 330 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5677.h | 48 ++++++- 2 files changed, 358 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 4b6f7d5..5d317c68 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1906,6 +1906,126 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if2_adc_tdm_swap_mux = SOC_DAPM_ENUM("IF2 ADC TDM Swap Source", rt5677_if2_adc_tdm_swap_enum); +/* TDM IF1/2 DAC Data Selection */ /* MX-3E[14:12][10:8][6:4][2:0] + MX-3F[14:12][10:8][6:4][2:0] + MX-43[14:12][10:8][6:4][2:0] + MX-44[14:12][10:8][6:4][2:0] */ +static const char * const rt5677_if12_dac_tdm_sel_src[] = { + "Slot0", "Slot1", "Slot2", "Slot3", "Slot4", "Slot5", "Slot6", "Slot7" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac0_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC0_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac0_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC0 TDM Source", rt5677_if1_dac0_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac1_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC1_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac1_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC1 TDM Source", rt5677_if1_dac1_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac2_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC2_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac2_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC2 TDM Source", rt5677_if1_dac2_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac3_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC3_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac3_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC3 TDM Source", rt5677_if1_dac3_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac4_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC4_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac4_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC4 TDM Source", rt5677_if1_dac4_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac5_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC5_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac5_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC5 TDM Source", rt5677_if1_dac5_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac6_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC6_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac6_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC6 TDM Source", rt5677_if1_dac6_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac7_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC7_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac7_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC7 TDM Source", rt5677_if1_dac7_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac0_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC0_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac0_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC0 TDM Source", rt5677_if2_dac0_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac1_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC1_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac1_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC1 TDM Source", rt5677_if2_dac1_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac2_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC2_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac2_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC2 TDM Source", rt5677_if2_dac2_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac3_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC3_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac3_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC3 TDM Source", rt5677_if2_dac3_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac4_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC4_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac4_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC4 TDM Source", rt5677_if2_dac4_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac5_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC5_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac5_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC5 TDM Source", rt5677_if2_dac5_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac6_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC6_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac6_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC6 TDM Source", rt5677_if2_dac6_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac7_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC7_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac7_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC7 TDM Source", rt5677_if2_dac7_tdm_sel_enum); + static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2389,6 +2509,40 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_MUX("SLB ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_slb_adc4_mux), + SND_SOC_DAPM_MUX("IF1 DAC0 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac4_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC5 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac5_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC6 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac6_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC7 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac7_tdm_sel_mux), + + SND_SOC_DAPM_MUX("IF2 DAC0 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac4_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC5 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac5_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC6 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac6_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC7 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac7_tdm_sel_mux), + /* Audio Interface */ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), @@ -3036,14 +3190,86 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF1 DAC6", NULL, "I2S1" }, { "IF1 DAC7", NULL, "I2S1" }, - { "IF1 DAC01", NULL, "IF1 DAC0" }, - { "IF1 DAC01", NULL, "IF1 DAC1" }, - { "IF1 DAC23", NULL, "IF1 DAC2" }, - { "IF1 DAC23", NULL, "IF1 DAC3" }, - { "IF1 DAC45", NULL, "IF1 DAC4" }, - { "IF1 DAC45", NULL, "IF1 DAC5" }, - { "IF1 DAC67", NULL, "IF1 DAC6" }, - { "IF1 DAC67", NULL, "IF1 DAC7" }, + { "IF1 DAC0 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC0 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC0 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC0 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC0 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC0 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC0 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC0 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC1 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC1 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC1 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC1 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC1 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC1 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC1 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC1 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC2 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC2 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC2 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC2 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC2 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC2 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC2 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC2 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC3 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC3 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC3 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC3 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC3 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC3 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC3 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC3 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC4 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC4 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC4 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC4 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC4 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC4 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC4 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC4 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC5 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC5 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC5 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC5 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC5 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC5 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC5 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC5 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC6 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC6 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC6 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC6 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC6 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC6 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC6 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC6 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC7 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC7 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC7 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC7 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC7 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC7 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC7 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC7 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC01", NULL, "IF1 DAC0 Mux" }, + { "IF1 DAC01", NULL, "IF1 DAC1 Mux" }, + { "IF1 DAC23", NULL, "IF1 DAC2 Mux" }, + { "IF1 DAC23", NULL, "IF1 DAC3 Mux" }, + { "IF1 DAC45", NULL, "IF1 DAC4 Mux" }, + { "IF1 DAC45", NULL, "IF1 DAC5 Mux" }, + { "IF1 DAC67", NULL, "IF1 DAC6 Mux" }, + { "IF1 DAC67", NULL, "IF1 DAC7 Mux" }, { "IF2 DAC0", NULL, "AIF2RX" }, { "IF2 DAC1", NULL, "AIF2RX" }, @@ -3062,14 +3288,86 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF2 DAC6", NULL, "I2S2" }, { "IF2 DAC7", NULL, "I2S2" }, - { "IF2 DAC01", NULL, "IF2 DAC0" }, - { "IF2 DAC01", NULL, "IF2 DAC1" }, - { "IF2 DAC23", NULL, "IF2 DAC2" }, - { "IF2 DAC23", NULL, "IF2 DAC3" }, - { "IF2 DAC45", NULL, "IF2 DAC4" }, - { "IF2 DAC45", NULL, "IF2 DAC5" }, - { "IF2 DAC67", NULL, "IF2 DAC6" }, - { "IF2 DAC67", NULL, "IF2 DAC7" }, + { "IF2 DAC0 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC0 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC0 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC0 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC0 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC0 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC0 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC0 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC1 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC1 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC1 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC1 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC1 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC1 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC1 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC1 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC2 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC2 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC2 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC2 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC2 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC2 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC2 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC2 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC3 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC3 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC3 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC3 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC3 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC3 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC3 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC3 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC4 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC4 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC4 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC4 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC4 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC4 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC4 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC4 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC5 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC5 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC5 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC5 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC5 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC5 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC5 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC5 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC6 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC6 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC6 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC6 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC6 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC6 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC6 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC6 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC7 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC7 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC7 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC7 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC7 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC7 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC7 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC7 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC01", NULL, "IF2 DAC0 Mux" }, + { "IF2 DAC01", NULL, "IF2 DAC1 Mux" }, + { "IF2 DAC23", NULL, "IF2 DAC2 Mux" }, + { "IF2 DAC23", NULL, "IF2 DAC3 Mux" }, + { "IF2 DAC45", NULL, "IF2 DAC4 Mux" }, + { "IF2 DAC45", NULL, "IF2 DAC5 Mux" }, + { "IF2 DAC67", NULL, "IF2 DAC6 Mux" }, + { "IF2 DAC67", NULL, "IF2 DAC7 Mux" }, { "IF3 DAC", NULL, "AIF3RX" }, { "IF3 DAC", NULL, "I2S3" }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 9d473b2..2979d5a 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -799,7 +799,7 @@ #define RT5677_PDM2_I2C_EXE (0x1 << 1) #define RT5677_PDM2_I2C_BUSY (0x1 << 0) -/* MX3B TDM1 control 1 (0x3b) */ +/* TDM1 control 1 (0x3b) */ #define RT5677_IF1_ADC_MODE_MASK (0x1 << 12) #define RT5677_IF1_ADC_MODE_SFT 12 #define RT5677_IF1_ADC_MODE_I2S (0x0 << 12) @@ -813,7 +813,7 @@ #define RT5677_IF1_ADC4_SWAP_MASK (0x3 << 0) #define RT5677_IF1_ADC4_SWAP_SFT 0 -/* MX3C TDM1 control 2 (0x3c) */ +/* TDM1 control 2 (0x3c) */ #define RT5677_IF1_ADC4_MASK (0x3 << 10) #define RT5677_IF1_ADC4_SFT 10 #define RT5677_IF1_ADC3_MASK (0x3 << 8) @@ -825,7 +825,27 @@ #define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0) #define RT5677_IF1_ADC_CTRL_SFT 0 -/* MX40 TDM2 control 1 (0x40) */ +/* TDM1 control 4 (0x3e) */ +#define RT5677_IF1_DAC0_MASK (0x7 << 12) +#define RT5677_IF1_DAC0_SFT 12 +#define RT5677_IF1_DAC1_MASK (0x7 << 8) +#define RT5677_IF1_DAC1_SFT 8 +#define RT5677_IF1_DAC2_MASK (0x7 << 4) +#define RT5677_IF1_DAC2_SFT 4 +#define RT5677_IF1_DAC3_MASK (0x7 << 0) +#define RT5677_IF1_DAC3_SFT 0 + +/* TDM1 control 5 (0x3f) */ +#define RT5677_IF1_DAC4_MASK (0x7 << 12) +#define RT5677_IF1_DAC4_SFT 12 +#define RT5677_IF1_DAC5_MASK (0x7 << 8) +#define RT5677_IF1_DAC5_SFT 8 +#define RT5677_IF1_DAC6_MASK (0x7 << 4) +#define RT5677_IF1_DAC6_SFT 4 +#define RT5677_IF1_DAC7_MASK (0x7 << 0) +#define RT5677_IF1_DAC7_SFT 0 + +/* TDM2 control 1 (0x40) */ #define RT5677_IF2_ADC_MODE_MASK (0x1 << 12) #define RT5677_IF2_ADC_MODE_SFT 12 #define RT5677_IF2_ADC_MODE_I2S (0x0 << 12) @@ -839,7 +859,7 @@ #define RT5677_IF2_ADC4_SWAP_MASK (0x3 << 0) #define RT5677_IF2_ADC4_SWAP_SFT 0 -/* MX41 TDM2 control 2 (0x41) */ +/* TDM2 control 2 (0x41) */ #define RT5677_IF2_ADC4_MASK (0x3 << 10) #define RT5677_IF2_ADC4_SFT 10 #define RT5677_IF2_ADC3_MASK (0x3 << 8) @@ -851,6 +871,26 @@ #define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0) #define RT5677_IF2_ADC_CTRL_SFT 0 +/* TDM2 control 4 (0x43) */ +#define RT5677_IF2_DAC0_MASK (0x7 << 12) +#define RT5677_IF2_DAC0_SFT 12 +#define RT5677_IF2_DAC1_MASK (0x7 << 8) +#define RT5677_IF2_DAC1_SFT 8 +#define RT5677_IF2_DAC2_MASK (0x7 << 4) +#define RT5677_IF2_DAC2_SFT 4 +#define RT5677_IF2_DAC3_MASK (0x7 << 0) +#define RT5677_IF2_DAC3_SFT 0 + +/* TDM2 control 5 (0x44) */ +#define RT5677_IF2_DAC4_MASK (0x7 << 12) +#define RT5677_IF2_DAC4_SFT 12 +#define RT5677_IF2_DAC5_MASK (0x7 << 8) +#define RT5677_IF2_DAC5_SFT 8 +#define RT5677_IF2_DAC6_MASK (0x7 << 4) +#define RT5677_IF2_DAC6_SFT 4 +#define RT5677_IF2_DAC7_MASK (0x7 << 0) +#define RT5677_IF2_DAC7_SFT 0 + /* Digital Microphone Control 1 (0x50) */ #define RT5677_DMIC_1_EN_MASK (0x1 << 15) #define RT5677_DMIC_1_EN_SFT 15 -- 1.8.1.1.439.g50a6b54

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Re: [alsa-devel] ALC1150 COEF
by Kailang 12 Nov '14

12 Nov '14

I merge all codec into alc_fill_eapd_coef(). Please see attach. Many Thanks. ^^ > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, November 12, 2014 4:39 PM > To: Kailang > Subject: Re: ALC1150 COEF > > At Wed, 12 Nov 2014 08:27:56 +0000, > Kailang wrote: > > > > Hi Takashi, > > > > I cooked a patch for this issue as attach. > > It is not include ALC26x ALC28x ALC29x. > > If you agree this patch, > > could it merge into one patch? > > Yes, it looks good. Could you submit it with proper > changelog etc as usual to alsa-devel ML? Then I'll queue it > for 3.18-rc5. > > > thanks, > > Takashi > > > > > BR, > > Kailang > > > > > -----Original Message----- > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > Sent: Wednesday, November 12, 2014 4:08 AM > > > To: Kailang > > > Subject: ALC1150 COEF > > > > > > Hi Kailang, > > > > > > we've got bug reports about the silent output on mobos > with ALC1150 > > > after Windows reboot. It seems that COEF 0x07 bit > > > 0x02 plays some role. Could you check it? > > > https://bugzilla.kernel.org/show_bug.cgi?id=87771 > > > > > > thanks, > > > > > > Takashi > > > > > > ------Please consider the environment before printing this e-mail. > > > > > [2 0000-reset-eapd-control.patch <application/octet-stream > (base64)>] > > >

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[alsa-devel] [PATCH 0/4] Arndale: Adding Sound Machine Driver for ALC5631 Codec
by Krishna Mohan Dani 12 Nov '14

12 Nov '14

These patches add machine driver to instantiate I2S based realtek's ALC5631 based sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec based card hence support for ALC5631 based machine driver is being added. Most of the code has been pulled from insignal's git for Arndale and modified to suit the maintain syntax. Machine driver has been made DT compatible. Krishna Mohan Dani (4): ASoC: ALC5631/RT5631: Add device tree binding documentation ASoC: Samsung: Add arndale_rt5631 machine driver Sound: Kconfig: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631 codec driver Documentation/devicetree/bindings/sound/rt5631.txt | 48 ++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/rt5631.c | 11 ++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 166 ++++++++++++++++++++ 6 files changed, 235 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c -- 1.7.9.5

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[alsa-devel] [PATCH 1/4] ASoC: rt5670: make bias level more reasonable
by bardliao＠realtek.com 12 Nov '14

12 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index c4c42d5..85dc210 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2309,6 +2309,8 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2330,16 +2332,27 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); - snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); - snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); - snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); - snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); snd_soc_update_bits(codec, RT5670_PWR_ANLG1, RT5670_LDO_SEL_MASK, 0x1); break; + case SND_SOC_BIAS_OFF: + if (rt5670->pdata.jd_mode) + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_MB | RT5670_PWR_BG); + else + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + break; default: break; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix memory leak in FTU quirk
by Takashi Iwai 11 Nov '14

11 Nov '14

M-audio FastTrack Ultra quirk doesn't release the kzalloc'ed memory. This patch adds the private_free callback to release it properly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: free the correct field sound/usb/mixer_quirks.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index f119a41ed9a9..7c83bab69dee 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -885,6 +885,11 @@ static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl, return changed; } +static void kctl_private_value_free(struct snd_kcontrol *kctl) +{ + kfree((void *)kctl->private_value); +} + static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, int validx, int bUnitID) { @@ -919,6 +924,7 @@ static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, return -ENOMEM; } + kctl->private_free = kctl_private_value_free; err = snd_ctl_add(mixer->chip->card, kctl); if (err < 0) return err; -- 2.1.3

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix memory leak in FTU quirk
by Takashi Iwai 11 Nov '14

11 Nov '14

M-audio FastTrack Ultra quirk doesn't release the kzalloc'ed memory. This patch adds the private_free callback to release it properly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 45203168ac6d..f6f8acf0446b 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1021,6 +1021,7 @@ static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, return -ENOMEM; } + kctl->private_free = usb_mixer_elem_free; err = snd_ctl_add(mixer->chip->card, kctl); if (err < 0) return err; -- 2.1.3

1 1

[alsa-devel] [PATCH v6 0/3] Add Phytec pbab01 audio support
by Dmitry Lavnikevich 11 Nov '14

11 Nov '14

This version of patchset contains following style changes since v5: - nodes sorted in alphabetical order; - pinctrl entries sorted alphabetically; - new line inserted between nodes.

2 4

[alsa-devel] [PATCH 0/2] Subject: [PATCH 0/2] ASoC: tlv320aic3x: soc_enum array conversion and pop reduction
by Peter Ujfalusi 11 Nov '14

11 Nov '14

Hi, Convert the soc_enum arrays to individual items and add the pop reduction patch on top of this change. Regards, Peter --- Misael Lopez Cruz (1): ASoC: tlv320aic3x: Add output driver pop reduction controls Peter Ujfalusi (1): ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to individual sound/soc/codecs/tlv320aic3x.c | 165 +++++++++++++++++++++++------------------ 1 file changed, 93 insertions(+), 72 deletions(-) -- 2.1.3

2 3

[alsa-devel] Realtek HDA SKU/assembly parsing
by Daniel Drake 11 Nov '14

11 Nov '14

Hi, I see that there is a function that parses the subsystem ID for various realtek codecs, alc_auto_parse_customize_define Is there a public specification that documents the specific meaning of each bit here? Thanks Daniel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH] ASoC: rt5645: remove unused rt5645_clk_sel_put
by Bard Liao 11 Nov '14

11 Nov '14

Remove rt5645_clk_sel_put function since it is never used. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 59 ----------------------------------------------- 1 file changed, 59 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1adab38..d6298b3 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -443,65 +443,6 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); -static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); - unsigned int u_bit = 0, p_bit = 0; - struct soc_enum *em = - (struct soc_enum *)kcontrol->private_value; - - switch (em->reg) { - case RT5645_ASRC_2: - switch (em->shift_l) { - case 0: - u_bit = 0x8; - p_bit = RT5645_PWR_ADC_S1F; - break; - case 4: - u_bit = 0x100; - p_bit = RT5645_PWR_DAC_MF_R; - break; - case 8: - u_bit = 0x200; - p_bit = RT5645_PWR_DAC_MF_L; - break; - case 12: - u_bit = 0x400; - p_bit = RT5645_PWR_DAC_S1F; - break; - } - break; - case RT5645_ASRC_3: - switch (em->shift_l) { - case 0: - u_bit = 0x1; - p_bit = RT5645_PWR_ADC_MF_R; - break; - case 4: - u_bit = 0x2; - p_bit = RT5645_PWR_ADC_MF_L; - break; - } - break; - } - - if (u_bit || p_bit) { - switch (ucontrol->value.integer.value[0]) { - case 1 ... 4: /*enable*/ - if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) - snd_soc_update_bits(codec, - RT5645_ASRC_1, u_bit, u_bit); - break; - default: /*disable*/ - snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); - break; - } - } - - return snd_soc_put_enum_double(kcontrol, ucontrol); -} - static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ASoC: rt5670: add rt5672 codec support
by bardliao＠realtek.com 11 Nov '14

11 Nov '14

From: Bard Liao <bardliao(a)realtek.com> rt5672 is very similar to rt5670. Therefore we use one codec driver to support both codecs. The difference between rt5670 and rt5672 is there is some difference in their dapm routing table. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Add { "rt5671", 0 }, in rt5670_i2c_id[] --- sound/soc/codecs/rt5670.c | 71 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/rt5670.h | 6 ++++ 2 files changed, 65 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9bd8b4f..c4c42d5 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1595,29 +1595,40 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { /* PDM */ SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, RT5670_PWR_PDM1_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, - RT5670_PWR_PDM2_BIT, 0, NULL, 0), SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL, RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux), SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL, RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux), - SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, - RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), - SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, - RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), /* Output Lines */ SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), SND_SOC_DAPM_OUTPUT("PDM1L"), SND_SOC_DAPM_OUTPUT("PDM1R"), SND_SOC_DAPM_OUTPUT("PDM2L"), SND_SOC_DAPM_OUTPUT("PDM2R"), }; +static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = { + SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPOLP"), + SND_SOC_DAPM_OUTPUT("SPOLN"), + SND_SOC_DAPM_OUTPUT("SPORP"), + SND_SOC_DAPM_OUTPUT("SPORN"), +}; + static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1970,12 +1981,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, { "PDM1 R Mux", NULL, "PDM1 Power" }, - { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, - { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, - { "PDM2 L Mux", NULL, "PDM2 Power" }, - { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, - { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, - { "PDM2 R Mux", NULL, "PDM2 Power" }, { "HP Amp", NULL, "HPO MIX" }, { "HP Amp", NULL, "Mic Det Power" }, @@ -1993,13 +1998,30 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "LOUTR", NULL, "LOUT R Playback" }, { "LOUTL", NULL, "Improve HP Amp Drv" }, { "LOUTR", NULL, "Improve HP Amp Drv" }, +}; +static const struct snd_soc_dapm_route rt5670_specific_dapm_routes[] = { + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, { "PDM1L", NULL, "PDM1 L Mux" }, { "PDM1R", NULL, "PDM1 R Mux" }, { "PDM2L", NULL, "PDM2 L Mux" }, { "PDM2R", NULL, "PDM2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5672_specific_dapm_routes[] = { + { "SPO Amp", NULL, "PDM1 L Mux" }, + { "SPO Amp", NULL, "PDM1 R Mux" }, + { "SPOLP", NULL, "SPO Amp" }, + { "SPOLN", NULL, "SPO Amp" }, + { "SPORP", NULL, "SPO Amp" }, + { "SPORN", NULL, "SPO Amp" }, +}; + static int rt5670_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2331,6 +2353,29 @@ static int rt5670_probe(struct snd_soc_codec *codec) { struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (snd_soc_read(codec, RT5670_RESET) & RT5670_ID_MASK) { + case RT5670_ID_5670: + case RT5670_ID_5671: + snd_soc_dapm_new_controls(&codec->dapm, + rt5670_specific_dapm_widgets, + ARRAY_SIZE(rt5670_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5670_specific_dapm_routes, + ARRAY_SIZE(rt5670_specific_dapm_routes)); + break; + case RT5670_ID_5672: + snd_soc_dapm_new_controls(&codec->dapm, + rt5672_specific_dapm_widgets, + ARRAY_SIZE(rt5672_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5672_specific_dapm_routes, + ARRAY_SIZE(rt5672_specific_dapm_routes)); + break; + default: + dev_err(codec->dev, + "The driver is for RT5670 RT5671 or RT5672 only\n"); + return -ENODEV; + } rt5670->codec = codec; return 0; @@ -2452,6 +2497,8 @@ static const struct regmap_config rt5670_regmap = { static const struct i2c_device_id rt5670_i2c_id[] = { { "rt5670", 0 }, + { "rt5671", 0 }, + { "rt5672", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index a0b5c85..d11b9c2 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -228,6 +228,12 @@ #define RT5670_R_VOL_MASK (0x3f) #define RT5670_R_VOL_SFT 0 +/* SW Reset & Device ID (0x00) */ +#define RT5670_ID_MASK (0x3 << 1) +#define RT5670_ID_5670 (0x0 << 1) +#define RT5670_ID_5672 (0x1 << 1) +#define RT5670_ID_5671 (0x2 << 1) + /* Combo Jack Control 1 (0x0a) */ #define RT5670_CBJ_BST1_MASK (0xf << 12) #define RT5670_CBJ_BST1_SFT (12) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/6] ASoC: davinci-mcasp: fixes and over/underrun handling
by Peter Ujfalusi 11 Nov '14

11 Nov '14

Hi, Some fixes for the mcasp code: - rx formating fix - constraint for number of channels and sample bits handling of over/underrun: When such event happens (in McASP vs DMA level) we force restart the stream in order to get McASP and the DMA in sync again. Regards, Peter --- Misael Lopez Cruz (2): ASoC: davinci-mcasp: Active slots depend on active serializers ASoC: davinci-mcasp: Add overrun/underrun event handling Peter Ujfalusi (4): ASoC: davinci-mcasp: Validate tdm_slots parameter at probe time ASoC: davinci-mcasp: Place constraint on number of channels ASoC: davinci-mcasp: Symmetric sample bits for IIS mode ASoC: davinci-mcasp: Fix rx format when more bclk is used on the bus .../bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 212 +++++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 10 + 3 files changed, 210 insertions(+), 14 deletions(-) -- 2.1.3

2 14

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Add output driver pop reduction controls
by Peter Ujfalusi 11 Nov '14

11 Nov '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Output driver has two parameters that can be configured to reduce pop noise: power-on delay and ramp-up step time. Two new kcontrols have been added to set these parameters. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f7c2a575a892..70f8b8be9173 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -270,6 +270,15 @@ static const struct soc_enum aic3x_agc_decay_enum[] = { SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), }; +static const char * const aic3x_poweron_time[] = { + "0us", "10us", "100us", "1ms", "10ms", "50ms", + "100ms", "200ms", "400ms", "800ms", "2s", "4s" }; +static const char * const aic3x_rampup_step[] = { "0ms", "1ms", "2ms", "4ms" }; +static const struct soc_enum aic3x_pop_reduction_enum[] = { + SOC_ENUM_SINGLE(HPOUT_POP_REDUCTION, 4, 12, aic3x_poweron_time), + SOC_ENUM_SINGLE(HPOUT_POP_REDUCTION, 2, 4, aic3x_rampup_step), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -399,6 +408,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + /* Pop reduction */ + SOC_ENUM("Output Driver Power-On time", aic3x_pop_reduction_enum[0]), + SOC_ENUM("Output Driver Ramp-up step", aic3x_pop_reduction_enum[1]), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { -- 2.1.3

2 6

[alsa-devel] [PATCH 0/2 v6] Digidesign Mbox 1 enhancements
by Damien Zammit 11 Nov '14

11 Nov '14

Revision 6. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign Mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 ++++++++++++---- 3 files changed, 171 insertions(+), 10 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: Fix kerneldoc errors
by Takashi Iwai 11 Nov '14

11 Nov '14

Lots of typos and missing parameter descriptions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 11 ++++++++--- sound/soc/soc-core.c | 21 ++++++++++++++------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-devres.c | 2 +- sound/soc/soc-jack.c | 2 +- 5 files changed, 25 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130037a0..25d0bdbbeb11 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -583,6 +583,7 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * + * @list: linked list of assigned pins to the same jack * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported @@ -602,7 +603,7 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container + * @list: list container */ struct snd_soc_jack_zone { unsigned int min_mv; @@ -618,16 +619,20 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debouce time in ms * @wake: enable as wake source + * @jack: assigned jack + * @work: delayed_work for GPIO detection + * @desc: GPIO descriptor * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an * ADC). + * @data: arbitrary data passed to jack_status_check callback */ struct snd_soc_jack_gpio { unsigned int gpio; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a23a0e9..74a8a80e2d9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2113,6 +2113,8 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); /** * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * @ops: AC97 opts to set + * @pdev: platform to deal with * * This function sets the reset and warm_reset properties of ops and parses * the device node of pdev to get pinctrl states and gpio numbers to use. @@ -2685,7 +2687,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * @@ -3458,7 +3460,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -4082,6 +4084,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core + * @dev: device to unregister * */ void snd_soc_unregister_component(struct device *dev) @@ -4119,7 +4122,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -4158,7 +4161,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: platform to register + * @platform_drv: the driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -4215,7 +4219,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -4306,7 +4310,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: codec to register + * @codec_drv: the codec driver to register + * @dai_drv: the array of DAI drivers to register + * @num_dai: the number of DAIs to register */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -4406,7 +4413,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..7050ec1d5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2680,7 +2680,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card object containing widgets * * Checks the codec for any new dapm widgets and creates them if found. * diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 057e5ef7dcce..a57921eeee81 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res) /** * devm_snd_soc_register_platform - resource managed platform registration * @dev: Device used to manage platform - * @platform: platform to register + * @platform_drv: platform to register * * Register a platform driver with automatic unregistration when the device is * unregistered. diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..ef1d42d7c6f6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -116,7 +116,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * * @jack: ASoC jack * @count: Number of zones - * @zone: Array of zones + * @zones: Array of zones * * After this function has been called the zones specified in the * array will be associated with the jack. -- 2.1.3

2 8

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add ASRC support
by bardliao＠realtek.com 11 Nov '14

11 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch add ASRC support for rt5645 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 144 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 144 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc442f4..3131e91 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -443,6 +443,65 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); +static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int u_bit = 0, p_bit = 0; + struct soc_enum *em = + (struct soc_enum *)kcontrol->private_value; + + switch (em->reg) { + case RT5645_ASRC_2: + switch (em->shift_l) { + case 0: + u_bit = 0x8; + p_bit = RT5645_PWR_ADC_S1F; + break; + case 4: + u_bit = 0x100; + p_bit = RT5645_PWR_DAC_MF_R; + break; + case 8: + u_bit = 0x200; + p_bit = RT5645_PWR_DAC_MF_L; + break; + case 12: + u_bit = 0x400; + p_bit = RT5645_PWR_DAC_S1F; + break; + } + break; + case RT5645_ASRC_3: + switch (em->shift_l) { + case 0: + u_bit = 0x1; + p_bit = RT5645_PWR_ADC_MF_R; + break; + case 4: + u_bit = 0x2; + p_bit = RT5645_PWR_ADC_MF_L; + break; + } + break; + } + + if (u_bit || p_bit) { + switch (ucontrol->value.integer.value[0]) { + case 1 ... 4: /*enable*/ + if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) + snd_soc_update_bits(codec, + RT5645_ASRC_1, u_bit, u_bit); + break; + default: /*disable*/ + snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); + break; + } + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -554,6 +613,53 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + + switch (source->shift) { + case 0: + reg = RT5645_ASRC_3; + shift = 0; + break; + case 1: + reg = RT5645_ASRC_3; + shift = 4; + break; + case 3: + reg = RT5645_ASRC_2; + shift = 0; + break; + case 8: + reg = RT5645_ASRC_2; + shift = 4; + break; + case 9: + reg = RT5645_ASRC_2; + shift = 8; + break; + case 10: + reg = RT5645_ASRC_2; + shift = 12; + break; + default: + return 0; + } + + val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + switch (val) { + case 1: + case 2: + case 3: + case 4: + return 1; + default: + return 0; + } + +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, @@ -1246,6 +1352,30 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5645_PWR_VOL, RT5645_PWR_MIC_DET_BIT, 0, NULL, 0), + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5645_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5645_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5645_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5645_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5645_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5645_ASRC_1, + 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5645_ASRC_1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5645_ASRC_1, + 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5645_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5645_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5645_ASRC_1, + 0, 0, NULL, 0), + /* Input Side */ /* micbias */ SND_SOC_DAPM_MICBIAS("micbias1", RT5645_PWR_ANLG2, @@ -1504,6 +1634,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { + { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "dac mono left filter", NULL, "DAC MONO L ASRC", is_using_asrc }, + { "dac mono right filter", NULL, "DAC MONO R ASRC", is_using_asrc }, + { "dac stereo1 filter", NULL, "DAC STO ASRC", is_using_asrc }, + + { "I2S1", NULL, "I2S1 ASRC" }, + { "I2S2", NULL, "I2S2 ASRC" }, + { "IN1P", NULL, "LDO2" }, { "IN2P", NULL, "LDO2" }, @@ -1550,12 +1691,15 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC" }, { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC" }, { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC" }, { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] Issue with hda_intel on i7+Z97 chipset
by Daniel Silverstone 10 Nov '14

10 Nov '14

Hi, Firstly, I hope this is a suitable way to report this issue. I am using Debian/Wheezy on a reasonably recent i7 based system which has a Z97 motherboard chipset. In Windows, sound is perfect, but in Linux it's very crackly (direct through alsa) and if I bring pulseaudio into the mix then it reports buffer/interrupt issues on the microphone along the lines of: Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: ALSA woke us up to read new data from the device, but there was actually nothing to read! Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Similar messages are shown for issues with playback: user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. In both cases, sound exhibts crackles, pops, crunches, stalls, repeats etc. I can mitigate the issue by working one or more of the CPU cores hard with bzip2 -c </dev/urandom > /dev/null Of course, this makes my system fans run harder etc, which is a bit of a pain for me, and it's only masking the issue, not solving it. For reference, my motherboard's dmi shows: Manufacturer: Gigabyte Technology Co., Ltd. Product Name: Z97-D3H-CF And the CPU reports as: Family: Core i7 Manufacturer: Intel ID: C3 06 03 00 FF FB EB BF Signature: Type 0, Family 6, Model 60, Stepping 3 Version: Intel(R) Core(TM) i7-4790K CPU @ 4.00GHz The relevant PCI devices are: 00:03.0 0403: 8086:0c0c (rev 06) Subsystem: 8086:2010 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 51 Region 0: Memory at f7834000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel and 00:1b.0 0403: 8086:8ca0 Subsystem: 1458:a182 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at f7830000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel and it's possible that this kernel message is relevant: [ 17.248493] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [ 17.251735] hda-intel: Invalid position buffer, using LPIB read method instead. I have tried a number of bdl_pos_adj between 1 and 32 and I have tried enable_msi=1 Neither seem to have helped significantly. I am currently running 3.12-0.bpo.1-amd64 from Debian's wheezy-backports repository. Unfortunately anything newer seems to lock up during boot and I've not managed to track down what or why as yet. I was hoping that you may have some idea of what is going on and either some suggestions for improving my situation on 3.12 or else some desire for further debug data such that a newer kernel (once I can persuade one to boot) may perform better for me. Please let me know if you require further details from me in order to address this. Regards, Daniel. -- Daniel Silverstone http://www.digital-scurf.org/ PGP mail accepted and encouraged. Key Id: 3CCE BABE 206C 3B69

1 1

[alsa-devel] [PATCH 0/3] ASoC: twl4030: support routine to external VOICE source.
by NeilBrown 10 Nov '14

10 Nov '14

The 'voice' port of the twl4030 on my board is connected to a GSM modem, not to the CPU. As such it is not visible to ASLA and normal approaches to configuring the interface (as one end of a DAI) don't apply. I need a way to tell the twl4030 that the connected device will be master of 'clk' and 'FRM', and whether they are inverted. Using device tree seems the correct approach - I am describing the properties of the hardware attached the 'voice' port. Using magic numbers from include/sound/soc-dai.h almost certainly is not the right way to represent this information, but it isn't clear to me what a good way would be. So the current patch uses magic numbers as a proof of concept, and I'm asking if there is any establish precedent, or any suggestions on how to record in device tree the 'master' and 'polarity' of CLK and FRM signals for an external device. [I currently use (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS)] This is in the second patch. The first silences a pointless warning (when it is pointless) and the last allowed the voice linked to be turned on or off via ALSA controls. If the first and last can be applied without finalizing the device-tree part yet, that would be great. Thanks, NeilBrown --- NeilBrown (3): ASoC: twl4030: don't report EBUSY if no change requested. ASoC: twl4030: allow voice port to be connected externally. ASoC: twl4030: enable routing audio to 'voice' interface. .../devicetree/bindings/mfd/twl4030-audio.txt | 7 ++ include/linux/i2c/twl.h | 3 + sound/soc/codecs/twl4030.c | 78 ++++++++++++++++++-- 3 files changed, 82 insertions(+), 6 deletions(-) -- Signature

4 15

[alsa-devel] [PATCH] ASoC: mxs: mxs-saif: Register the irq with the device name
by Fabio Estevam 10 Nov '14

10 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Instead of registering the irq name with the driver name, it's better to pass the device name so that we have a more explicit indication as to what saif instance the irq is related: $ cat /proc/interrupts CPU0 ... 214: 4 - 59 80042000.saif 215: 0 - 58 80046000.saif Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index eade2742..d986508 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -773,7 +773,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->dev = &pdev->dev; ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, - "mxs-saif", saif); + dev_name(&pdev->dev), saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/2] Fix some 'const' problems
by Jean-Francois Moine 10 Nov '14

10 Nov '14

A cast was often needed to call some OF functions, removing a 'const' attribute. This patchset removes 'const' from the device_node pointers of the sound/soc structures and also removes some useless casts in the sound/soc tree. v2: remove const from the pointers instead of modifying the OF functions (Russell King) Jean-Francois Moine (2): ASoC: Remove 'const' from the device_node pointers ASoC: simple-card: Remove useless casts include/sound/soc.h | 6 +++--- sound/soc/generic/simple-card.c | 7 ++----- sound/soc/samsung/odroidx2_max98090.c | 4 ++-- sound/soc/ux500/mop500.c | 6 ++---- 4 files changed, 9 insertions(+), 14 deletions(-) -- 2.1.3

3 8

[alsa-devel] [PATCH 0/8] ASoC: Replace remaining users of the CODEC mutex
by Lars-Peter Clausen 10 Nov '14

10 Nov '14

We've stopped using the snd_soc_codec mutex in the ASoC core itself a while ago. There is still a handful of drivers which use the mutex to protect some of their data structures. This patch moves the lock for the remaining users to the drivers private data struct and then removes it from the snd_soc_codec struct. - Lars Lars-Peter Clausen (8): ASoC: ab8500-codec: Move control lock to the driver level ASoC: max98095: Move mutex to the driver level ASoC: wm5102: Move ultrasonic response settings lock to the driver level ASoC: wm8731: Move the deemph lock to the driver level ASoC: wm8903: Move the deemph lock to the driver level ASoC: wm8958: Move DSP firmware lock to driver level ASoC: wm8962: Move DSP enable lock to the driver level ASoC: Remove CODEC mutex include/linux/mfd/arizona/core.h | 1 + include/sound/soc.h | 1 - sound/soc/codecs/ab8500-codec.c | 32 ++++++++++++++++---------------- sound/soc/codecs/arizona.c | 4 ++-- sound/soc/codecs/max98095.c | 12 ++++++++---- sound/soc/codecs/wm5102.c | 16 ++++++++-------- sound/soc/codecs/wm8731.c | 9 +++++++-- sound/soc/codecs/wm8903.c | 8 ++++++-- sound/soc/codecs/wm8958-dsp2.c | 12 ++++++------ sound/soc/codecs/wm8962.c | 8 ++++++-- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm8994.h | 2 ++ sound/soc/soc-core.c | 1 - 13 files changed, 64 insertions(+), 44 deletions(-) -- 1.8.0

3 10

[alsa-devel] [PATCH 0/4] Yet more kenreldoc fixes
by Takashi Iwai 10 Nov '14

10 Nov '14

Hi, these are a series of patches to fix kerneldoc errors / warnings. These aren't actaully compiled from DocBook directory, so they are harmless, but better to be fixed, of course. [PATCH 1/4] ALSA: vx: Fix missing kenreldoc parameter descriptions [PATCH 2/4] ALSA: hda - Fix kerneldoc errors in patch_ca0132.c [PATCH 3/4] ALSA: mixart: Fix kerneldoc comments [PATCH 4/4] ALSA: Fix invalid kerneldoc markers Takashi

2 8

[alsa-devel] [PATCH 0/5 v2] ASoC: rsnd: Add Volume Ramp support
by Kuninori Morimoto 10 Nov '14

10 Nov '14

Hi Mark These are v2 of Renesas R-Car sound Volume Ramp support. It uses "Volume Ramp Up/Down Rate" on last patch instead of "Period". Because it is good match naming, and it was difficult to calculate real period time on this HW. Kuninori Morimoto (5): ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() ASoC: rsnd: enable multiple DVC valume settings ASoC: rsnd: enable single DVC valume settings ASoC: rsnd: Add Volume Ramp support sound/soc/sh/rcar/dvc.c | 126 +++++++++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/gen.c | 3 ++ sound/soc/sh/rcar/rsnd.h | 6 +++ 3 files changed, 116 insertions(+), 19 deletions(-) Best regards --- Kuninori Morimoto

2 21

[alsa-devel] [PATCH] ASoC: rsnd: fallback to PIO mode if DMA mode was failed
by Kuninori Morimoto 10 Nov '14

10 Nov '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current Renesas R-Car sound driver probe will be failed if it try to use DMA mode and it couldn't use for some reasons. But PIO mode works even though in such case. This patch try to fallback to PIO mode if DMA mode probing was failed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 74 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/sh/rcar/ssi.c | 15 ++++++++++ 2 files changed, 85 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5205618..110b99d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -349,7 +349,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_name); if (!dma->chan) { dev_err(dev, "can't get dma channel\n"); - return -EIO; + goto rsnd_dma_channel_err; } ret = dmaengine_slave_config(dma->chan, &cfg); @@ -363,8 +363,15 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, rsnd_dma_init_err: rsnd_dma_quit(priv, dma); +rsnd_dma_channel_err: - return ret; + /* + * DMA failed. try to PIO mode + * see + * rsnd_ssi_dma_remove() + * rsnd_rdai_continuance_probe() + */ + return -EAGAIN; } void rsnd_dma_quit(struct rsnd_priv *priv, @@ -456,6 +463,13 @@ static int rsnd_dai_connect(struct rsnd_mod *mod, return 0; } +static void rsnd_dai_disconnect(struct rsnd_mod *mod, + struct rsnd_dai_stream *io) +{ + mod->io = NULL; + io->mod[mod->type] = NULL; +} + int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) { int id = rdai - priv->rdai; @@ -686,6 +700,20 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { ret; \ }) +#define rsnd_path_break(priv, io, type) \ +{ \ + struct rsnd_mod *mod; \ + int id = -1; \ + \ + if (rsnd_is_enable_path(io, type)) { \ + id = rsnd_info_id(priv, io, type); \ + if (id >= 0) { \ + mod = rsnd_##type##_mod_get(priv, id); \ + rsnd_dai_disconnect(mod, io); \ + } \ + } \ +} + static int rsnd_path_init(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -977,6 +1005,44 @@ static const struct snd_soc_component_driver rsnd_soc_component = { .name = "rsnd", }; +static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + int is_play) +{ + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; + int ret; + + ret = rsnd_dai_call(probe, io, rdai); + if (ret == -EAGAIN) { + /* + * Fallback to PIO mode + */ + + /* + * call "remove" for SSI/SRC/DVC + * SSI will be switch to PIO mode if it was DMA mode + * see + * rsnd_dma_init() + * rsnd_ssi_dma_remove() + */ + rsnd_dai_call(remove, io, rdai); + + /* + * remove SRC/DVC from DAI, + */ + rsnd_path_break(priv, io, src); + rsnd_path_break(priv, io, dvc); + + /* + * retry to "probe". + * DAI has SSI which is PIO mode only now. + */ + ret = rsnd_dai_call(probe, io, rdai); + } + + return ret; +} + /* * rsnd probe */ @@ -1038,11 +1104,11 @@ static int rsnd_probe(struct platform_device *pdev) } for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(probe, &rdai->playback, rdai); + ret = rsnd_rdai_continuance_probe(priv, rdai, 1); if (ret) goto exit_snd_probe; - ret = rsnd_dai_call(probe, &rdai->capture, rdai); + ret = rsnd_rdai_continuance_probe(priv, rdai, 0); if (ret) goto exit_snd_probe; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index cae08b7..346d3dc 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -465,8 +465,23 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod)); + /* + * fallback to PIO + * + * SSI .probe might be called again. + * see + * rsnd_rdai_continuance_probe() + */ + mod->ops = &rsnd_ssi_pio_ops; + + dev_info(dev, "%s[%d] fallback to PIO mode\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup debug message format and timing
by Kuninori Morimoto 10 Nov '14

10 Nov '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current Renesas R-Car sound driver debug message is using random format (ex "ssi0: xxx" / "SSI0 xxx" / "ssi[0]: xxx") and confusable timing ("xxx probe failed" and "xxx probed" are shown in same time) This patch fixes these Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/dvc.c | 3 ++- sound/soc/sh/rcar/gen.c | 6 +++--- sound/soc/sh/rcar/src.c | 11 +++++++---- sound/soc/sh/rcar/ssi.c | 25 ++++++++++++++++--------- 5 files changed, 29 insertions(+), 18 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1922ec5..5205618 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -409,7 +409,7 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod) ({ \ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \ struct device *dev = rsnd_priv_to_dev(priv); \ - dev_dbg(dev, "%s [%d] %s\n", \ + dev_dbg(dev, "%s[%d] %s\n", \ rsnd_mod_name(mod), rsnd_mod_id(mod), #func); \ (mod)->ops->func(mod, rdai); \ }) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index d6d8e64..e2c8473 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -141,7 +141,8 @@ static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "%s (Gen2) is probed\n", rsnd_mod_name(mod)); + dev_dbg(dev, "%s[%d] (Gen2) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return 0; } diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 4cb3202..87a6f2d 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -78,7 +78,7 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; - dev_dbg(dev, "r %s(%d) - %4d : %08x\n", + dev_dbg(dev, "r %s[%d] - %4d : %08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, val); regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); @@ -96,7 +96,7 @@ void rsnd_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "w %s(%d) - %4d : %08x\n", + dev_dbg(dev, "w %s[%d] - %4d : %08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data); regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); @@ -111,7 +111,7 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "b %s(%d) - %4d : %08x/%08x\n", + dev_dbg(dev, "b %s[%d] - %4d : %08x/%08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data, mask); regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 9183e01..4679501 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -438,7 +438,8 @@ static int rsnd_src_probe_gen1(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "%s (Gen1) is probed\n", rsnd_mod_name(mod)); + dev_dbg(dev, "%s[%d] (Gen1) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return 0; } @@ -578,9 +579,11 @@ static int rsnd_src_probe_gen2(struct rsnd_mod *mod, rsnd_info_is_playback(priv, src), src->info->dma_id); if (ret < 0) - dev_err(dev, "SRC DMA failed\n"); - - dev_dbg(dev, "%s (Gen2) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (Gen2) failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (Gen2) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 34e8400..cae08b7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -159,7 +159,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(j); - dev_dbg(dev, "ssi%d outputs %u Hz\n", + dev_dbg(dev, "%s[%d] outputs %u Hz\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod), rate); return 0; @@ -206,7 +207,8 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, ssi->usrcnt++; - dev_dbg(dev, "ssi%d hw started\n", rsnd_mod_id(&ssi->mod)); + dev_dbg(dev, "%s[%d] hw started\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod)); } static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, @@ -249,7 +251,8 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, clk_disable_unprepare(ssi->clk); } - dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); + dev_dbg(dev, "%s[%d] hw stopped\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod)); } /* @@ -385,9 +388,11 @@ static int rsnd_ssi_pio_probe(struct rsnd_mod *mod, IRQF_SHARED, dev_name(dev), ssi); if (ret) - dev_err(dev, "SSI request interrupt failed\n"); - - dev_dbg(dev, "%s (PIO) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (PIO) request interrupt failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (PIO) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } @@ -448,9 +453,11 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, dma_id); if (ret < 0) - dev_err(dev, "SSI DMA failed\n"); - - dev_dbg(dev, "%s (DMA) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (DMA) is failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (DMA) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/4] Arndale: Adding Sound Machine Driver for ALC5631 Codec
by Krishna Mohan Dani 10 Nov '14

10 Nov '14

These patches add machine driver to instantiate I2S based realtek's ALC5631 based sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec based card hence support for ALC5631 based machine driver is being added. Most of the code has been pulled from insignal's git for Arndale and modified to suit the maintain syntax. Machine driver has been made DT compatible. Krishna Mohan Dani (4): ASoC: ALC5631: Add device tree binding documentation Sound: Samsung: Add sound card machine driver for Arndale Board Sound: Kconfig: Adding the description of the codec Sound: DT: Adding Device Tree compatibility to Realtek's ALC5631 codec driver Documentation/devicetree/bindings/sound/rt5631.txt | 41 +++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/rt5631.c | 11 ++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 167 ++++++++++++++++++++ 6 files changed, 229 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c -- 1.7.9.5

3 7

[alsa-devel] [PATCH] ASoC: davinvi-mcasp: Balance pm_runtime_enable() on probe failure
by Anil Kumar 10 Nov '14

10 Nov '14

If probe fails then we need to call pm_runtime_disable() to balance out the previous pm_runtime_enable() call. Signed-off-by: Anil Kumar <anilk4.v(a)gmail.com> --- :100644 100644 0eed9b1... 99c747d... M sound/soc/davinci/davinci-mcasp.c sound/soc/davinci/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0eed9b1..99c747d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1235,6 +1235,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = pm_runtime_get_sync(&pdev->dev); if (IS_ERR_VALUE(ret)) { dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); + pm_runtime_disable(&pdev->dev); return ret; } -- 1.9.2

3 2

[alsa-devel] sound: enable sound support on n900 on devicetree-based boot
by Pavel Machek 10 Nov '14

10 Nov '14

With device tree, it is possible (and encouraged) to build N900 kernels without CONFIG_MACH_NOKIA_RX51. Update config file to enable the driver build in this case. This makes sound work on my n900 under 3.18-rc1. Signed-off-by: Pavel Machek <pavel(a)ucw.cz> diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d44463a..0d7b707 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) && I2C + depends on SND_OMAP_SOC && ARM && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

2 4

[alsa-devel] [PATCH 0/2 v5] Digidesign Mbox 1 enhancements
by Damien Zammit 10 Nov '14

10 Nov '14

This is revision 5. I made a mistake interpreting your reviews, sorry guys. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 +++ sound/usb/mixer_quirks.c | 140 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 +++++++++++---- 3 files changed, 182 insertions(+), 10 deletions(-) -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: korg1212: cleanup of printk
by Sudip Mukherjee 10 Nov '14

10 Nov '14

replaced all references of the debug messages via printk with dev_* macro (mostly dev_dbg). one reference was changed to pr_err as there the card might have been uninitialized. while compiling after these modifications, we are getting three warnings in function snd_korg1212_create: 2301: warning: format "%d" expects argument of type "int", but argument 6 has type "long unsigned int" [-Wformat] 2376: warning: format "%x" expects argument of type "unsigned int", but argument 5 has type "dma_addr_t" [-Wformat] 2376: warning: format "%d" expects argument of type "int", but argument 6 has type "size_t" [-Wformat] these warnings were prviously not visible as K1212_DEBUG_LEVEL was defined as 0. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/korg1212/korg1212.c | 406 +++++++++++++++++++++++------------------- 1 file changed, 219 insertions(+), 187 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 59d21c9..ab8e805 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -39,21 +39,6 @@ #include <asm/io.h> // ---------------------------------------------------------------------------- -// Debug Stuff -// ---------------------------------------------------------------------------- -#define K1212_DEBUG_LEVEL 0 -#if K1212_DEBUG_LEVEL > 0 -#define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) -#else -#define K1212_DEBUG_PRINTK(fmt,...) -#endif -#if K1212_DEBUG_LEVEL > 1 -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args) -#else -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,...) -#endif - -// ---------------------------------------------------------------------------- // Record/Play Buffer Allocation Method. If K1212_LARGEALLOC is defined all // buffers are alocated as a large piece inside KorgSharedBuffer. // ---------------------------------------------------------------------------- @@ -530,12 +515,12 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, int rc = K1212_CMDRET_Success; if (!korg1212->outDoorbellPtr) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: CardUninitialized\n"); + pr_err("snd_korg1212_Send1212Command: Card Uninitialized\n"); return K1212_CMDRET_CardUninitialized; } - K1212_DEBUG_PRINTK("K1212_DEBUG: Card <- 0x%08x 0x%08x [%s]\n", - doorbellVal, mailBox0Val, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Card <- 0x%08x 0x%08x [%s]\n", + doorbellVal, mailBox0Val, stateName[korg1212->cardState]); for (retryCount = 0; retryCount < MAX_COMMAND_RETRIES; retryCount++) { writel(mailBox3Val, korg1212->mailbox3Ptr); writel(mailBox2Val, korg1212->mailbox2Ptr); @@ -562,7 +547,7 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, mailBox3Lo = readl(korg1212->mailbox3Ptr); if (mailBox3Lo & COMMAND_ACK_MASK) { if ((mailBox3Lo & DOORBELL_VAL_MASK) == (doorbellVal & DOORBELL_VAL_MASK)) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- Success\n"); + dev_dbg(korg1212->card->dev, "Card <- Success\n"); rc = K1212_CMDRET_Success; break; } @@ -571,7 +556,7 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, korg1212->cmdRetryCount += retryCount; if (retryCount >= MAX_COMMAND_RETRIES) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- NoAckFromCard\n"); + dev_err(korg1212->card->dev, "Card <- NoAckFromCard\n"); rc = K1212_CMDRET_NoAckFromCard; } @@ -612,20 +597,21 @@ static void snd_korg1212_timer_func(unsigned long data) korg1212->stop_pending_cnt = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Stop ack'ed [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Stop ack'ed [%s]\n", + stateName[korg1212->cardState]); } else { if (--korg1212->stop_pending_cnt > 0) { /* reprogram timer */ korg1212->timer.expires = jiffies + 1; add_timer(&korg1212->timer); } else { - snd_printd("korg1212_timer_func timeout\n"); + dev_err(korg1212->card->dev, + "korg1212_timer_func timeout\n"); korg1212->sharedBufferPtr->cardCommand = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); - K1212_DEBUG_PRINTK("K1212_DEBUG: Stop timeout [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Stop timeout [%s]\n", + stateName[korg1212->cardState]); } } spin_unlock_irqrestore(&korg1212->lock, flags); @@ -660,8 +646,8 @@ static inline void snd_korg1212_setCardState(struct snd_korg1212 * korg1212, enu static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: OpenCard [%s] %d\n", - stateName[korg1212->cardState], korg1212->opencnt); + dev_dbg(korg1212->card->dev, "K1212_DEBUG: OpenCard [%s] %d\n", + stateName[korg1212->cardState], korg1212->opencnt); mutex_lock(&korg1212->open_mutex); if (korg1212->opencnt++ == 0) { snd_korg1212_TurnOffIdleMonitor(korg1212); @@ -674,8 +660,8 @@ static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard [%s] %d\n", - stateName[korg1212->cardState], korg1212->opencnt); + dev_dbg(korg1212->card->dev, "CloseCard [%s] %d\n", + stateName[korg1212->cardState], korg1212->opencnt); mutex_lock(&korg1212->open_mutex); if (--(korg1212->opencnt)) { @@ -687,8 +673,9 @@ static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_StopPlay, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "CloseCard - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { mutex_unlock(&korg1212->open_mutex); return 0; @@ -711,8 +698,8 @@ static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->setcnt); + dev_dbg(korg1212->card->dev, "SetupForPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->setcnt); if (korg1212->setcnt++) return 0; @@ -721,8 +708,8 @@ static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212) rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_SetupPlay, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "SetupForPlay - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { return 1; } @@ -734,8 +721,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->playcnt); + dev_dbg(korg1212->card->dev, "TriggerPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->playcnt); if (korg1212->playcnt++) return 0; @@ -743,8 +730,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) snd_korg1212_setCardState(korg1212, K1212_STATE_PLAYING); rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_TriggerPlay, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "TriggerPlay - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { return 1; } @@ -754,8 +741,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) /* spinlock already held */ static int snd_korg1212_StopPlay(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: StopPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->playcnt); + dev_dbg(korg1212->card->dev, "StopPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->playcnt); if (--(korg1212->playcnt)) return 0; @@ -866,8 +853,9 @@ static int snd_korg1212_SetRate(struct snd_korg1212 *korg1212, int rate) ClockSourceSelector[korg1212->clkSrcRate], 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Set Clock Source Selector - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); return 0; } @@ -902,8 +890,8 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) u16 count; unsigned long flags; - K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "WriteADCSensivity [%s]\n", + stateName[korg1212->cardState]); // ---------------------------------------------------------------------------- // initialize things. The local init bit is always set when writing to the @@ -1026,8 +1014,9 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_MonitorOn, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "WriteADCSensivity - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); } spin_unlock_irqrestore(&korg1212->lock, flags); @@ -1039,8 +1028,8 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) { int channel, rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is complete. [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "DSP download is complete. [%s]\n", + stateName[korg1212->cardState]); // ---------------------------------------------------- // tell the card to boot @@ -1048,8 +1037,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_BootFromDSPPage4, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Boot from Page 4 - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Boot from Page 4 - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); msleep(DSP_BOOT_DELAY_IN_MS); // -------------------------------------------------------------------------------- @@ -1065,8 +1055,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Buffer Memory - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Configure Buffer Memory - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); udelay(INTERCOMMAND_DELAY); @@ -1079,8 +1070,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Misc Memory - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Configure Misc Memory - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); // -------------------------------------------------------------------------------- // Initialize the routing and volume tables, then update the card's state. @@ -1100,15 +1092,17 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ClockSourceSelector[korg1212->clkSrcRate], 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Set Clock Source Selector - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); rc = snd_korg1212_TurnOnIdleMonitor(korg1212); snd_korg1212_setCardState(korg1212, K1212_STATE_READY); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Monitor On - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "K1212_DEBUG: Set Monitor On - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); snd_korg1212_setCardState(korg1212, K1212_STATE_DSP_COMPLETE); } @@ -1133,9 +1127,10 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) switch (doorbellValue) { case K1212_DB_DSPDownloadDone: - K1212_DEBUG_PRINTK("K1212_DEBUG: IRQ DNLD count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ DNLD count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); if (korg1212->cardState == K1212_STATE_DSP_IN_PROCESS) { korg1212->dsp_is_loaded = 1; wake_up(&korg1212->wait); @@ -1146,10 +1141,11 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) // an error occurred - stop the card // ------------------------------------------------------------------------ case K1212_DB_DMAERROR: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DMAE count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); - snd_printk(KERN_ERR "korg1212: DMA Error\n"); + dev_dbg(korg1212->card->dev, + "IRQ DMAE count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); + dev_err(korg1212->card->dev, "DMA Error\n"); korg1212->errorcnt++; korg1212->totalerrorcnt++; korg1212->sharedBufferPtr->cardCommand = 0; @@ -1161,16 +1157,19 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) // the semaphore in case someone is waiting for this. // ------------------------------------------------------------------------ case K1212_DB_CARDSTOPPED: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ CSTP count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ CSTP count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); korg1212->sharedBufferPtr->cardCommand = 0; break; default: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", - korg1212->irqcount, doorbellValue, - korg1212->currentBuffer, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", + korg1212->irqcount, doorbellValue, + korg1212->currentBuffer, + stateName[korg1212->cardState]); if ((korg1212->cardState > K1212_STATE_SETUP) || korg1212->idleMonitorOn) { korg1212->currentBuffer++; @@ -1206,8 +1205,8 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is starting... [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "DSP download is starting... [%s]\n", + stateName[korg1212->cardState]); // --------------------------------------------------------------- // verify the state of the card before proceeding. @@ -1221,8 +1220,9 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) UpperWordSwap(korg1212->dma_dsp.addr), 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Start DSP Download RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Start DSP Download RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); korg1212->dsp_is_loaded = 0; wait_event_timeout(korg1212->wait, korg1212->dsp_is_loaded, HZ * CARD_BOOT_TIMEOUT); @@ -1281,20 +1281,20 @@ static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int coun struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos; int i; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n", - pos, offset, size, count); + dev_dbg(korg1212->card->dev, + "K1212_DEBUG: snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n", + pos, offset, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", - dst, i); + dev_dbg(korg1212->card->dev, + "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", + dst, i); return -EFAULT; } -#endif memset((void*) dst + offset, 0, size); dst++; } @@ -1307,22 +1307,25 @@ static int snd_korg1212_copy_to(struct snd_korg1212 *korg1212, void __user *dst, struct KorgAudioFrame * src = korg1212->recordDataBufsPtr[0].bufferData + pos; int i, rc; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_to pos=%d offset=%d size=%d\n", - pos, offset, size); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to pos=%d offset=%d size=%d\n", + pos, offset, size); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) src < (void *) korg1212->recordDataBufsPtr || (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } -#endif rc = copy_to_user(dst + offset, src, size); if (rc) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } src++; @@ -1337,23 +1340,26 @@ static int snd_korg1212_copy_from(struct snd_korg1212 *korg1212, void __user *sr struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos; int i, rc; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n", - pos, offset, size, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n", + pos, offset, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } -#endif rc = copy_from_user((void*) dst + offset, src, size); if (rc) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } dst++; @@ -1367,8 +1373,8 @@ static void snd_korg1212_free_pcm(struct snd_pcm *pcm) { struct snd_korg1212 *korg1212 = pcm->private_data; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_free_pcm [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_free_pcm [%s]\n", + stateName[korg1212->cardState]); korg1212->pcm = NULL; } @@ -1379,8 +1385,8 @@ static int snd_korg1212_playback_open(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_open [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_open [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_OpenCard(korg1212); @@ -1408,8 +1414,8 @@ static int snd_korg1212_capture_open(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_open [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_open [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_OpenCard(korg1212); @@ -1435,8 +1441,8 @@ static int snd_korg1212_playback_close(struct snd_pcm_substream *substream) unsigned long flags; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_close [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_close [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_silence(korg1212, 0, K1212_MAX_SAMPLES, 0, korg1212->channels * 2); @@ -1457,8 +1463,8 @@ static int snd_korg1212_capture_close(struct snd_pcm_substream *substream) unsigned long flags; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_close [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_close [%s]\n", + stateName[korg1212->cardState]); spin_lock_irqsave(&korg1212->lock, flags); @@ -1475,14 +1481,18 @@ static int snd_korg1212_capture_close(struct snd_pcm_substream *substream) static int snd_korg1212_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_ioctl: cmd=%d\n", cmd); + struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); + + dev_dbg(korg1212->card->dev, "snd_korg1212_ioctl: cmd=%d\n", cmd); if (cmd == SNDRV_PCM_IOCTL1_CHANNEL_INFO ) { struct snd_pcm_channel_info *info = arg; info->offset = 0; info->first = info->channel * 16; info->step = 256; - K1212_DEBUG_PRINTK("K1212_DEBUG: channel_info %d:, offset=%ld, first=%d, step=%d\n", info->channel, info->offset, info->first, info->step); + dev_dbg(korg1212->card->dev, + "channel_info %d:, offset=%ld, first=%d, step=%d\n", + info->channel, info->offset, info->first, info->step); return 0; } @@ -1498,8 +1508,8 @@ static int snd_korg1212_hw_params(struct snd_pcm_substream *substream, pid_t this_pid; pid_t other_pid; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_hw_params [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_hw_params [%s]\n", + stateName[korg1212->cardState]); spin_lock_irqsave(&korg1212->lock, flags); @@ -1546,15 +1556,16 @@ static int snd_korg1212_prepare(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_prepare [%s]\n", + stateName[korg1212->cardState]); spin_lock_irq(&korg1212->lock); /* FIXME: we should wait for ack! */ if (korg1212->stop_pending_cnt > 0) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare - Stop is pending... [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "snd_korg1212_prepare - Stop is pending... [%s]\n", + stateName[korg1212->cardState]); spin_unlock_irq(&korg1212->lock); return -EAGAIN; /* @@ -1579,8 +1590,8 @@ static int snd_korg1212_trigger(struct snd_pcm_substream *substream, struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_trigger [%s] cmd=%d\n", - stateName[korg1212->cardState], cmd); + dev_dbg(korg1212->card->dev, "snd_korg1212_trigger [%s] cmd=%d\n", + stateName[korg1212->cardState], cmd); spin_lock(&korg1212->lock); switch (cmd) { @@ -1621,8 +1632,9 @@ static snd_pcm_uframes_t snd_korg1212_playback_pointer(struct snd_pcm_substream pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_pointer [%s] %ld\n", - stateName[korg1212->cardState], pos); + dev_dbg(korg1212->card->dev, + "snd_korg1212_playback_pointer [%s] %ld\n", + stateName[korg1212->cardState], pos); return pos; } @@ -1634,8 +1646,8 @@ static snd_pcm_uframes_t snd_korg1212_capture_pointer(struct snd_pcm_substream * pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_pointer [%s] %ld\n", - stateName[korg1212->cardState], pos); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_pointer [%s] %ld\n", + stateName[korg1212->cardState], pos); return pos; } @@ -1648,8 +1660,9 @@ static int snd_korg1212_playback_copy(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_copy [%s] %ld %ld\n", - stateName[korg1212->cardState], pos, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_playback_copy [%s] %ld %ld\n", + stateName[korg1212->cardState], pos, count); return snd_korg1212_copy_from(korg1212, src, pos, count, 0, korg1212->channels * 2); @@ -1662,8 +1675,8 @@ static int snd_korg1212_playback_silence(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_silence [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_silence [%s]\n", + stateName[korg1212->cardState]); return snd_korg1212_silence(korg1212, pos, count, 0, korg1212->channels * 2); } @@ -1676,8 +1689,9 @@ static int snd_korg1212_capture_copy(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_copy [%s] %ld %ld\n", - stateName[korg1212->cardState], pos, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_capture_copy [%s] %ld %ld\n", + stateName[korg1212->cardState], pos, count); return snd_korg1212_copy_to(korg1212, dst, pos, count, 0, korg1212->channels * 2); } @@ -2138,7 +2152,6 @@ snd_korg1212_free(struct snd_korg1212 *korg1212) static int snd_korg1212_dev_free(struct snd_device *device) { struct snd_korg1212 *korg1212 = device->device_data; - K1212_DEBUG_PRINTK("K1212_DEBUG: Freeing device\n"); return snd_korg1212_free(korg1212); } @@ -2210,19 +2223,15 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, ioport_size = pci_resource_len(korg1212->pci, 1); iomem2_size = pci_resource_len(korg1212->pci, 2); - K1212_DEBUG_PRINTK("K1212_DEBUG: resources:\n" - " iomem = 0x%lx (%d)\n" - " ioport = 0x%lx (%d)\n" - " iomem = 0x%lx (%d)\n" - " [%s]\n", - korg1212->iomem, iomem_size, - korg1212->ioport, ioport_size, - korg1212->iomem2, iomem2_size, - stateName[korg1212->cardState]); + dev_dbg(card->dev, + "resources:\n\tiomem = 0x%lx (%d)\n\tioport = 0x%lx (%d)\n\tiomem = 0x%lx (%d)\n\t[%s]\n", + korg1212->iomem, iomem_size, korg1212->ioport, ioport_size, + korg1212->iomem2, iomem2_size, stateName[korg1212->cardState]); if ((korg1212->iobase = ioremap(korg1212->iomem, iomem_size)) == NULL) { - snd_printk(KERN_ERR "korg1212: unable to remap memory region 0x%lx-0x%lx\n", korg1212->iomem, - korg1212->iomem + iomem_size - 1); + dev_dbg(card->dev, + "unable to remap memory region 0x%lx-0x%lx\n", + korg1212->iomem, korg1212->iomem + iomem_size - 1); snd_korg1212_free(korg1212); return -EBUSY; } @@ -2232,7 +2241,8 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, KBUILD_MODNAME, korg1212); if (err) { - snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq); + dev_dbg(card->dev, "unable to grab IRQ %d\n", + pci->irq); snd_korg1212_free(korg1212); return -EBUSY; } @@ -2252,40 +2262,46 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->sensRegPtr = (u16 __iomem *) (korg1212->iobase + SENS_CONTROL_OFFSET); korg1212->idRegPtr = (u32 __iomem *) (korg1212->iobase + DEV_VEND_ID_OFFSET); - K1212_DEBUG_PRINTK("K1212_DEBUG: card registers:\n" - " Status register = 0x%p\n" - " OutDoorbell = 0x%p\n" - " InDoorbell = 0x%p\n" - " Mailbox0 = 0x%p\n" - " Mailbox1 = 0x%p\n" - " Mailbox2 = 0x%p\n" - " Mailbox3 = 0x%p\n" - " ControlReg = 0x%p\n" - " SensReg = 0x%p\n" - " IDReg = 0x%p\n" - " [%s]\n", - korg1212->statusRegPtr, - korg1212->outDoorbellPtr, - korg1212->inDoorbellPtr, - korg1212->mailbox0Ptr, - korg1212->mailbox1Ptr, - korg1212->mailbox2Ptr, - korg1212->mailbox3Ptr, - korg1212->controlRegPtr, - korg1212->sensRegPtr, - korg1212->idRegPtr, - stateName[korg1212->cardState]); + dev_dbg(card->dev, + "card registers:\n" + "\tStatus register = 0x%p\n" + "\tOutDoorbell = 0x%p\n" + "\tInDoorbell = 0x%p\n" + "\tMailbox0 = 0x%p\n" + "\tMailbox1 = 0x%p\n" + "\tMailbox2 = 0x%p\n" + "\tMailbox3 = 0x%p\n" + "\tControlReg = 0x%p\n" + "\tSensReg = 0x%p\n" + "\tIDReg = 0x%p\n" + "\t[%s]\n", + korg1212->statusRegPtr, + korg1212->outDoorbellPtr, + korg1212->inDoorbellPtr, + korg1212->mailbox0Ptr, + korg1212->mailbox1Ptr, + korg1212->mailbox2Ptr, + korg1212->mailbox3Ptr, + korg1212->controlRegPtr, + korg1212->sensRegPtr, + korg1212->idRegPtr, + stateName[korg1212->cardState]); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), sizeof(struct KorgSharedBuffer), &korg1212->dma_shared) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%Zd bytes)\n", sizeof(struct KorgSharedBuffer)); + dev_err(card->dev, + "can not allocate shared buffer memory (%Zd bytes)\n", + sizeof(struct KorgSharedBuffer)); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->sharedBufferPtr = (struct KorgSharedBuffer *)korg1212->dma_shared.area; korg1212->sharedBufferPhy = korg1212->dma_shared.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Shared Buffer Area = 0x%p (0x%08lx), %d bytes\n", korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, sizeof(struct KorgSharedBuffer)); + dev_dbg(card->dev, + "Shared Buffer Area = 0x%p (0x%08lx), %d bytes\n", + korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, + sizeof(struct KorgSharedBuffer)); #ifndef K1212_LARGEALLOC @@ -2293,27 +2309,35 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), korg1212->DataBufsSize, &korg1212->dma_play) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate play data buffer memory (%d bytes)\n", korg1212->DataBufsSize); + dev_err(card->dev, + "can not allocate play data buffer memory (%d bytes)\n", + korg1212->DataBufsSize); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->playDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_play.area; korg1212->PlayDataPhy = korg1212->dma_play.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Play Data Area = 0x%p (0x%08x), %d bytes\n", - korg1212->playDataBufsPtr, korg1212->PlayDataPhy, korg1212->DataBufsSize); + dev_dbg(card->dev, + "Play Data Area = 0x%p (0x%08x), %d bytes\n", + korg1212->playDataBufsPtr, korg1212->PlayDataPhy, + korg1212->DataBufsSize); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), korg1212->DataBufsSize, &korg1212->dma_rec) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate record data buffer memory (%d bytes)\n", korg1212->DataBufsSize); + dev_err(card->dev, + "can not allocate record data buffer memory (%d bytes)\n", + korg1212->DataBufsSize); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->recordDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_rec.area; korg1212->RecDataPhy = korg1212->dma_rec.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Record Data Area = 0x%p (0x%08x), %d bytes\n", - korg1212->recordDataBufsPtr, korg1212->RecDataPhy, korg1212->DataBufsSize); + dev_dbg(card->dev, + "Record Data Area = 0x%p (0x%08x), %d bytes\n", + korg1212->recordDataBufsPtr, korg1212->RecDataPhy, + korg1212->DataBufsSize); #else // K1212_LARGEALLOC @@ -2334,22 +2358,25 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, err = request_firmware(&dsp_code, "korg/k1212.dsp", &pci->dev); if (err < 0) { release_firmware(dsp_code); - snd_printk(KERN_ERR "firmware not available\n"); + dev_err(card->dev, "firmware not available\n"); snd_korg1212_free(korg1212); return err; } if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), dsp_code->size, &korg1212->dma_dsp) < 0) { - snd_printk(KERN_ERR "korg1212: cannot allocate dsp code memory (%zd bytes)\n", dsp_code->size); + dev_err(card->dev, + "cannot allocate dsp code memory (%zd bytes)\n", + dsp_code->size); snd_korg1212_free(korg1212); release_firmware(dsp_code); return -ENOMEM; } - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", - korg1212->dma_dsp.area, korg1212->dma_dsp.addr, dsp_code->size, - stateName[korg1212->cardState]); + dev_dbg(card->dev, + "DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", + korg1212->dma_dsp.area, korg1212->dma_dsp.addr, dsp_code->size, + stateName[korg1212->cardState]); memcpy(korg1212->dma_dsp.area, dsp_code->data, dsp_code->size); @@ -2358,7 +2385,8 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_RebootCard, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Reboot Card - RC = %d [%s]\n", rc, stateName[korg1212->cardState]); + dev_dbg(card->dev, "Reboot Card - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, korg1212, &ops)) < 0) { snd_korg1212_free(korg1212); @@ -2372,18 +2400,22 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, if (snd_korg1212_downloadDSPCode(korg1212)) return -EBUSY; - K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], " - "PlayDataPhy = %08x L[%08x]\n" - "korg1212: RecDataPhy = %08x L[%08x], " - "VolumeTablePhy = %08x L[%08x]\n" - "korg1212: RoutingTablePhy = %08x L[%08x], " - "AdatTimeCodePhy = %08x L[%08x]\n", - (int)korg1212->dma_dsp.addr, UpperWordSwap(korg1212->dma_dsp.addr), - korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy), - korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy), - korg1212->VolumeTablePhy, LowerWordSwap(korg1212->VolumeTablePhy), - korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy), - korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy)); + dev_dbg(card->dev, + "dspMemPhy = %08x U[%08x], PlayDataPhy = %08x L[%08x]\n", + (int)korg1212->dma_dsp.addr, + UpperWordSwap(korg1212->dma_dsp.addr), + korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy)); + dev_dbg(card->dev, + "RecDataPhy = %08x L[%08x], VolumeTablePhy = %08x L[%08x]\n", + korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy), + korg1212->VolumeTablePhy, + LowerWordSwap(korg1212->VolumeTablePhy)); + dev_dbg(card->dev, + "RoutingTablePhy = %08x L[%08x], AdatTimeCodePhy = %08x L[%08x]\n", + korg1212->RoutingTablePhy, + LowerWordSwap(korg1212->RoutingTablePhy), + korg1212->AdatTimeCodePhy, + LowerWordSwap(korg1212->AdatTimeCodePhy)); if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0) return err; @@ -2446,7 +2478,7 @@ snd_korg1212_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, korg1212->iomem, korg1212->irq); - K1212_DEBUG_PRINTK("K1212_DEBUG: %s\n", card->longname); + dev_dbg(card->dev, "%s\n", card->longname); if ((err = snd_card_register(card)) < 0) { snd_card_free(card); -- 1.8.1.2

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[alsa-devel] [PATCH v3 0/6] ASoC: Intel: ACPI support and machines
by Vinod Koul 10 Nov '14

10 Nov '14

This is third rev of patch series which fixes the modules split as discussed in ML. The core part is now module along with PCI and ACPI one. Added fix reported by Dan as well. Also added the BYTCR machine driver and MERR machine driver Updates in v3: - Fix export fo few more symbols - fix cleanup of probe and setup code - add MERR machine code Subhransu S. Prusty (1): ASoC: Intel: add BYTCR machine driver with RT5640 Vinod Koul (5): ASoC: Intel: mrfld - remove unnecessary check for pointer ASoC: Intel: mrfld - create separate module for pci part ASoC: Intel: mrfld - add shim save restore ASoC: Intel: mrfld- add ACPI module ASoC: Intel: Add Merrifield machine driver sound/soc/intel/Kconfig | 37 +++- sound/soc/intel/Makefile | 4 + sound/soc/intel/bytcr_dpcm_rt5640.c | 258 ++++++++++++++++++++++ sound/soc/intel/merr_dpcm_wm8958.c | 416 +++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.c | 2 +- sound/soc/intel/sst/Makefile | 8 +- sound/soc/intel/sst/sst.c | 254 +++++---------------- sound/soc/intel/sst/sst.h | 9 +- sound/soc/intel/sst/sst_acpi.c | 377 +++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pci.c | 209 ++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 3 + 11 files changed, 1378 insertions(+), 199 deletions(-) create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c create mode 100644 sound/soc/intel/merr_dpcm_wm8958.c create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_pci.c

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[alsa-devel] [PATCH 0/3 v4] Digidesign Mbox 1 enhancements
by Damien Zammit 10 Nov '14

10 Nov '14

This is fourth revision based on all reviews/tweaks so far. Damien Zammit (3): ALSA: usb-audio: Move mutex define to more general location ALSA: usb-audio: Add mixer control for Digidesign mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/card.c | 1 - sound/usb/mixer_maps.c | 9 +++ sound/usb/mixer_quirks.c | 144 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 ++++++++++---- sound/usb/usbaudio.h | 2 + 5 files changed, 188 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v4 0/4] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 10 Nov '14

10 Nov '14

This patch set adds multi-CODEC support to the simple card. v4: (Mark Brown) - document the new core function - calculation of the number of codecs moved to the core function - split the patch about simple-card.c - less code changes - remove the cast of pointers - less changes in the DT binding example 2 v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (4): ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: simple-card: add multi-CODECs in DT ASoC: simple-card: Remove useless check .../devicetree/bindings/sound/simple-card.txt | 20 ++-- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 87 ++++++++-------- sound/soc/soc-core.c | 113 ++++++++++++++++++--- 4 files changed, 154 insertions(+), 69 deletions(-) -- 2.1.3

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[alsa-devel] [PATCH 0/2 v3] Digidesign Mbox 1 enhancements
by Damien Zammit 10 Nov '14

10 Nov '14

This is the third patchset to provide mixer control and duplex mode. Was rebased on top of 85a8181329a and tested on the hardware. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 +++ sound/usb/mixer_quirks.c | 142 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 ++++++++++---- 3 files changed, 184 insertions(+), 10 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/2] Digidesign Mbox 1 duplex mode and mixer
by Damien Zammit 10 Nov '14

10 Nov '14

Takashi asked me to rewrite the patches and send them without attachments. I hope this email series works and they can be reviewed. These two patches provide the following for Digidesign Mbox 1: o Mixer control for clock source selection: Internal/SPDIF o Duplex mode (playback and capture) Damien Zammit (2): snd-usb-audio: Add mixer control for Digidesign Mbox 1 clock source snd-usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/card.h | 5 ++ sound/usb/mixer_maps.c | 9 +++ sound/usb/mixer_quirks.c | 177 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 73 ++++++++++++------- sound/usb/quirks.c | 30 ++++++++ sound/usb/usbaudio.h | 1 + 6 files changed, 271 insertions(+), 24 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 0/2] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 09 Nov '14

09 Nov '14

This patch set adds multi-CODEC support to the simple card. v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (2): ASoC: core: add multi-codec support in DT ASoC: simple-card: add multi-CODECs in DT .../devicetree/bindings/sound/simple-card.txt | 28 ++--- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 135 +++++++++++++-------- sound/soc/soc-core.c | 75 +++++++++--- 4 files changed, 157 insertions(+), 84 deletions(-) -- 2.1.1

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