January 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] soc: intel: sst-haswell-ipc: Remove unused functions
by Rickard Strandqvist 07 Feb '15

07 Feb '15

Removes some functions that are not used anywhere: sst_hsw_stream_unmute() sst_hsw_stream_mute() msg_set_stage_type() sst_hsw_dx_get_state() sst_hsw_stream_set_write_position() sst_hsw_stream_get_vol_reg() sst_hsw_stream_get_peak_reg() sst_hsw_stream_get_pointer_reg() sst_hsw_stream_get_read_reg() sst_hsw_stream_get_mixer_id() sst_hsw_stream_get_hw_id() sst_hsw_mixer_set_volume_curve() sst_hsw_mixer_unmute() sst_hsw_mixer_mute() sst_hsw_stream_set_volume_curve() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/intel/sst-haswell-ipc.c | 168 ------------------------------------- sound/soc/intel/sst-haswell-ipc.h | 31 ------- 2 files changed, 199 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c482..d693d4a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -337,12 +337,6 @@ static inline u32 msg_get_stage_type(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } -static inline u32 msg_set_stage_type(u32 msg, u32 type) -{ - return (msg & ~IPC_STG_TYPE_MASK) + - (type << IPC_STG_TYPE_SHIFT); -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -969,45 +963,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, } /* Mixer Controls */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, - &stream->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - stream->mute[channel] = 1; - return 0; -} - -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) - -{ - int ret; - - stream->mute[channel] = 0; - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, - stream->mute_volume[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - return 0; -} - int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume) { @@ -1021,17 +976,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return 0; } -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - stream->vol_req.curve_duration = curve_duration; - stream->vol_req.curve_type = curve; - - return 0; -} - /* stream volume */ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) @@ -1083,42 +1027,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, return 0; } -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, - &hsw->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 1; - return 0; -} - -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, - hsw->mixer_info.volume_register_address[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 0; - return 0; -} - int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume) { @@ -1132,16 +1040,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, return 0; } -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - hsw->curve_duration = curve_duration; - hsw->curve_type = curve; - - return 0; -} - /* global mixer volume */ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume) @@ -1434,48 +1332,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.stream_hw_id; -} - -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.mixer_hw_id; -} - -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.read_position_register_address; -} - -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.presentation_position_register_address; -} - -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.peak_meter_register_address[channel]; -} - -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.volume_register_address[channel]; -} - int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; @@ -1598,30 +1454,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, return ppos; } -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position) -{ - u32 header; - int ret; - - trace_stream_write_position(stream->reply.stream_hw_id, position); - - 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); - header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); - header |= (stage_id << IPC_STG_ID_SHIFT); - stream->wpos.position = position; - - ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, - sizeof(stream->wpos)); - if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", - stream->reply.stream_hw_id, position); - - return ret; -} - /* physical BE config */ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 138e894..c1ad901 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, u32 create_channel_map(enum sst_hsw_channel_config config); /* Stream Mixer Controls - */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); - int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve); - /* Global Mixer Controls - */ -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); - int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume); int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve); - /* Stream API */ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), @@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ @@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position); u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw, struct sst_hsw_stream *stream); u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, @@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, /* DX Config */ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source); /* init */ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: Intel:Add Cherrytrail & Braswell machine driver cht_bsw_rt5645
by yang.a.fang＠intel.com 05 Feb '15

05 Feb '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5645 codec Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/Kconfig | 11 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/cht_bsw_rt5645.c | 317 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 330 insertions(+) create mode 100644 sound/soc/intel/cht_bsw_rt5645.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f06fcf1..12093fd 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -110,3 +110,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH platforms with RT5672 audio codec. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5645_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5645 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5645 audio codec. + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index e928ec3..a8e53c4 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -28,6 +28,7 @@ snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o 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 @@ -35,6 +36,7 @@ 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-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c new file mode 100644 index 0000000..7a6f1e5 --- /dev/null +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -0,0 +1,317 @@ +/* + * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5645 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Fang, Yang A <yang.a.fang(a)intel.com> + * N,Harshapriya <harshapriya.n(a)intel.com> + * This file is modified from cht_bsw_rt5672.c + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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 <sound/jack.h> +#include "../codecs/rt5645.h" +#include "sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "rt5645-aif1" + +struct cht_mc_private { + struct snd_soc_jack hp_jack; + struct snd_soc_jack mic_jack; +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + if (!SND_SOC_DAPM_EVENT_OFF(event)) + return 0; + + /* Set codec sysclk source to its internal clock because codec PLL will + * be off when idle and MCLK will also be off by ACPI when codec is + * runtime suspended. Codec needs clock for jack detection and button + * press. + */ + snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK, + 0, SND_SOC_CLOCK_IN); + + return 0; +} + +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), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), +}; + +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, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, + {"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"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + +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; + + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_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, RT5645_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_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + + /* Select clk_i2s1_asrc as ASRC clock source */ + snd_soc_write(codec, RT5645_ASRC_2, 0x1111); + + /* 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, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Headphone Jack", + SND_JACK_HEADPHONE, + &ctx->hp_jack); + if (ret) { + dev_err(runtime->dev, "Headphone jack creation failed\n"); + return ret; + } + + ret = snd_soc_jack_new(codec, "Mic Jack", + SND_JACK_MICROPHONE, + &ctx->mic_jack); + if (ret) { + dev_err(runtime->dev, "Mic jack creation failed\n"); + return ret; + } + + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); + + return ret; +} + +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 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[] = { + [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", + .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", + }, + /* CODEC<->CODEC link */ + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5645:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .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 = "chtrt5645", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .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; + struct cht_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &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 struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5645", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); +MODULE_AUTHOR("Fang, Yang A,N,Harshapriya"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5645"); -- 1.7.9.5

3 13

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add support for tlv320aic3104
by Jyri Sarha 04 Feb '15

04 Feb '15

Disables GPIO support and LINE2 input and renames Mic3 input to Mic2, if tlv320aic3104 mode is seleced. Devicetree binding document is updated accordingly. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- .../devicetree/bindings/sound/tlv320aic3x.txt | 10 +- sound/soc/codecs/tlv320aic3x.c | 344 ++++++++++++++------ 2 files changed, 252 insertions(+), 102 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 5e6040c..47a213c 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -9,6 +9,7 @@ Required properties: "ti,tlv320aic33" - TLV320AIC33 "ti,tlv320aic3007" - TLV320AIC3007 "ti,tlv320aic3106" - TLV320AIC3106 + "ti,tlv320aic3104" - TLV320AIC3104 - reg - <int> - I2C slave address @@ -18,6 +19,7 @@ Optional properties: - gpio-reset - gpio pin number used for codec reset - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + - Not supported on tlv320aic3104 - ai3x-micbias-vg - MicBias Voltage required. 1 - MICBIAS output is powered to 2.0V, 2 - MICBIAS output is powered to 2.5V, @@ -36,7 +38,13 @@ CODEC output pins: * HPLCOM * HPRCOM -CODEC input pins: +CODEC input pins for TLV320AIC3104: + * MIC2L + * MIC2R + * LINE1L + * LINE1R + +CODEC input pins for other compatible codecs: * MIC3L * MIC3R * LINE1L diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b7ebce0..49ba3c7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -87,6 +87,7 @@ struct aic3x_priv { #define AIC3X_MODEL_3X 0 #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 +#define AIC3X_MODEL_3104 3 u16 model; /* Selects the micbias voltage */ @@ -316,52 +317,37 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * only for swapped L-to-R and R-to-L routes. See below stereo controls * for direct L-to-L and R-to-R routes. */ - SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", - LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left Line Mixer PGAR Bypass Volume", PGAR_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left Line Mixer DACR1 Playback Volume", DACR1_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", - LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right Line Mixer PGAL Bypass Volume", PGAL_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right Line Mixer DACL1 Playback Volume", DACL1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", - LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HP Mixer PGAR Bypass Volume", PGAR_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HP Mixer DACR1 Playback Volume", DACR1_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", - LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HP Mixer PGAL Bypass Volume", PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HP Mixer DACL1 Playback Volume", DACL1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", - LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HPCOM Mixer PGAR Bypass Volume", PGAR_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HPCOM Mixer DACR1 Playback Volume", DACR1_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", - LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HPCOM Mixer PGAL Bypass Volume", PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HPCOM Mixer DACL1 Playback Volume", DACL1_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), /* Stereo output controls for direct L-to-L and R-to-R routes */ - SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", - LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("Line PGA Bypass Volume", PGAL_2_LLOPM_VOL, PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), @@ -369,9 +355,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", - LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("HP PGA Bypass Volume", PGAL_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -379,9 +362,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", - LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("HPCOM PGA Bypass Volume", PGAL_2_HPLCOM_VOL, PGAR_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), @@ -424,6 +404,45 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), }; +/* For other than tlv320aic3104 */ +static const struct snd_kcontrol_new aic3x_extra_snd_controls[] = { + /* + * Output controls that map to output mixer switches. Note these are + * only for swapped L-to-R and R-to-L routes. See below stereo controls + * for direct L-to-L and R-to-R routes. + */ + SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", + LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", + LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", + LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", + LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", + LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", + LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + + /* Stereo output controls for direct L-to-L and R-to-R routes */ + SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", + LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", + LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", + LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), +}; + static const struct snd_kcontrol_new aic3x_mono_controls[] = { SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, @@ -464,22 +483,24 @@ SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum); /* Left Line Mixer */ static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_LLOPM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), }; /* Right Line Mixer */ static const struct snd_kcontrol_new aic3x_right_line_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_RLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), }; /* Mono Mixer */ @@ -494,42 +515,46 @@ static const struct snd_kcontrol_new aic3x_mono_mixer_controls[] = { /* Left HP Mixer */ static const struct snd_kcontrol_new aic3x_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLOUT_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), }; /* Right HP Mixer */ static const struct snd_kcontrol_new aic3x_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), }; /* Left HPCOM Mixer */ static const struct snd_kcontrol_new aic3x_left_hpcom_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLCOM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), }; /* Right HPCOM Mixer */ static const struct snd_kcontrol_new aic3x_right_hpcom_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPRCOM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; /* Left PGA Mixer */ @@ -550,6 +575,22 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), }; +/* Left PGA Mixer for tlv320aic3104 */ +static const struct snd_kcontrol_new aic3104_left_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1), +}; + +/* Right PGA Mixer for tlv320aic3104 */ +static const struct snd_kcontrol_new aic3104_right_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), +}; + /* Left Line1 Mux */ static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = SOC_DAPM_ENUM("Route", aic3x_line1l_2_l_enum); @@ -593,26 +634,56 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), - SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_left_pga_mixer_controls[0], - ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1l_mux_controls), SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1r_mux_controls), - SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, - &aic3x_left_line2_mux_controls), /* Inputs to Right ADC */ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", LINE1R_2_RADC_CTRL, 2, 0), - SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_right_pga_mixer_controls[0], - ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1l_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1r_mux_controls), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("LLOUT"), + SND_SOC_DAPM_OUTPUT("RLOUT"), + SND_SOC_DAPM_OUTPUT("HPLOUT"), + SND_SOC_DAPM_OUTPUT("HPROUT"), + SND_SOC_DAPM_OUTPUT("HPLCOM"), + SND_SOC_DAPM_OUTPUT("HPRCOM"), + + SND_SOC_DAPM_INPUT("LINE1L"), + SND_SOC_DAPM_INPUT("LINE1R"), + + /* + * Virtual output pin to detection block inside codec. This can be + * used to keep codec bias on if gpio or detection features are needed. + * Force pin on or construct a path with an input jack and mic bias + * widgets. + */ + SND_SOC_DAPM_OUTPUT("Detection"), +}; + +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_widget aic3x_extra_dapm_widgets[] = { + /* Inputs to Left ADC */ + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_left_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line2_mux_controls), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), @@ -637,11 +708,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), - /* Mic Bias */ - SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_line_mixer_controls[0], @@ -662,27 +728,46 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_right_hpcom_mixer_controls[0], ARRAY_SIZE(aic3x_right_hpcom_mixer_controls)), - SND_SOC_DAPM_OUTPUT("LLOUT"), - SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("HPLOUT"), - SND_SOC_DAPM_OUTPUT("HPROUT"), - SND_SOC_DAPM_OUTPUT("HPLCOM"), - SND_SOC_DAPM_OUTPUT("HPRCOM"), - SND_SOC_DAPM_INPUT("MIC3L"), SND_SOC_DAPM_INPUT("MIC3R"), - SND_SOC_DAPM_INPUT("LINE1L"), - SND_SOC_DAPM_INPUT("LINE1R"), SND_SOC_DAPM_INPUT("LINE2L"), SND_SOC_DAPM_INPUT("LINE2R"), +}; - /* - * Virtual output pin to detection block inside codec. This can be - * used to keep codec bias on if gpio or detection features are needed. - * Force pin on or construct a path with an input jack and mic bias - * widgets. - */ - SND_SOC_DAPM_OUTPUT("Detection"), +/* For tlv320aic3104 */ +static const struct snd_soc_dapm_widget aic3104_extra_dapm_widgets[] = { + /* Inputs to Left ADC */ + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3104_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3104_left_pga_mixer_controls)), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3104_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3104_right_pga_mixer_controls)), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_line_mixer_controls[0], + ARRAY_SIZE(aic3x_left_line_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_line_mixer_controls[0], + ARRAY_SIZE(aic3x_right_line_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hp_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hp_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Left HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hpcom_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hpcom_mixer_controls) - 2), + + SND_SOC_DAPM_INPUT("MIC2L"), + SND_SOC_DAPM_INPUT("MIC2R"), }; static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { @@ -712,17 +797,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line1R Mux", "single-ended", "LINE1R"}, {"Left Line1R Mux", "differential", "LINE1R"}, - {"Left Line2L Mux", "single-ended", "LINE2L"}, - {"Left Line2L Mux", "differential", "LINE2L"}, - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, - {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, - {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, - {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Left ADC", NULL, "Left PGA Mixer"}, - {"Left ADC", NULL, "GPIO1 dmic modclk"}, /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, @@ -730,25 +808,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line1L Mux", "single-ended", "LINE1L"}, {"Right Line1L Mux", "differential", "LINE1L"}, - {"Right Line2R Mux", "single-ended", "LINE2R"}, - {"Right Line2R Mux", "differential", "LINE2R"}, - {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"}, {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, - {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, - {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, - {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, - {"Right ADC", NULL, "GPIO1 dmic modclk"}, - - /* - * Logical path between digital mic enable and GPIO1 modulator clock - * output function - */ - {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, - {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, - {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, /* Left DAC Output */ {"Left DAC Mux", "DAC_L1", "Left DAC"}, @@ -761,10 +824,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right DAC Mux", "DAC_R3", "Right DAC"}, /* Left Line Output */ - {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left Line Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left Line Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -773,10 +834,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"LLOUT", NULL, "Left Line Out"}, /* Right Line Output */ - {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right Line Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right Line Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -785,10 +844,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"RLOUT", NULL, "Right Line Out"}, /* Left HP Output */ - {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left HP Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left HP Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -797,10 +854,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPLOUT", NULL, "Left HP Out"}, /* Right HP Output */ - {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right HP Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right HP Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -809,10 +864,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPROUT", NULL, "Right HP Out"}, /* Left HPCOM Output */ - {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -823,10 +876,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPLCOM", NULL, "Left HP Com"}, /* Right HPCOM Output */ - {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -839,6 +890,72 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_route intercon_extra[] = { + /* Left Input */ + {"Left Line2L Mux", "single-ended", "LINE2L"}, + {"Left Line2L Mux", "differential", "LINE2L"}, + + {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, + {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Left ADC", NULL, "GPIO1 dmic modclk"}, + + /* Right Input */ + {"Right Line2R Mux", "single-ended", "LINE2R"}, + {"Right Line2R Mux", "differential", "LINE2R"}, + + {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Right ADC", NULL, "GPIO1 dmic modclk"}, + + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, + + /* Left Line Output */ + {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right Line Output */ + {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Left HP Output */ + {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right HP Output */ + {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Left HPCOM Output */ + {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right HPCOM Output */ + {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, +}; + +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_route intercon_extra_3104[] = { + /* Left Input */ + {"Left PGA Mixer", "Mic2L Switch", "MIC2L"}, + {"Left PGA Mixer", "Mic2R Switch", "MIC2R"}, + + /* Right Input */ + {"Right PGA Mixer", "Mic2L Switch", "MIC2L"}, + {"Right PGA Mixer", "Mic2R Switch", "MIC2R"}, +}; + static const struct snd_soc_dapm_route intercon_mono[] = { /* Mono Output */ {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, @@ -867,17 +984,31 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, + ARRAY_SIZE(aic3x_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra, + ARRAY_SIZE(intercon_extra)); snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, ARRAY_SIZE(aic3x_dapm_mono_widgets)); snd_soc_dapm_add_routes(dapm, intercon_mono, ARRAY_SIZE(intercon_mono)); break; case AIC3X_MODEL_3007: + snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, + ARRAY_SIZE(aic3x_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra, + ARRAY_SIZE(intercon_extra)); snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); break; + case AIC3X_MODEL_3104: + snd_soc_dapm_new_controls(dapm, aic3104_extra_dapm_widgets, + ARRAY_SIZE(aic3104_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra_3104, + ARRAY_SIZE(intercon_extra_3104)); + break; } return 0; @@ -1438,23 +1569,33 @@ static int aic3x_probe(struct snd_soc_codec *codec) aic3x_init(codec); if (aic3x->setup) { - /* setup GPIO functions */ - snd_soc_write(codec, AIC3X_GPIO1_REG, - (aic3x->setup->gpio_func[0] & 0xf) << 4); - snd_soc_write(codec, AIC3X_GPIO2_REG, - (aic3x->setup->gpio_func[1] & 0xf) << 4); + if (aic3x->model != AIC3X_MODEL_3104) { + /* setup GPIO functions */ + snd_soc_write(codec, AIC3X_GPIO1_REG, + (aic3x->setup->gpio_func[0] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO2_REG, + (aic3x->setup->gpio_func[1] & 0xf) << 4); + } else { + dev_warn(codec->dev, "GPIO functionality is not supported on tlv320aic3104\n"); + } } switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + snd_soc_add_codec_controls(codec, aic3x_extra_snd_controls, + ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_codec_controls(codec, aic3x_mono_controls, ARRAY_SIZE(aic3x_mono_controls)); break; case AIC3X_MODEL_3007: + snd_soc_add_codec_controls(codec, aic3x_extra_snd_controls, + ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); break; + case AIC3X_MODEL_3104: + break; } /* set mic bias voltage */ @@ -1522,6 +1663,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, { "tlv320aic3106", AIC3X_MODEL_3X }, + { "tlv320aic3104", AIC3X_MODEL_3104 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); -- 1.7.9.5

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[alsa-devel] [PATCH V3 00/15] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 03 Feb '15

03 Feb '15

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 as reported by Daniel Drake [1], by exposing the CDCLK gate clock (and the two other clocks) through clk API. The upside is we can switch Odroid X2/U3 to the simple-card, once the CDCLK clock is taken care of by the clk core and DT. Changes since v2: - skipped the first, already merged patch, - modified description of the patch moving clk_prepare_enable() from DAI to the platform device probe(), - the last patch marked for stable. The patch series has been created on top of branch: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/samsung Changes since the first version: - removed check for the i2s_opclk1 mux input clock while creating the mux and div clocks, - the DT binding documentation changes reworked (addressing review comments), - added include/dt-bindings/sound/samsung-i2s.h header file defining the clk indices, it's been put into a separate patch together with the I2S DT binding documentation updates to make merging of the ASoC and the dts patches separately easier, - a patch fixing compatible strings of I2S1, I2S2 in exynos4.dtsi is included in this series. This whole series may need more testing on other SoCs, so far I only tested it on Odroid Exynos4412 X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (15): ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++ arch/arm/boot/dts/exynos4.dtsi | 13 +- arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 27 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- include/dt-bindings/sound/samsung-i2s.h | 8 + sound/soc/samsung/i2s.c | 361 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 8 files changed, 295 insertions(+), 158 deletions(-) create mode 100644 include/dt-bindings/sound/samsung-i2s.h -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: sgtl5000: add delay before first I2C access
by Eric Nelson 03 Feb '15

03 Feb '15

To quote from section 1.3.1 of the data sheet: The SGTL5000 has an internal reset that is deasserted 8 SYS_MCLK cycles after all power rails have been brought up. After this time, communication can start ... 1.0μs represents 8 SYS_MCLK cycles at the minimum 8.0 MHz SYS_MCLK. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 29cf7ce..7d65a5b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1462,6 +1462,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (ret) return ret; + /* Need 8 clocks before I2C accesses */ + usleep_range(1, 2); + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) -- 1.9.1

4 7

[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table
by Cyrus Harmon 03 Feb '15

03 Feb '15

Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

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[alsa-devel] [PATCH v4 0/3] Sound support for Exynos4412 Trats2 board
by Inha Song 03 Feb '15

03 Feb '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Changes in v4: - Move to clock out parent settings in exynos4412-trats2.dts - Remove unnecessary lines in Trats2 machine driver. - Change name of dai stream to more cleary. - Add MFD_WM8994 to select in machine driver kconfig. Changes in v3: - Set clock out parent to XUSBXTI in pmu_system_controller DT node. Changes in v2: - Remove unnecessary lines in Trats2 machine driver. - move clock enable/disable to .startup / .shutdown. - Fix code style issues. - Remove mclk DT property in Trats2 sound and add MCLK1 property in WM1811 node. (MCLK1 is optional property in WM8994 node - refer: Document/devicetree/bindings/sound/wm8994.txt) - Use mclk that defined in WM1811 DT node. - Add "samsung,audio-codec" property to required properties. Inha Song (3): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 25 +++ arch/arm/boot/dts/exynos4412-trats2.dts | 42 ++++ sound/soc/samsung/Kconfig | 9 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 211 +++++++++++++++++++++ 5 files changed, 289 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH v2] ASoC: Add sysfs entries via static attribute groups
by Takashi Iwai 02 Feb '15

02 Feb '15

Instead of calling device_create_file() manually, assign the static attribute group entries at the device registration. This simplifies the error handling and avoids the possible races. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Enable attributes conditionally via is_visible callback include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 58 ++++++++++++++++++++++++++++-------------------- sound/soc/soc-dapm.c | 14 ++++-------- 3 files changed, 39 insertions(+), 35 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 89823cfe6f04..e8e0220404d4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -405,7 +405,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_update *update); /* dapm sys fs - used by the core */ -int snd_soc_dapm_sys_add(struct device *dev); +extern struct attribute *soc_dapm_dev_attrs[]; void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c62620abca6..4bd4d77d4ba8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -191,6 +191,39 @@ static ssize_t pmdown_time_set(struct device *dev, static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); +static struct attribute *soc_dev_attrs[] = { + &dev_attr_codec_reg.attr, + &dev_attr_pmdown_time.attr, + NULL +}; + +static umode_t soc_dev_attr_is_visible(struct kobject *kobj, + struct attribute *attr, int idx) +{ + struct device *dev = kobj_to_dev(kobj); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + + if (attr == &dev_attr_pmdown_time.attr) + return attr->mode; /* always visible */ + return rtd->codec ? attr->mode : 0; /* enabled only with codec */ +} + +static const struct attribute_group soc_dapm_dev_group = { + .attrs = soc_dapm_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group soc_dev_roup = { + .attrs = soc_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group *soc_dev_attr_groups[] = { + &soc_dapm_dev_group, + &soc_dev_roup, + NULL +}; + #ifdef CONFIG_DEBUG_FS static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) @@ -949,8 +982,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_pmdown_time); - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1120,6 +1151,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; + rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); @@ -1136,23 +1168,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return ret; } rtd->dev_registered = 1; - - if (rtd->codec) { - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", - ret); - - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", - ret); - } - return 0; } @@ -1308,11 +1323,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } #endif - ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); - if (ret < 0) - dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", - ret); - if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ ret = soc_new_compress(rtd, num); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..6c5cf7d1ff54 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2127,15 +2127,10 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -int snd_soc_dapm_sys_add(struct device *dev) -{ - return device_create_file(dev, &dev_attr_dapm_widget); -} - -static void snd_soc_dapm_sys_remove(struct device *dev) -{ - device_remove_file(dev, &dev_attr_dapm_widget); -} +struct attribute *soc_dapm_dev_attrs[] = { + &dev_attr_dapm_widget.attr, + NULL +}; static void dapm_free_path(struct snd_soc_dapm_path *path) { @@ -3903,7 +3898,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) */ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(dapm->dev); dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); -- 2.2.2

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[alsa-devel] [PATCH] ASoC: rx51: Correct DAPM route for capture
by Peter Ujfalusi 02 Feb '15

02 Feb '15

"Mic Bias" is DAPM_SUPPLY so it has to be connected in the route accordingly. Fixes audio capture on the board. Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Tested-by: Pavel Machek <pavel(a)ucw.cz> --- sound/soc/omap/rx51.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 04896d6252a2..7f299357c2d2 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -250,14 +250,14 @@ static const struct snd_soc_dapm_route audio_map[] = { {"FM Transmitter", NULL, "LLOUT"}, {"FM Transmitter", NULL, "RLOUT"}, - {"DMic Rate 64", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "DMic"}, + {"DMic Rate 64", NULL, "DMic"}, + {"DMic", NULL, "Mic Bias"}, {"b LINE2R", NULL, "MONO_LOUT"}, {"Earphone", NULL, "b HPLOUT"}, - {"LINE1L", NULL, "b Mic Bias"}, - {"b Mic Bias", NULL, "HS Mic"} + {"LINE1L", NULL, "HS Mic"}, + {"HS Mic", NULL, "b Mic Bias"}, }; static const char * const spk_function[] = {"Off", "On"}; -- 2.2.2

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[alsa-devel] [PATCH v4 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 02 Feb '15

02 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 67 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 36 +++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 58 ++++++++- sound/pci/hda/hda_controller.c | 43 +++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 478 insertions(+), 59 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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