From 6c49a986957bc5fe39b29166cb7fad573dc242ba Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 19 Jun 2014 09:44:26 +0300 Subject: ASoC: max98090: Remove needless defines and line feeds Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7a8e89..3aec3ae78fe0 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -26,10 +26,6 @@ #include #include "max98090.h" -#define DEBUG -#define EXTMIC_METHOD -#define EXTMIC_METHOD_TEST - /* Allows for sparsely populated register maps */ static struct reg_default max98090_reg[] = { { 0x00, 0x00 }, /* 00 Software Reset */ @@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, else val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; - if (val >= 1) { if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { max98090->pa1en = val - 1; /* Update for volatile */ @@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("DMICL"), @@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("DMIC3"), SND_SOC_DAPM_INPUT("DMIC4"), @@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { - {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, @@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"SPKR", NULL, "SPK Right Out"}, {"RCVL", NULL, "RCV Left Out"}, {"RCVR", NULL, "RCV Right Out"}, - }; static const struct snd_soc_dapm_route max98091_dapm_routes[] = { - /* DMIC inputs */ {"DMIC3", NULL, "DMIC3_ENA"}, {"DMIC4", NULL, "DMIC4_ENA"}, {"DMIC3", NULL, "AHPF"}, {"DMIC4", NULL, "AHPF"}, - }; static int max98090_add_widgets(struct snd_soc_codec *codec) @@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, ARRAY_SIZE(max98091_dapm_routes)); - } return 0; @@ -2221,7 +2209,6 @@ static void max98090_handle_pdata(struct snd_soc_codec *codec) dev_err(codec->dev, "No platform data\n"); return; } - } static int max98090_probe(struct snd_soc_codec *codec) -- cgit v1.2.3-55-g7522 From a28d167fbbef1f31d79ad3ad65a59ea6fa4d1b1f Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Jun 2014 09:57:45 +0530 Subject: ASoC: mc13783: Add missing of_node_put of_get_child_by_name expects of_node_put be called when done. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } -- cgit v1.2.3-55-g7522 From 4131eceb4a5e471f5a866ee10e680f0081376e3b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 6 Jun 2014 11:10:11 +0300 Subject: ASoC: Intel: Show Baytrail SST DSP firmware details during init DSP initialization complete message IPC_IA_FW_INIT_CMPLT is a large message carrying firmware details in mailbox. Read and show those details during init in order to be able to get that information to QA reports. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-ipc.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index d207b22ea330..67673a2c0f41 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -122,6 +122,26 @@ struct sst_byt_tstamp { u32 channel_peak[8]; } __packed; +struct sst_byt_fw_version { + u8 build; + u8 minor; + u8 major; + u8 type; +} __packed; + +struct sst_byt_fw_build_info { + u8 date[16]; + u8 time[16]; +} __packed; + +struct sst_byt_fw_init { + struct sst_byt_fw_version fw_version; + struct sst_byt_fw_build_info build_info; + u16 result; + u8 module_id; + u8 debug_info; +} __packed; + /* driver internal IPC message structure */ struct ipc_message { struct list_head list; @@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt; struct sst_fw *byt_sst_fw; + struct sst_byt_fw_init init; int err; dev_dbg(dev, "initialising Byt DSP IPC\n"); @@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) goto boot_err; } + /* show firmware information */ + sst_dsp_inbox_read(byt->dsp, &init, sizeof(init)); + dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n", + init.fw_version.major, init.fw_version.minor, + init.fw_version.build, init.fw_version.type); + dev_info(byt->dev, "Build type: %x\n", init.fw_version.type); + dev_info(byt->dev, "Build date: %s %s\n", + init.build_info.date, init.build_info.time); + pdata->dsp = byt; byt->fw = byt_sst_fw; -- cgit v1.2.3-55-g7522 From c9a8e3bd3df0e25d4ac9f6be1ba294004bb0bc9a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 9 Jun 2014 14:39:06 +0300 Subject: ASoC: Intel: byt-rt5640: Enable headset mic bias voltage Connect "Headset Mic" to "MICBIAS1" supply widget of RT5640 in order to enable bias voltage for headset microphones. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 53d160d39972..234a58de3c53 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { }; static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { + {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"IN2N", NULL, "Headset Mic"}, {"DMIC1", NULL, "Internal Mic"}, -- cgit v1.2.3-55-g7522 From 6cc0f4e63994a2b77fb6cd7c3bc1e25b7bdb9881 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:51 +0530 Subject: ASoC: Intel: mfld_pcm: move stream handling to dai_ops This helps us to handle pcm and compress ops seperately and per dai Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 112 ++++++++++++++++++-------------- 1 file changed, 63 insertions(+), 49 deletions(-) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7c790f51d259..0d46005752bc 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -230,19 +230,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) } /* end -- helper functions */ -static int sst_platform_open(struct snd_pcm_substream *substream) +static int sst_media_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + int ret_val = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; - int ret_val; - - pr_debug("sst_platform_open called\n"); - - snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); - ret_val = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret_val < 0) - return ret_val; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) @@ -251,50 +244,54 @@ static int sst_platform_open(struct snd_pcm_substream *substream) /* get the sst ops */ mutex_lock(&sst_lock); - if (!sst) { + if (!sst || + !try_module_get(sst->dev->driver->owner)) { pr_err("no device available to run\n"); - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - if (!try_module_get(sst->dev->driver->owner)) { - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; + ret_val = -ENODEV; + goto out_ops; } stream->ops = sst->ops; mutex_unlock(&sst_lock); stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; /* allocate memory for SST API set */ runtime->private_data = stream; - return 0; + /* Make sure, that the period size is always even */ + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIODS, 2); + + return snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); +out_ops: + kfree(stream); + mutex_unlock(&sst_lock); + return ret_val; } -static int sst_platform_close(struct snd_pcm_substream *substream) +static void sst_media_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; - pr_debug("sst_platform_close called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->ops->close(str_id); module_put(sst->dev->driver->owner); kfree(stream); - return ret_val; + return; } -static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +static int sst_media_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; - pr_debug("sst_platform_pcm_prepare called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { @@ -316,6 +313,41 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) return ret_val; } +static int sst_media_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + return 0; +} + +static int sst_media_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_soc_dai_ops sst_media_dai_ops = { + .startup = sst_media_open, + .shutdown = sst_media_close, + .prepare = sst_media_prepare, + .hw_params = sst_media_hw_params, + .hw_free = sst_media_hw_free, +}; + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (substream->pcm->internal) + return 0; + + runtime = substream->runtime; + runtime->hw = sst_platform_pcm_hw; + return 0; +} + static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -377,32 +409,14 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } - return stream->stream_info.buffer_ptr; -} - -static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - - return 0; -} - -static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); + return str_info->buffer_ptr; } static struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, - .close = sst_platform_close, .ioctl = snd_pcm_lib_ioctl, - .prepare = sst_platform_pcm_prepare, .trigger = sst_platform_pcm_trigger, .pointer = sst_platform_pcm_pointer, - .hw_params = sst_platform_pcm_hw_params, - .hw_free = sst_platform_pcm_hw_free, }; static void sst_pcm_free(struct snd_pcm *pcm) @@ -413,15 +427,15 @@ static void sst_pcm_free(struct snd_pcm *pcm) static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int retval = 0; - pr_debug("sst_pcm_new called\n"); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + if (dai->driver->playback.channels_min || + dai->driver->capture.channels_min) { retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { pr_err("dma buffer allocationf fail\n"); -- cgit v1.2.3-55-g7522 From 9daa5bd34f84e43f23ce996d43da5f39348ae8fd Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:52 +0530 Subject: ASoC: Intel: mfld-pcm rename period callback arg The argument was called mad_substream which is no longer apt as older driver is not used anymore so rename as arg Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 10 +++++----- sound/soc/intel/sst-mfld-platform.h | 4 ++-- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 0d46005752bc..4528946f5e9e 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -192,9 +192,9 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) return ret_val; } -static void sst_period_elapsed(void *mad_substream) +static void sst_period_elapsed(void *arg) { - struct snd_pcm_substream *substream = mad_substream; + struct snd_pcm_substream *substream = arg; struct sst_runtime_stream *stream; int status; @@ -218,7 +218,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) pr_debug("setting buffer ptr param\n"); sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; ret_val = stream->ops->device_control( @@ -255,7 +255,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, stream->stream_info.str_id = 0; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; /* allocate memory for SST API set */ runtime->private_data = stream; @@ -363,7 +363,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, pr_debug("sst: Trigger Start\n"); str_cmd = SST_SND_START; status = SST_PLATFORM_RUNNING; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c5e7dc49e3c..6d929c7d7bbb 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -39,8 +39,8 @@ extern struct sst_device *sst; struct pcm_stream_info { int str_id; - void *mad_substream; - void (*period_elapsed) (void *mad_substream); + void *arg; + void (*period_elapsed) (void *arg); unsigned long long buffer_ptr; int sfreq; }; -- cgit v1.2.3-55-g7522 From 2a6358250081c24cc1e564cb82ecbfd19d8c7238 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:53 +0530 Subject: ASoc: Intel: mfld-pcm: report pcm delay Now the DSP is capable of reporting the delay, report it to upper layers Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 1 + sound/soc/intel/sst-mfld-platform.h | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 4528946f5e9e..80879e5fcb49 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -409,6 +409,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } + substream->runtime->delay = str_info->pcm_delay; return str_info->buffer_ptr; } diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6d929c7d7bbb..33a0a2776238 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -42,6 +42,7 @@ struct pcm_stream_info { void *arg; void (*period_elapsed) (void *arg); unsigned long long buffer_ptr; + unsigned long long pcm_delay; int sfreq; }; -- cgit v1.2.3-55-g7522 From aa9b045f70160c664291d5482270baf2ed89cc1b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:54 +0530 Subject: ASoC: Intel: add the mrfld fw IPC definations This will be used to update current driver as well as in support for the mrfld patches Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-dsp.h | 414 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 403 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h index 8d482d76475a..2c887855e7d8 100644 --- a/sound/soc/intel/sst-mfld-dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -3,7 +3,7 @@ /* * sst_mfld_dsp.h - Intel SST Driver for audio engine * - * Copyright (C) 2008-12 Intel Corporation + * Copyright (C) 2008-14 Intel Corporation * Authors: Vinod Koul * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * @@ -19,6 +19,142 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#define SST_MAX_BIN_BYTES 1024 + +#define MAX_DBG_RW_BYTES 80 +#define MAX_NUM_SCATTER_BUFFERS 8 +#define MAX_LOOP_BACK_DWORDS 8 +/* IPC base address and mailbox, timestamp offsets */ +#define SST_MAILBOX_SIZE 0x0400 +#define SST_MAILBOX_SEND 0x0000 +#define SST_TIME_STAMP 0x1800 +#define SST_TIME_STAMP_MRFLD 0x800 +#define SST_RESERVED_OFFSET 0x1A00 +#define SST_SCU_LPE_MAILBOX 0x1000 +#define SST_LPE_SCU_MAILBOX 0x1400 +#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16) +#define PROCESS_MSG 0x80 + +/* Message ID's for IPC messages */ +/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */ + +/* I2L Firmware/Codec Download msgs */ +#define IPC_IA_PREP_LIB_DNLD 0x01 +#define IPC_IA_LIB_DNLD_CMPLT 0x02 +#define IPC_IA_GET_FW_VERSION 0x04 +#define IPC_IA_GET_FW_BUILD_INF 0x05 +#define IPC_IA_GET_FW_INFO 0x06 +#define IPC_IA_GET_FW_CTXT 0x07 +#define IPC_IA_SET_FW_CTXT 0x08 +#define IPC_IA_PREPARE_SHUTDOWN 0x31 +/* I2L Codec Config/control msgs */ +#define IPC_PREP_D3 0x10 +#define IPC_IA_SET_CODEC_PARAMS 0x10 +#define IPC_IA_GET_CODEC_PARAMS 0x11 +#define IPC_IA_SET_PPP_PARAMS 0x12 +#define IPC_IA_GET_PPP_PARAMS 0x13 +#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA +#define IPC_IA_ALG_PARAMS 0x1A +#define IPC_IA_TUNING_PARAMS 0x1B +#define IPC_IA_SET_RUNTIME_PARAMS 0x1C +#define IPC_IA_SET_PARAMS 0x1 +#define IPC_IA_GET_PARAMS 0x2 + +#define IPC_EFFECTS_CREATE 0xE +#define IPC_EFFECTS_DESTROY 0xF + +/* I2L Stream config/control msgs */ +#define IPC_IA_ALLOC_STREAM_MRFLD 0x2 +#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */ +#define IPC_IA_FREE_STREAM_MRFLD 0x03 +#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */ +#define IPC_IA_SET_STREAM_PARAMS 0x22 +#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12 +#define IPC_IA_GET_STREAM_PARAMS 0x23 +#define IPC_IA_PAUSE_STREAM 0x24 +#define IPC_IA_PAUSE_STREAM_MRFLD 0x4 +#define IPC_IA_RESUME_STREAM 0x25 +#define IPC_IA_RESUME_STREAM_MRFLD 0x5 +#define IPC_IA_DROP_STREAM 0x26 +#define IPC_IA_DROP_STREAM_MRFLD 0x07 +#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */ +#define IPC_IA_DRAIN_STREAM_MRFLD 0x8 +#define IPC_IA_CONTROL_ROUTING 0x29 +#define IPC_IA_VTSV_UPDATE_MODULES 0x20 +#define IPC_IA_VTSV_DETECTED 0x21 + +#define IPC_IA_START_STREAM_MRFLD 0X06 +#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */ + +#define IPC_IA_SET_GAIN_MRFLD 0x21 +/* Debug msgs */ +#define IPC_IA_DBG_MEM_READ 0x40 +#define IPC_IA_DBG_MEM_WRITE 0x41 +#define IPC_IA_DBG_LOOP_BACK 0x42 +#define IPC_IA_DBG_LOG_ENABLE 0x45 +#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47 + +/* L2I Firmware/Codec Download msgs */ +#define IPC_IA_FW_INIT_CMPLT 0x81 +#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01 +#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11 + +/* L2I Codec Config/control msgs */ +#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */ + +#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */ +#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */ +#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */ +#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */ +#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */ +#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */ + +#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */ +/* L2S messages */ +#define IPC_SC_DDR_LINK_UP 0xC0 +#define IPC_SC_DDR_LINK_DOWN 0xC1 +#define IPC_SC_SET_LPECLK_REQ 0xC2 +#define IPC_SC_SSP_BIT_BANG 0xC3 + +/* L2I Error reporting msgs */ +#define IPC_IA_MEM_ALLOC_FAIL 0xE0 +#define IPC_IA_PROC_ERR 0xE1 /* error in processing a + stream can be used by playback and + capture modules */ + +/* L2I Debug msgs */ +#define IPC_IA_PRINT_STRING 0xF0 + +/* Buffer under-run */ +#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B + +/* Mrfld specific defines: + * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR) + * received from FW, the format is: + * - IPC High: pvt_id is set to zero. Always short message. + * - msg_id is in lower 16-bits of IPC low payload. + * - pipe_id is in higher 16-bits of IPC low payload for period_elapsed. + * - error id is in higher 16-bits of IPC low payload for async errors. + */ +#define SST_ASYNC_DRV_ID 0 + +/* Command Response or Acknowledge message to any IPC message will have + * same message ID and stream ID information which is sent. + * There is no specific Ack message ID. The data field is used as response + * meaning. + */ +enum ackData { + IPC_ACK_SUCCESS = 0, + IPC_ACK_FAILURE, +}; + +enum ipc_ia_msg_id { + IPC_CMD = 1, /*!< Task Control message ID */ + IPC_SET_PARAMS = 2,/*!< Task Set param message ID */ + IPC_GET_PARAMS = 3, /*!< Task Get param message ID */ + IPC_INVALID = 0xFF, /*! Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 102 ++++++++++++++++++++++---------- sound/soc/intel/sst-mfld-platform.h | 2 +- 2 files changed, 71 insertions(+), 33 deletions(-) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 80879e5fcb49..6e7bfb1bc4aa 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -143,52 +143,90 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream) return state; } +static void sst_fill_alloc_params(struct snd_pcm_substream *substream, + struct snd_sst_alloc_params_ext *alloc_param) +{ + unsigned int channels; + snd_pcm_uframes_t period_size; + ssize_t periodbytes; + ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + + channels = substream->runtime->channels; + period_size = substream->runtime->period_size; + periodbytes = samples_to_bytes(substream->runtime, period_size); + alloc_param->ring_buf_info[0].addr = buffer_addr; + alloc_param->ring_buf_info[0].size = buffer_bytes; + alloc_param->sg_count = 1; + alloc_param->reserved = 0; + alloc_param->frag_size = periodbytes * channels; + +} static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) + struct snd_sst_stream_params *param) { + param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param->uc.pcm_params.sfreq = substream->runtime->rate; - param->num_chan = (u8) substream->runtime->channels; - param->pcm_wd_sz = substream->runtime->sample_bits; - param->reserved = 0; - param->sfreq = substream->runtime->rate; - param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); - param->period_count = substream->runtime->period_size; - param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); + /* PCM stream via ALSA interface */ + param->uc.pcm_params.use_offload_path = 0; + param->uc.pcm_params.reserved2 = 0; + memset(param->uc.pcm_params.channel_map, 0, sizeof(u8)); + +} +int sst_fill_stream_params(void *substream, + struct snd_sst_params *str_params, bool is_compress) +{ + struct snd_pcm_substream *pstream = NULL; + struct snd_compr_stream *cstream = NULL; + + if (is_compress == true) + cstream = (struct snd_compr_stream *)substream; + else + pstream = (struct snd_pcm_substream *)substream; + + str_params->stream_type = SST_STREAM_TYPE_MUSIC; + + /* For pcm streams */ + if (pstream) + str_params->ops = (u8)pstream->stream; + if (cstream) + str_params->ops = (u8)cstream->direction; + + return 0; } -static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, + struct snd_soc_platform *platform) { struct sst_runtime_stream *stream = substream->runtime->private_data; - struct sst_pcm_params param = {0}; - struct sst_stream_params str_params = {0}; - int ret_val; + struct snd_sst_stream_params param = {{{0,},},}; + struct snd_sst_params str_params = {0}; + struct snd_sst_alloc_params_ext alloc_params = {0}; + int ret_val = 0; /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); + sst_fill_alloc_params(substream, &alloc_params); substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; - str_params.codec = param.codec; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.device_type = substream->pcm->device + 1; - pr_debug("Playbck stream,Device %d\n", - substream->pcm->device); - } else { - str_params.ops = STREAM_OPS_CAPTURE; - str_params.device_type = SND_SST_DEVICE_CAPTURE; - pr_debug("Capture stream,Device %d\n", - substream->pcm->device); - } - ret_val = stream->ops->open(&str_params); - pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + str_params.aparams = alloc_params; + str_params.codec = SST_CODEC_TYPE_PCM; + + /* fill the device type and stream id to pass to SST driver */ + ret_val = sst_fill_stream_params(substream, &str_params, false); if (ret_val < 0) return ret_val; - stream->stream_info.str_id = ret_val; - pr_debug("str id : %d\n", stream->stream_info.str_id); + stream->stream_info.str_id = str_params.stream_id; + + ret_val = stream->ops->open(&str_params); + if (ret_val <= 0) + return ret_val; + + return ret_val; } @@ -300,8 +338,8 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, return ret_val; } - ret_val = sst_platform_alloc_stream(substream); - if (ret_val < 0) + ret_val = sst_platform_alloc_stream(substream, dai->platform); + if (ret_val <= 0) return ret_val; snprintf(substream->pcm->id, sizeof(substream->pcm->id), "%d", stream->stream_info.str_id); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 33a0a2776238..aa5ddbb26d93 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -125,7 +125,7 @@ struct compress_sst_ops { }; struct sst_ops { - int (*open) (struct sst_stream_params *str_param); + int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); int (*close) (unsigned int str_id); }; -- cgit v1.2.3-55-g7522 From 61b165caa686b8334379293d0e241f740fac195a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:56 +0530 Subject: ASoC: Intel: add mrfld pipelines Merrifield DSP used various pipelines to identify the streams and processing modules. Add these defination in the pcm driver and also add a table for device entries to firmware pipeline id conversion Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- arch/x86/include/asm/platform_sst_audio.h | 78 +++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.h | 30 +++++++++ sound/soc/intel/sst-mfld-platform-pcm.c | 100 +++++++++++++++++++++++++++--- sound/soc/intel/sst-mfld-platform.h | 18 ++++++ 4 files changed, 217 insertions(+), 9 deletions(-) create mode 100644 arch/x86/include/asm/platform_sst_audio.h create mode 100644 sound/soc/intel/sst-atom-controls.h diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h new file mode 100644 index 000000000000..0a4e140315b6 --- /dev/null +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -0,0 +1,78 @@ +/* + * platform_sst_audio.h: sst audio platform data header file + * + * Copyright (C) 2012-14 Intel Corporation + * Author: Jeeja KP + * Omair Mohammed Abdullah + * Vinod Koul ,vinod.koul@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. + */ +#ifndef _PLATFORM_SST_AUDIO_H_ +#define _PLATFORM_SST_AUDIO_H_ + +#include + +enum sst_audio_task_id_mrfld { + SST_TASK_ID_NONE = 0, + SST_TASK_ID_SBA = 1, + SST_TASK_ID_MEDIA = 3, + SST_TASK_ID_MAX = SST_TASK_ID_MEDIA, +}; + +/* Device IDs for Merrifield are Pipe IDs, + * ref: DSP spec v0.75 */ +enum sst_audio_device_id_mrfld { + /* Output pipeline IDs */ + PIPE_ID_OUT_START = 0x0, + PIPE_CODEC_OUT0 = 0x2, + PIPE_CODEC_OUT1 = 0x3, + PIPE_SPROT_LOOP_OUT = 0x4, + PIPE_MEDIA_LOOP1_OUT = 0x5, + PIPE_MEDIA_LOOP2_OUT = 0x6, + PIPE_VOIP_OUT = 0xC, + PIPE_PCM0_OUT = 0xD, + PIPE_PCM1_OUT = 0xE, + PIPE_PCM2_OUT = 0xF, + PIPE_MEDIA0_OUT = 0x12, + PIPE_MEDIA1_OUT = 0x13, +/* Input Pipeline IDs */ + PIPE_ID_IN_START = 0x80, + PIPE_CODEC_IN0 = 0x82, + PIPE_CODEC_IN1 = 0x83, + PIPE_SPROT_LOOP_IN = 0x84, + PIPE_MEDIA_LOOP1_IN = 0x85, + PIPE_MEDIA_LOOP2_IN = 0x86, + PIPE_VOIP_IN = 0x8C, + PIPE_PCM0_IN = 0x8D, + PIPE_PCM1_IN = 0x8E, + PIPE_MEDIA0_IN = 0x8F, + PIPE_MEDIA1_IN = 0x90, + PIPE_MEDIA2_IN = 0x91, + PIPE_RSVD = 0xFF, +}; + +/* The stream map for each platform consists of an array of the below + * stream map structure. + */ +struct sst_dev_stream_map { + u8 dev_num; /* device id */ + u8 subdev_num; /* substream */ + u8 direction; + u8 device_id; /* fw id */ + u8 task_id; /* fw task */ + u8 status; +}; + +struct sst_platform_data { + /* Intel software platform id*/ + struct sst_dev_stream_map *pdev_strm_map; + unsigned int strm_map_size; +}; + +int add_sst_platform_device(void); +#endif + diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h new file mode 100644 index 000000000000..14063ab8c7c5 --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.h @@ -0,0 +1,30 @@ +/* + * Copyright (C) 2013-14 Intel Corp + * Author: Ramesh Babu + * Omair M Abdullah + * Samreen Nilofer + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SST_CONTROLS_V2_H__ +#define __SST_CONTROLS_V2_H__ + +enum { + MERR_DPCM_AUDIO = 0, + MERR_DPCM_COMPR, +}; + + +#endif diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 6e7bfb1bc4aa..7de87887d9f8 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -1,7 +1,7 @@ /* * sst_mfld_platform.c - Intel MID Platform driver * - * Copyright (C) 2010-2013 Intel Corp + * Copyright (C) 2010-2014 Intel Corp * Author: Vinod Koul * Author: Harsha Priya * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -27,7 +27,9 @@ #include #include #include +#include #include "sst-mfld-platform.h" +#include "sst-atom-controls.h" struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -92,6 +94,13 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { .fifo_size = SST_FIFO_SIZE, }; +static struct sst_dev_stream_map dpcm_strm_map[] = { + {0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */ + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0}, +}; + /* MFLD - MSIC */ static struct snd_soc_dai_driver sst_platform_dai[] = { { @@ -175,12 +184,36 @@ static void sst_fill_pcm_params(struct snd_pcm_substream *substream, memset(param->uc.pcm_params.channel_map, 0, sizeof(u8)); } + +static int sst_get_stream_mapping(int dev, int sdev, int dir, + struct sst_dev_stream_map *map, int size) +{ + int i; + + if (map == NULL) + return -EINVAL; + + + /* index 0 is not used in stream map */ + for (i = 1; i < size; i++) { + if ((map[i].dev_num == dev) && (map[i].direction == dir)) + return i; + } + return 0; +} + int sst_fill_stream_params(void *substream, - struct snd_sst_params *str_params, bool is_compress) + const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress) { + int map_size; + int index; + struct sst_dev_stream_map *map; struct snd_pcm_substream *pstream = NULL; struct snd_compr_stream *cstream = NULL; + map = ctx->pdata->pdev_strm_map; + map_size = ctx->pdata->strm_map_size; + if (is_compress == true) cstream = (struct snd_compr_stream *)substream; else @@ -189,11 +222,32 @@ int sst_fill_stream_params(void *substream, str_params->stream_type = SST_STREAM_TYPE_MUSIC; /* For pcm streams */ - if (pstream) + if (pstream) { + index = sst_get_stream_mapping(pstream->pcm->device, + pstream->number, pstream->stream, + map, map_size); + if (index <= 0) + return -EINVAL; + + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + str_params->ops = (u8)pstream->stream; - if (cstream) - str_params->ops = (u8)cstream->direction; + } + + if (cstream) { + index = sst_get_stream_mapping(cstream->device->device, + 0, cstream->direction, + map, map_size); + if (index <= 0) + return -EINVAL; + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + str_params->ops = (u8)cstream->direction; + } return 0; } @@ -206,6 +260,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, struct snd_sst_params str_params = {0}; struct snd_sst_alloc_params_ext alloc_params = {0}; int ret_val = 0; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); @@ -216,7 +271,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, str_params.codec = SST_CODEC_TYPE_PCM; /* fill the device type and stream id to pass to SST driver */ - ret_val = sst_fill_stream_params(substream, &str_params, false); + ret_val = sst_fill_stream_params(substream, ctx, &str_params, false); if (ret_val < 0) return ret_val; @@ -321,7 +376,22 @@ static void sst_media_close(struct snd_pcm_substream *substream, ret_val = stream->ops->close(str_id); module_put(sst->dev->driver->owner); kfree(stream); - return; +} + +static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform, + struct snd_pcm_substream *substream) +{ + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; + struct sst_runtime_stream *stream = + substream->runtime->private_data; + u32 str_id = stream->stream_info.str_id; + unsigned int pipe_id; + pipe_id = map[str_id].device_id; + + pr_debug("%s: got pipe_id = %#x for str_id = %d\n", + __func__, pipe_id, str_id); + return pipe_id; } static int sst_media_prepare(struct snd_pcm_substream *substream, @@ -498,10 +568,22 @@ static const struct snd_soc_component_driver sst_component = { static int sst_platform_probe(struct platform_device *pdev) { + struct sst_data *drv; int ret; + struct sst_platform_data *pdata = pdev->dev.platform_data; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (sst == NULL) { + pr_err("kzalloc failed\n"); + return -ENOMEM; + } + + pdata->pdev_strm_map = dpcm_strm_map; + pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map); + drv->pdata = pdata; + mutex_init(&drv->lock); + dev_set_drvdata(&pdev->dev, drv); - pr_debug("sst_platform_probe called\n"); - sst = NULL; ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { pr_err("registering soc platform failed\n"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index aa5ddbb26d93..33891a86b3e7 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -144,10 +144,28 @@ struct sst_device { char *name; struct device *dev; struct sst_ops *ops; + struct platform_device *pdev; struct compress_sst_ops *compr_ops; }; +struct sst_data; + void sst_set_stream_status(struct sst_runtime_stream *stream, int state); +struct sst_algo_int_control_v2 { + struct soc_mixer_control mc; + u16 module_id; /* module identifieer */ + u16 pipe_id; /* location info: pipe_id + instance_id */ + u16 instance_id; + unsigned int value; /* Value received is stored here */ +}; + +struct sst_data { + struct platform_device *pdev; + struct sst_platform_data *pdata; + struct mutex lock; +}; + int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); + #endif -- cgit v1.2.3-55-g7522 From 0ec66fed40e31e74a762dd7166a9bf62ebbae5da Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:57 +0530 Subject: ASoC: Intel: use common stream allocation method for compressed stream As added in previosu patch along with stream to piep conversion si required for compressed audio too Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-compress.c | 11 ++++++++--- sound/soc/intel/sst-mfld-platform.h | 7 +++---- 2 files changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 02abd19fce1d..29c059ca19e8 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, int retval; struct snd_sst_params str_params; struct sst_compress_cb cb; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); stream = cstream->runtime->private_data; /* construct fw structure for this*/ memset(&str_params, 0, sizeof(str_params)); - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.stream_type = SST_STREAM_TYPE_MUSIC; - str_params.device_type = SND_SST_DEVICE_COMPRESS; + /* fill the device type and stream id to pass to SST driver */ + retval = sst_fill_stream_params(cstream, ctx, &str_params, true); + pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval); + if (retval < 0) + return retval; switch (params->codec.id) { case SND_AUDIOCODEC_MP3: { diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 33891a86b3e7..9dc962ff1e1d 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -149,8 +149,10 @@ struct sst_device { }; struct sst_data; - void sst_set_stream_status(struct sst_runtime_stream *stream, int state); +int sst_fill_stream_params(void *substream, const struct sst_data *ctx, + struct snd_sst_params *str_params, bool is_compress); + struct sst_algo_int_control_v2 { struct soc_mixer_control mc; u16 module_id; /* module identifieer */ @@ -158,14 +160,11 @@ struct sst_algo_int_control_v2 { u16 instance_id; unsigned int value; /* Value received is stored here */ }; - struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; struct mutex lock; }; - int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); - #endif -- cgit v1.2.3-55-g7522 From b29d7c5f7126d3c0b9984fcfd74ea82ec4fb3510 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:41 +0300 Subject: ASoC: Intel: byt-max98090: Do not change speaker and DMIC with jack state Kernel should not enable/disable speakers and digital microphone whenever jack is inserted/removed. This is more use-case than kernel specific decision. For instance one may want to play VoIP ring tones using both speakers and headphone but play music only from one of them. Because of above reason remove "Ext Spk" and "Int Mic" update when jack state is changed. Also this update was illogical anyway: "Ext Spk" was enabled when jack was inserted and disabled when jack was removed. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 5fc98c64a3f4..da0b2ee28dff 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -64,14 +64,6 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Ext Spk", - .mask = SND_JACK_LINEOUT, - }, - { - .pin = "Int Mic", - .mask = SND_JACK_LINEIN, - }, }; static struct snd_soc_jack_gpio hs_jack_gpios[] = { -- cgit v1.2.3-55-g7522 From 52b896cfef00289b5966b9b0e22b865511238216 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:22:55 +0100 Subject: ASoC: kirkwood-i2s: provide helper KIRKWOOD_RECCTL_ENABLE_MASK definition Add a KIRKWOOD_RECCTL_ENABLE_MASK definition to complement the existing PLAYCTL definition, and make use of it where we wish to clear both enable bits. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 7 +++---- sound/soc/kirkwood/kirkwood.h | 3 +++ 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 9f842222e798..55af6c8e4603 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -322,8 +322,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, else ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ - value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | - KIRKWOOD_RECCTL_SPDIF_EN); + value = ctl & ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -347,7 +346,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, /* disable all records */ value = readl(priv->io + KIRKWOOD_RECCTL); - value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + value &= ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); break; @@ -411,7 +410,7 @@ static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) writel(value, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_RECCTL); - value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + value &= ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); return 0; diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index bf23afbba1d7..ab21de090938 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -38,6 +38,9 @@ #define KIRKWOOD_RECCTL_SIZE_24 (1<<0) #define KIRKWOOD_RECCTL_SIZE_32 (0<<0) +#define KIRKWOOD_RECCTL_ENABLE_MASK (KIRKWOOD_RECCTL_SPDIF_EN | \ + KIRKWOOD_RECCTL_I2S_EN) + #define KIRKWOOD_REC_BUF_ADDR 0x1004 #define KIRKWOOD_REC_BUF_SIZE 0x1008 #define KIRKWOOD_REC_BYTE_COUNT 0x100C -- cgit v1.2.3-55-g7522 From 6772190632ebce6c5c6010d2bc77d5de866831b6 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:00 +0100 Subject: ASoC: kirkwood-i2s: fix RECCTL masking Since we wish to disable capture inputs for some formats, we need to ensure that we clear the enable bits in our cached record control register. This seems to have been missed, resulting in the register only accumulating enable bits. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 55af6c8e4603..ef1a164d8703 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -212,7 +212,8 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, KIRKWOOD_PLAYCTL_SIZE_MASK); priv->ctl_play |= ctl_play; } else { - priv->ctl_rec &= ~KIRKWOOD_RECCTL_SIZE_MASK; + priv->ctl_rec &= ~(KIRKWOOD_RECCTL_ENABLE_MASK | + KIRKWOOD_RECCTL_SIZE_MASK); priv->ctl_rec |= ctl_rec; } -- cgit v1.2.3-55-g7522 From 2fbc38219c0af91afbeb3c9d97c62e1c7c74df61 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:05 +0100 Subject: ASoC: kirkwood-i2s: fix mute handling The spec requires that the mute bits must be set while the channel is disabled. Ensure that this is the case by providing a helper which ensures that the appropriate mute bit is set while the enable bit is clear. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index ef1a164d8703..b601ad680d7b 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -222,6 +222,15 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +static unsigned kirkwood_i2s_play_mute(unsigned ctl) +{ + if (!(ctl & KIRKWOOD_PLAYCTL_I2S_EN)) + ctl |= KIRKWOOD_PLAYCTL_I2S_MUTE; + if (!(ctl & KIRKWOOD_PLAYCTL_SPDIF_EN)) + ctl |= KIRKWOOD_PLAYCTL_SPDIF_MUTE; + return ctl; +} + static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -257,7 +266,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ else ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ - + ctl = kirkwood_i2s_play_mute(ctl); value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); @@ -296,6 +305,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | KIRKWOOD_PLAYCTL_SPDIF_MUTE); + ctl = kirkwood_i2s_play_mute(ctl); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; -- cgit v1.2.3-55-g7522 From 4d2097e51795b760c392d3fbc6ca6b6f77c83419 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:10 +0100 Subject: ASoC: kirkwood-i2s: fix pause handling some more We still see the occasional timeout waiting for busy to clear. As the spec is contradictory, and we know that the current implementation doesn't work, try an alternative interpretation from the spec. This one appears to work - I have yet to find any issue with it during my testing over several months. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index b601ad680d7b..e98650c01eba 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -238,7 +238,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, uint32_t ctl, value; ctl = readl(priv->io + KIRKWOOD_PLAYCTL); - if (ctl & KIRKWOOD_PLAYCTL_PAUSE) { + if ((ctl & KIRKWOOD_PLAYCTL_ENABLE_MASK) == 0) { unsigned timeout = 5000; /* * The Armada510 spec says that if we enter pause mode, the -- cgit v1.2.3-55-g7522 From a622251c01b628cbbd1b02a877a6469303ec2b58 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:15 +0100 Subject: ASoC: kirkwood: allow smaller audio periods and smaller number of periods There is no hardware restriction requiring a minimum of 8 periods, or a minimum of 2048 bytes in a period. Let's drop these values so that userspace has more flexibility in choosing these parameters. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index ab21de090938..90e32a781424 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -124,9 +124,9 @@ /* Theses values come from the marvell alsa driver */ /* need to find where they come from */ -#define KIRKWOOD_SND_MIN_PERIODS 8 +#define KIRKWOOD_SND_MIN_PERIODS 2 #define KIRKWOOD_SND_MAX_PERIODS 16 -#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 256 #define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000 #define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ * KIRKWOOD_SND_MAX_PERIODS) -- cgit v1.2.3-55-g7522 From 920ec4e595faf89f7db022a068a4729a4d2c48ae Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:20 +0100 Subject: ASoC: kirkwood: implement NO_PERIOD_WAKEUP support Permit ALSA to run without hardware interrupts from the audio interface. Instead, ALSA will use a kernel timer to decide when to check the buffer state, resulting in a lighter workload for the CPU. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 11 ++++++----- sound/soc/kirkwood/kirkwood-i2s.c | 9 ++++++--- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index aac22fccdcdc..4cf2245950d7 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -28,11 +28,12 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) } static struct snd_pcm_hardware kirkwood_dma_snd_hw = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index e98650c01eba..0704cd6d2314 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -234,6 +234,7 @@ static unsigned kirkwood_i2s_play_mute(unsigned ctl) static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { + struct snd_pcm_runtime *runtime = substream->runtime; struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); uint32_t ctl, value; @@ -271,9 +272,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, writel(value, priv->io + KIRKWOOD_PLAYCTL); /* enable interrupts */ - value = readl(priv->io + KIRKWOOD_INT_MASK); - value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; - writel(value, priv->io + KIRKWOOD_INT_MASK); + if (!runtime->no_period_wakeup) { + value = readl(priv->io + KIRKWOOD_INT_MASK); + value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; + writel(value, priv->io + KIRKWOOD_INT_MASK); + } /* enable playback */ writel(ctl, priv->io + KIRKWOOD_PLAYCTL); -- cgit v1.2.3-55-g7522 From 053e69d57cc6253b19ea661f929c8c1b6a907bff Mon Sep 17 00:00:00 2001 From: Wonjoon Lee Date: Fri, 20 Jun 2014 13:33:15 +0530 Subject: ASoC: max98090: Add max98091 compatible string The MAX98091 CODEC is the same as MAX98090 CODEC, but with an extra microphone. Existing driver for MAX98090 CODEC already has support for MAX98091 CODEC. Adding proper compatible string so that MAX98091 CODEC can be specified from device tree. Signed-off-by: Wonjoon Lee Signed-off-by: Doug Anderson Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/max98090.txt | 2 +- sound/soc/codecs/max98090.c | 2 ++ 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index a5e63fa47dc5..c454e67f54bb 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "maxim,max98090". +- compatible : "maxim,max98090" or "maxim,max98091". - reg : The I2C address of the device. diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 3aec3ae78fe0..c00b36872dfe 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2447,12 +2447,14 @@ static const struct dev_pm_ops max98090_pm = { static const struct i2c_device_id max98090_i2c_id[] = { { "max98090", MAX98090 }, + { "max98091", MAX98091 }, { } }; MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); static const struct of_device_id max98090_of_match[] = { { .compatible = "maxim,max98090", }, + { .compatible = "maxim,max98091", }, { } }; MODULE_DEVICE_TABLE(of, max98090_of_match); -- cgit v1.2.3-55-g7522 From 07cf7cbadb4d97a78be61119a406de8fe446467e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 20 Jun 2014 14:41:13 +0800 Subject: ASoC: add RT286 CODEC driver This patch adds Realtek ALC286 codec driver. ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. We map verb command to regmap structure. However, we read most registers from cache to prevent the asymmetry read/write issue in rt286. Signed-off-by: Bard Liao Signed-off-by: Gustaw Lewandowski Signed-off-by: Mark Brown --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1208 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 193 ++++++++ 5 files changed, 1426 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 000000000000..eb773d1485f2 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * 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 __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..115e5de7b7eb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,6 +74,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C @@ -449,6 +450,9 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b8d73f..c39449a862b3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -233,6 +234,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 000000000000..acfba9c74c52 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1208 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rt286.h" + +#define RT286_VENDOR_ID 0x10ec0286 + +struct rt286_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + struct reg_default *index_cache; +}; + +static struct reg_default rt286_index_def[] = { + { 0x01, 0xaaaa }, + { 0x02, 0x8aaa }, + { 0x03, 0x0002 }, + { 0x04, 0xaf01 }, + { 0x08, 0x000d }, + { 0x09, 0xd810 }, + { 0x0a, 0x0060 }, + { 0x0b, 0x0000 }, + { 0x0f, 0x0000 }, + { 0x19, 0x0a17 }, + { 0x20, 0x0020 }, + { 0x33, 0x0208 }, + { 0x49, 0x0004 }, + { 0x4f, 0x50e9 }, + { 0x50, 0x2c00 }, + { 0x63, 0x2902 }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) + +static const struct reg_default rt286_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x0000007f }, + { 0x0023a000, 0x0000007f }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x000000c3 }, + { 0x00936000, 0x000000c3 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000097 }, + { 0x00b37200, 0x00000097 }, + { 0x00b37300, 0x00000097 }, + { 0x00c37000, 0x00000000 }, + { 0x00c37100, 0x00000080 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01439000, 0x00000080 }, + { 0x0143a000, 0x00000080 }, + { 0x01470700, 0x00000000 }, + { 0x01470500, 0x00000400 }, + { 0x01470c00, 0x00000000 }, + { 0x01470100, 0x00000000 }, + { 0x01837000, 0x00000000 }, + { 0x01870500, 0x00000400 }, + { 0x02050000, 0x00000000 }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000000 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x02040000, 0x00004002 }, + { 0x01870700, 0x00000020 }, + { 0x00830000, 0x000000c3 }, + { 0x00930000, 0x000000c3 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt286_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_PROC_COEF: + return true; + default: + return false; + } + + +} + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_SET_AUDIO_POWER: + case RT286_SET_HPO_POWER: + case RT286_SET_SPK_POWER: + case RT286_SET_DMIC1_POWER: + case RT286_SPK_MUX: + case RT286_HPO_MUX: + case RT286_ADC0_MUX: + case RT286_ADC1_MUX: + case RT286_SET_MIC1: + case RT286_SET_PIN_HPO: + case RT286_SET_PIN_SPK: + case RT286_SET_PIN_DMIC1: + case RT286_SPK_EAPD: + case RT286_SET_AMP_GAIN_HPO: + case RT286_SET_DMIC2_DEFAULT: + case RT286_DACL_GAIN: + case RT286_DACR_GAIN: + case RT286_ADCL_GAIN: + case RT286_ADCR_GAIN: + case RT286_MIC_GAIN: + case RT286_SPOL_GAIN: + case RT286_SPOR_GAIN: + case RT286_HPOL_GAIN: + case RT286_HPOR_GAIN: + case RT286_F_DAC_SWITCH: + case RT286_F_RECMIX_SWITCH: + case RT286_REC_MIC_SWITCH: + case RT286_REC_I2S_SWITCH: + case RT286_REC_LINE_SWITCH: + case RT286_REC_BEEP_SWITCH: + case RT286_DAC_FORMAT: + case RT286_ADC_FORMAT: + case RT286_COEF_INDEX: + case RT286_PROC_COEF: + case RT286_SET_AMP_GAIN_ADC_IN1: + case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_POWER(RT286_DAC_OUT1): + case RT286_SET_POWER(RT286_DAC_OUT2): + case RT286_SET_POWER(RT286_ADC_IN1): + case RT286_SET_POWER(RT286_ADC_IN2): + case RT286_SET_POWER(RT286_DMIC2): + case RT286_SET_POWER(RT286_MIC1): + return true; + default: + return false; + } +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + struct rt286_priv *rt286 = i2c_get_clientdata(client); + u8 data[4]; + int ret, i; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + if (reg == rt286->index_cache[i].reg) { + rt286->index_cache[i].def = value; + break; + } + + } + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + + if (ret == 4) + return 0; + else + pr_err("ret=%d\n", ret); + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + } + + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static void rt286_index_sync(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + } +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL2, 0x4, 0x4); + snd_soc_write(codec, RT286_SET_MIC1, 0x24); + val = snd_soc_read(codec, RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = snd_soc_read(codec, + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + snd_soc_write(codec, + RT286_SET_MIC1, 0x20); + /* power off HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x1001); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } else { + *mic = false; + } + + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE); + *mic = buf & 0x80000000; + } + + return 0; +} + +static void rt286_jack_detect_work(struct work_struct *work) +{ + struct rt286_priv *rt286 = + container_of(work, struct rt286_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, + RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN, + RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK, + RT286_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN, + RT286_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN, + RT286_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt286_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum); + +static const char * const rt286_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* 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), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, + RT286_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + else + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, + RT286_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + /* Clear IRQ */ + snd_soc_update_bits(rt286->codec, + RT286_IRQ_CTRL, 0x1, 0x1); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i, ret; + + ret = snd_soc_read(codec, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID)); + if (ret != RT286_VENDOR_ID) { + dev_err(codec->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + + snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + snd_soc_write(codec, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + snd_soc_write(codec, RT286_CBJ_CTRL2, 0x0000); + snd_soc_write(codec, RT286_MIC1_DET_CTRL, 0x0816); + snd_soc_write(codec, RT286_MISC_CTRL1, 0x0000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x8, 0x0); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + rt286->i2c->irq = 0; + if (rt286->i2c->irq) { + snd_soc_update_bits(codec, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(codec->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt286->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt286_suspend(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, true); + regcache_mark_dirty(rt286->regmap); + + return 0; +} + +static int rt286_resume(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, false); + rt286_index_sync(codec); + regcache_sync(rt286->regmap); + + return 0; +} +#else +#define rt286_suspend NULL +#define rt286_resume NULL +#endif + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt286-aif2", + .id = RT286_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .suspend = rt286_suspend, + .resume = rt286_resume, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x02370100, + .volatile_reg = rt286_volatile_register, + .readable_reg = rt286_readable_register, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt286_reg, + .num_reg_defaults = ARRAY_SIZE(rt286_reg), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static const struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt286->index_cache = rt286_index_def; + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt286); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 000000000000..21c570f88e9b --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,193 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu + * + * 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 __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_SET_PIN_SFT 6 +#define RT286_SET_PIN_ENABLE 0x40 +#define RT286_SET_PIN_DISABLE 0 +#define RT286_SET_EAPD_HIGH 0x2 +#define RT286_SET_EAPD_LOW 0 + +#define RT286_MUTE_SFT 7 + +/* Verb commands */ +#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT) +#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT) +#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1) +#define RT286_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0) +#define RT286_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0) +#define RT286_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0) +#define RT286_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0) +#define RT286_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0) +#define RT286_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0) +#define RT286_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0) +#define RT286_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) +#define RT286_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000) +#define RT286_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000) +#define RT286_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000) +#define RT286_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000) +#define RT286_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000) +#define RT286_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000) +#define RT286_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000) +#define RT286_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000) +#define RT286_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100) +#define RT286_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000) +#define RT286_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100) +#define RT286_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200) +#define RT286_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300) +#define RT286_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0) +#define RT286_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0) +#define RT286_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0) +#define RT286_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ + -- cgit v1.2.3-55-g7522 From 978b641f9563019a24032d5dee8a75963cd248ff Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 4 Jul 2014 14:42:16 +0530 Subject: ASoC: max98090: Add check for CODEC type CODEC type (MAX98090/MAX98091) can be specified from device-tree file, it can also be obtained from the CODEC during runtime. Add an explicit check to figure out if both are matching, else print a message warning about the same. Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c00b36872dfe..2c2c5b22f60f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2215,6 +2215,7 @@ static int max98090_probe(struct snd_soc_codec *codec) { struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); struct max98090_cdata *cdata; + enum max98090_type devtype; int ret = 0; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2250,16 +2251,21 @@ static int max98090_probe(struct snd_soc_codec *codec) } if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { - max98090->devtype = MAX98091; + devtype = MAX98091; dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); } else { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); } + if (max98090->devtype != devtype) { + dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n"); + max98090->devtype = devtype; + } + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); -- cgit v1.2.3-55-g7522 From eba843201a8e5824c5e6e539db6cd1a6ba84f145 Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 4 Jul 2014 14:42:17 +0530 Subject: ASoC: max98090: Remove redundant max98090_handle_pdata() max98090_handle_pdata() is not doing anything other than printing a message if pdata is not valid. This can be removed. Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 2c2c5b22f60f..0e59e5117e43 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2200,17 +2200,6 @@ static struct snd_soc_dai_driver max98090_dai[] = { } }; -static void max98090_handle_pdata(struct snd_soc_codec *codec) -{ - struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); - struct max98090_pdata *pdata = max98090->pdata; - - if (!pdata) { - dev_err(codec->dev, "No platform data\n"); - return; - } -} - static int max98090_probe(struct snd_soc_codec *codec) { struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); @@ -2310,8 +2299,6 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); - max98090_handle_pdata(codec); - max98090_add_widgets(codec); err_access: -- cgit v1.2.3-55-g7522 From 305b8d8782c3b4aa572d496769b93cc3db2ae892 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Jul 2014 16:48:36 +0800 Subject: ASoC: RT286: remove test code Remove test code. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index acfba9c74c52..7c5f9d0f0af2 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -993,7 +993,6 @@ static int rt286_probe(struct snd_soc_codec *codec) codec->dapm.bias_level = SND_SOC_BIAS_OFF; rt286->codec = codec; - rt286->i2c->irq = 0; if (rt286->i2c->irq) { snd_soc_update_bits(codec, RT286_IRQ_CTRL, 0x2, 0x2); -- cgit v1.2.3-55-g7522 From 4b21768a95d68fe26a6a9f08ca93a7c59c13fcac Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Jul 2014 16:48:37 +0800 Subject: ASoC: RT286: check ID in i2c level Move ID check from asoc level to i2c level. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 7c5f9d0f0af2..53eb7f37bb73 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -952,14 +952,6 @@ static int rt286_probe(struct snd_soc_codec *codec) struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); int i, ret; - ret = snd_soc_read(codec, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID)); - if (ret != RT286_VENDOR_ID) { - dev_err(codec->dev, - "Device with ID register %x is not rt286\n", ret); - return -ENODEV; - } - snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) @@ -1164,6 +1156,14 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return ret; } + regmap_read(rt286->regmap, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); + if (ret != RT286_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + rt286->index_cache = rt286_index_def; rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); -- cgit v1.2.3-55-g7522 From 61a414c412886bdb98c8842c00b2f0a3d4436b12 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Jul 2014 16:48:38 +0800 Subject: ASoC: RT286: move initial settings to _i2c_probe Move codec initial settings from asoc probe to i2c probe. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 99 ++++++++++++++++++++++++------------------------ 1 file changed, 49 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 53eb7f37bb73..e6f33ab78954 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -950,59 +950,10 @@ static irqreturn_t rt286_irq(int irq, void *data) static int rt286_probe(struct snd_soc_codec *codec) { struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); - int i, ret; - - snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); - - for (i = 0; i < RT286_POWER_REG_LEN; i++) - snd_soc_write(codec, - RT286_SET_POWER(rt286_support_power_controls[i]), - AC_PWRST_D1); - - if (!rt286->pdata.cbj_en) { - snd_soc_write(codec, RT286_CBJ_CTRL2, 0x0000); - snd_soc_write(codec, RT286_MIC1_DET_CTRL, 0x0816); - snd_soc_write(codec, RT286_MISC_CTRL1, 0x0000); - snd_soc_update_bits(codec, - RT286_CBJ_CTRL1, 0xf000, 0xb000); - } else { - snd_soc_update_bits(codec, - RT286_CBJ_CTRL1, 0xf000, 0x5000); - } - - mdelay(10); - - if (!rt286->pdata.gpio2_en) - snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0x4000); - else - snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0); - - mdelay(10); - - /*Power down LDO2*/ - snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x8, 0x0); codec->dapm.bias_level = SND_SOC_BIAS_OFF; rt286->codec = codec; - if (rt286->i2c->irq) { - snd_soc_update_bits(codec, - RT286_IRQ_CTRL, 0x2, 0x2); - - INIT_DELAYED_WORK(&rt286->jack_detect_work, - rt286_jack_detect_work); - schedule_delayed_work(&rt286->jack_detect_work, - msecs_to_jiffies(1250)); - - ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, - IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); - if (ret != 0) { - dev_err(codec->dev, - "Failed to reguest IRQ: %d\n", ret); - return ret; - } - } - return 0; } @@ -1141,7 +1092,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int ret; + int i, ret; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1171,6 +1122,54 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + regmap_write(rt286->regmap, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000); + regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816); + regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + + if (rt286->i2c->irq) { + regmap_update_bits(rt286->regmap, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, rt286_dai, ARRAY_SIZE(rt286_dai)); -- cgit v1.2.3-55-g7522 From bc6c4e455af9037bae619340bc95bf569806ba8b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Jul 2014 19:15:30 +0800 Subject: ASoC: RT286: Fix silent at the beginning of stream This patch fix the issue that the output is almost silent at the beginning of starting a playback. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 20 +++++++++++++++++++- sound/soc/codecs/rt286.h | 5 +++++ 2 files changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e6f33ab78954..81033154a412 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -54,6 +54,7 @@ static struct reg_default rt286_index_def[] = { { 0x09, 0xd810 }, { 0x0a, 0x0060 }, { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, { 0x19, 0x0a17 }, { 0x20, 0x0020 }, @@ -62,6 +63,9 @@ static struct reg_default rt286_index_def[] = { { 0x4f, 0x50e9 }, { 0x50, 0x2c00 }, { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, }; #define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) @@ -902,14 +906,23 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_PREPARE: - if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x200); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(10); break; case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); break; default: @@ -1152,6 +1165,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, /*Power down LDO2*/ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + /*Set depop parameter*/ + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); + if (rt286->i2c->irq) { regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index 21c570f88e9b..b539b7320a79 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -127,6 +127,7 @@ #define RT286_I2S_CTRL1 0x09 #define RT286_I2S_CTRL2 0x0a #define RT286_CLK_DIV 0x0b +#define RT286_DC_GAIN 0x0d #define RT286_POWER_CTRL3 0x0f #define RT286_MIC1_DET_CTRL 0x19 #define RT286_MISC_CTRL1 0x20 @@ -135,6 +136,10 @@ #define RT286_CBJ_CTRL1 0x4f #define RT286_CBJ_CTRL2 0x50 #define RT286_PLL_CTRL 0x63 +#define RT286_DEPOP_CTRL1 0x66 +#define RT286_DEPOP_CTRL2 0x67 +#define RT286_DEPOP_CTRL3 0x68 +#define RT286_DEPOP_CTRL4 0x69 /* SPDIF (0x06) */ #define RT286_SPDIF_SEL_SFT 0 -- cgit v1.2.3-55-g7522 From 121eb444135c25701051eb849e7ccf0dd412382b Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Mon, 7 Jul 2014 15:18:19 +0200 Subject: ASoC: max98090: Fix build warning The max98090_{suspend,resume}() functions are used for system sleep and therefore need to be guarded by CONFIG_PM_SLEEP rather than CONFIG_PM. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 0e59e5117e43..6816578ea8f7 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2408,7 +2408,7 @@ static int max98090_runtime_suspend(struct device *dev) } #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int max98090_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); -- cgit v1.2.3-55-g7522 From 15446c0b8dc79f5dfabfb689879609023713f421 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:11:09 +0800 Subject: ASoC: Intel: Merge wild cat point ADSP DRAM regions Merge D-SRAM0 D-SRAM1 D-SRAM2 to D-SRAM, for wild cat point ADSP mem regions. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 535f517629fd..4720382260b0 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -313,9 +313,7 @@ static const struct sst_adsp_memregion lp_region[] = { /* wild cat point ADSP mem regions */ static const struct sst_adsp_memregion wpt_region[] = { - {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ - {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ - {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */ + {0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */ {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */ }; -- cgit v1.2.3-55-g7522 From 548793232fd29cfa1553bb45247aa5963632405c Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:11:10 +0800 Subject: ASoC: Intel: Use a table for ADSP SRAM shift Use a table for ADSP IRAM/DRAM bit shift. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4720382260b0..40bb0205d5c0 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -337,21 +337,40 @@ static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) return 0; } +struct sst_sram_shift { + u32 dev_id; /* SST Device IDs */ + u32 iram_shift; + u32 dram_shift; +}; + +static const struct sst_sram_shift sram_shift[] = { + {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */ + {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */ +}; static u32 hsw_block_get_bit(struct sst_mem_block *block) { - u32 bit = 0, shift = 0; + u32 bit = 0, shift = 0, index; + struct sst_dsp *sst = block->dsp; - switch (block->type) { - case SST_MEM_DRAM: - shift = 16; - break; - case SST_MEM_IRAM: - shift = 6; - break; - default: - return 0; + for (index = 0; index < ARRAY_SIZE(sram_shift); index++) { + if (sram_shift[index].dev_id == sst->id) + break; } + if (index < ARRAY_SIZE(sram_shift)) { + switch (block->type) { + case SST_MEM_DRAM: + shift = sram_shift[index].dram_shift; + break; + case SST_MEM_IRAM: + shift = sram_shift[index].iram_shift; + break; + default: + shift = 0; + } + } else + shift = 0; + bit = 1 << (block->index + shift); return bit; -- cgit v1.2.3-55-g7522 From 85e63007bbef7abc7145c807ed59d01738e09d39 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:11:12 +0800 Subject: ASoC: Intel: Start with all memory banks disabled All required banks are enabled during boot procedure. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 40bb0205d5c0..977e29779d11 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -505,8 +505,9 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } } - /* set default power gating mask */ - writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0); + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0); return 0; } -- cgit v1.2.3-55-g7522 From 23c4fd5c9719e8fc60d589b9f9c7451120f4f3e9 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 14 Jul 2014 10:18:04 +0800 Subject: ASoC: rt286: make rt286_i2c_driver static Signed-off-by: Fengguang Wu Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- 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 81033154a412..218f86efd196 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static int rt286_i2c_remove(struct i2c_client *i2c) } -struct i2c_driver rt286_i2c_driver = { +static struct i2c_driver rt286_i2c_driver = { .driver = { .name = "rt286", .owner = THIS_MODULE, -- cgit v1.2.3-55-g7522 From 1ad0e33060a64121c1c7acb7ec21a4fdef4aaed6 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Jul 2014 14:57:49 +0530 Subject: ASoC: Intel: add sst shim register start-end variables the shim registers start and end can be useful while parsing the shim addresses, so add these Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index e44423be66c4..967fb32c981d 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -53,6 +53,10 @@ #define SST_CSR2 0x80 #define SST_LTRC 0xE0 #define SST_HDMC 0xE8 + +#define SST_SHIM_BEGIN SST_CSR +#define SST_SHIM_END SST_HDMC + #define SST_DBGO 0xF0 #define SST_SHIM_SIZE 0x100 -- cgit v1.2.3-55-g7522 From 8813e66db73bfac940bfd08c31592365f0a56c74 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Jul 2014 14:57:50 +0530 Subject: ASoC: Intel: mfld: add dsp error codes DSP returns error codes for IPC return so add them in driver Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-dsp.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h index 2c887855e7d8..4257263157cd 100644 --- a/sound/soc/intel/sst-mfld-dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -171,6 +171,21 @@ enum stream_type { SST_STREAM_TYPE_MUSIC = 1, }; +enum sst_error_codes { + /* Error code,response to msgId: Description */ + /* Common error codes */ + SST_SUCCESS = 0, /* Success */ + SST_ERR_INVALID_STREAM_ID = 1, + SST_ERR_INVALID_MSG_ID = 2, + SST_ERR_INVALID_STREAM_OP = 3, + SST_ERR_INVALID_PARAMS = 4, + SST_ERR_INVALID_CODEC = 5, + SST_ERR_INVALID_MEDIA_TYPE = 6, + SST_ERR_STREAM_ERR = 7, + + SST_ERR_STREAM_IN_USE = 15, +}; + struct ipc_dsp_hdr { u16 mod_index_id:8; /*!< DSP Command ID specific to tasks */ u16 pipe_id:8; /*!< instance of the module in the pipeline */ -- cgit v1.2.3-55-g7522 From e310fb9141436672db680923228fba5aab206062 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Jul 2014 14:57:51 +0530 Subject: ASoC: Intel: mfld: add generic parameter interface This interface will be used by subsequent patches to set/get parameters from DSP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform.h | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 9dc962ff1e1d..6c6a42c08e24 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -63,7 +63,9 @@ enum sst_controls { SST_SND_BUFFER_POINTER = 0x05, SST_SND_STREAM_INIT = 0x06, SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, + SST_SET_BYTE_STREAM = 0x100A, + SST_GET_BYTE_STREAM = 0x100B, + SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, }; enum sst_stream_ops { @@ -127,6 +129,7 @@ struct compress_sst_ops { struct sst_ops { int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); + int (*set_generic_params)(enum sst_controls cmd, void *arg); int (*close) (unsigned int str_id); }; -- cgit v1.2.3-55-g7522 From 1d34f3ef4b6cd33c74b414df74b41a4606d1a306 Mon Sep 17 00:00:00 2001 From: Lv Zheng Date: Wed, 16 Jul 2014 16:59:02 +0800 Subject: ASoC: intel: Cleanup useless ACPI inclusion. The sst-haswell-dsp.c is an ACPI independent file, this patch removes ACPI header files for it. Signed-off-by: Lv Zheng Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 977e29779d11..7b8ad9923fe4 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -28,9 +28,6 @@ #include #include -#include -#include - #include "sst-dsp.h" #include "sst-dsp-priv.h" #include "sst-haswell-ipc.h" -- cgit v1.2.3-55-g7522 From 249adddb1a3155718876c8473ef57717d5208e37 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 15 Jul 2014 08:51:12 +0800 Subject: ASoC: Intel: Update FW version readback Update FW version readback. IPC_GLB_GET_FW_VERSION reads back the ABI version whilst the release version is in the mailbox. Update to use mailbox version for info logging. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 434236343ddf..96373ab46f8c 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready { u32 inbox_size; u32 outbox_size; u32 fw_info_size; - u8 fw_info[1]; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; } __attribute__((packed)); struct ipc_message { @@ -569,6 +569,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) { struct sst_hsw_ipc_fw_ready fw_ready; u32 offset; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; + char *tmp[5], *pinfo; + int i = 0; offset = (header & 0x1FFFFFFF) << 3; @@ -589,6 +592,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) fw_ready.inbox_offset, fw_ready.inbox_size); dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n", fw_ready.outbox_offset, fw_ready.outbox_size); + if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) { + fw_ready.fw_info[fw_ready.fw_info_size] = 0; + dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info); + + /* log the FW version info got from the mailbox here. */ + memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); + pinfo = &fw_info[0]; + for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + tmp[i] = strsep(&pinfo, " "); + dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " + "version: %s.%s, build %s, source commit id: %s\n", + tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]); + } } static void hsw_notification_work(struct work_struct *work) @@ -1775,8 +1791,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); - dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n", - version.type, version.major, version.minor, version.build); /* get the globalmixer */ ret = sst_hsw_mixer_get_info(hsw); -- cgit v1.2.3-55-g7522 From afdb74fd708fb4330485212f76a70b91967b1f70 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 14 Jul 2014 10:35:40 +0800 Subject: ASoC: Intel: Add Broadwell Machine support Add support for Broadwell based machines with SST DSP audio. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 12 +++ sound/soc/intel/Makefile | 2 + sound/soc/intel/broadwell.c | 251 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/broadwell.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c30fedb3e149..0b305f9da3db 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH help This adds audio driver for Intel Baytrail platform based boards with the MAX98090 audio codec. + +config SND_SOC_INTEL_BROADWELL_MACH + tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_HASWELL + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT286 + help + This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell + Ultrabook platforms. + 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 4bfca79a42ba..7acbfc43a0c6 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(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 diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c new file mode 100644 index 000000000000..0e550f14028f --- /dev/null +++ b/sound/soc/intel/broadwell.c @@ -0,0 +1,251 @@ +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * 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. + * + * 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 +#include +#include +#include +#include +#include + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt286.h" + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphones", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_ssp0_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 ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int broadwell_rt286_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; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + /* always connected - check HP for jack detect */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "DMIC1"); + snd_soc_dapm_enable_pin(dapm, "DMIC2"); + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = broadwell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 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_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 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_playback = 1, + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .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, + }, + { + .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 */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .name = "broadwell-rt286", + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + broadwell_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&broadwell_rt286); + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); -- cgit v1.2.3-55-g7522 From 270a85c80c8645368f9768b120bd9eac4db23590 Mon Sep 17 00:00:00 2001 From: Andrew Lunn Date: Thu, 10 Jul 2014 23:36:23 +0200 Subject: ASoC: kirkwood: Remove unused drivers Both kirkwood-openrd and kirkwood-t5325 drivers have been replaced with DT based simple-card equivelents. So remove these drivers. Signed-off-by: Andrew Lunn Acked-by: Jason Cooper Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 17 ----- sound/soc/kirkwood/Makefile | 4 -- sound/soc/kirkwood/kirkwood-openrd.c | 109 -------------------------------- sound/soc/kirkwood/kirkwood-t5325.c | 116 ----------------------------------- 4 files changed, 246 deletions(-) delete mode 100644 sound/soc/kirkwood/kirkwood-openrd.c delete mode 100644 sound/soc/kirkwood/kirkwood-t5325.c diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 06f4e8aa93ae..1f7c7ee3527a 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB Say Y if you want to add support for SoC audio on the Armada 370 Development Board. -config SND_KIRKWOOD_SOC_OPENRD - tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) - depends on I2C - select SND_SOC_CS42L51 - help - Say Y if you want to add support for SoC audio on - Openrd Client. - -config SND_KIRKWOOD_SOC_T5325 - tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C - select SND_SOC_ALC5623 - help - Say Y if you want to add support for SoC audio on - the HP t5325 thin client. - diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 7c1d8fe09e6b..c36b03d8006c 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -snd-soc-openrd-objs := kirkwood-openrd.o -snd-soc-t5325-objs := kirkwood-t5325.o snd-soc-armada-370-db-objs := armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c deleted file mode 100644 index 65f2a5b9ec3b..000000000000 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * kirkwood-openrd.c - * - * (c) 2010 Arnaud Patard - * (c) 2010 Arnaud Patard - * - * 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 -#include -#include -#include -#include -#include -#include -#include "../codecs/cs42l51.h" - -static int openrd_client_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; - unsigned int freq; - - switch (params_rate(params)) { - default: - case 44100: - freq = 11289600; - break; - case 48000: - freq = 12288000; - break; - case 96000: - freq = 24576000; - break; - } - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops openrd_client_ops = { - .hw_params = openrd_client_hw_params, -}; - - -static struct snd_soc_dai_link openrd_client_dai[] = { -{ - .name = "CS42L51", - .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "cs42l51-hifi", - .codec_name = "cs42l51-codec.0-004a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &openrd_client_ops, -}, -}; - - -static struct snd_soc_card openrd_client = { - .name = "OpenRD Client", - .owner = THIS_MODULE, - .dai_link = openrd_client_dai, - .num_links = ARRAY_SIZE(openrd_client_dai), -}; - -static int openrd_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &openrd_client; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int openrd_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver openrd_driver = { - .driver = { - .name = "openrd-client-audio", - .owner = THIS_MODULE, - }, - .probe = openrd_probe, - .remove = openrd_remove, -}; - -module_platform_driver(openrd_driver); - -/* Module information */ -MODULE_AUTHOR("Arnaud Patard "); -MODULE_DESCRIPTION("ALSA SoC OpenRD Client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:openrd-client-audio"); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c deleted file mode 100644 index 844b8415a011..000000000000 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ /dev/null @@ -1,116 +0,0 @@ -/* - * kirkwood-t5325.c - * - * (c) 2010 Arnaud Patard - * - * 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 -#include -#include -#include -#include -#include -#include -#include "../codecs/alc5623.h" - -static int t5325_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; - unsigned int freq; - - freq = params_rate(params) * 256; - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops t5325_ops = { - .hw_params = t5325_hw_params, -}; - -static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route t5325_route[] = { - { "Headphone Jack", NULL, "HPL" }, - { "Headphone Jack", NULL, "HPR" }, - - {"Speaker", NULL, "SPKOUT"}, - {"Speaker", NULL, "SPKOUTN"}, - - { "MIC1", NULL, "Mic Jack" }, - { "MIC2", NULL, "Mic Jack" }, -}; - -static struct snd_soc_dai_link t5325_dai[] = { -{ - .name = "ALC5621", - .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "alc5621-hifi", - .codec_name = "alc562x-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &t5325_ops, -}, -}; - -static struct snd_soc_card t5325 = { - .name = "t5325", - .owner = THIS_MODULE, - .dai_link = t5325_dai, - .num_links = ARRAY_SIZE(t5325_dai), - - .dapm_widgets = t5325_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets), - .dapm_routes = t5325_route, - .num_dapm_routes = ARRAY_SIZE(t5325_route), -}; - -static int t5325_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &t5325; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int t5325_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver t5325_driver = { - .driver = { - .name = "t5325-audio", - .owner = THIS_MODULE, - }, - .probe = t5325_probe, - .remove = t5325_remove, -}; - -module_platform_driver(t5325_driver); - -MODULE_AUTHOR("Arnaud Patard "); -MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:t5325-audio"); -- cgit v1.2.3-55-g7522 From 6145dfc6d96a350f50b939d8b5b5301e17f4c2c5 Mon Sep 17 00:00:00 2001 From: Andrew Lunn Date: Thu, 10 Jul 2014 23:36:24 +0200 Subject: ASoC: kirkwood: Remove ARCH_KIRKWOOD dependency mach-kirkwood has been removed, now that kirkwood lives in mach-mvebu. Remove ARCH_KIRKWOOD since ARCH_MVEBU is sufficient. Signed-off-by: Andrew Lunn Acked-by: Jason Cooper Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 1f7c7ee3527a..132bb83f8e99 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST + depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- cgit v1.2.3-55-g7522 From ee4a6ce6cd74a9eace247656c5b109f31c73ab8d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:05:44 +0800 Subject: ASoC: Intel: Fix naming of HMDC register macros. HMDC is the correct naming for this register. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 8 ++++---- sound/soc/intel/sst-haswell-dsp.c | 6 +++--- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 967fb32c981d..21a85eb196f1 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -52,7 +52,7 @@ #define SST_CLKCTL 0x78 #define SST_CSR2 0x80 #define SST_LTRC 0xE0 -#define SST_HDMC 0xE8 +#define SST_HMDC 0xE8 #define SST_SHIM_BEGIN SST_CSR #define SST_SHIM_END SST_HDMC @@ -122,9 +122,9 @@ /* LTRC */ #define SST_LTRC_VAL(x) (x << 0) -/* HDMC */ -#define SST_HDMC_HDDA0(x) (x << 0) -#define SST_HDMC_HDDA1(x) (x << 7) +/* HMDC */ +#define SST_HMDC_HDDA0(x) (x << 0) +#define SST_HMDC_HDDA1(x) (x << 7) /* SST Vendor Defined Registers and bits */ diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 7b8ad9923fe4..0e1dde8c35e6 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -269,9 +269,9 @@ static void hsw_boot(struct sst_dsp *sst) SST_CSR2_SDFD_SSP1); /* enable DMA engine 0,1 all channels to access host memory */ - sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC, - SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff), - SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff)); + sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC, + SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff), + SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff)); /* disable all clock gating */ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); -- cgit v1.2.3-55-g7522 From d7d7d1eda0a2baaa4f6e02f0d58e81ea71dcbea2 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:08:18 +0800 Subject: ASoC: Intel: Add macros for SST shim register bits. Add some register definitions for other shim register bits. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 21a85eb196f1..3165dfa97408 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -77,6 +77,8 @@ #define SST_CSR_S0IOCS (0x1 << 21) #define SST_CSR_S1IOCS (0x1 << 23) #define SST_CSR_LPCS (0x1 << 31) +#define SST_CSR_24MHZ_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS) +#define SST_CSR_24MHZ_NO_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1) #define SST_BYT_CSR_RST (0x1 << 0) #define SST_BYT_CSR_VECTOR_SEL (0x1 << 1) #define SST_BYT_CSR_STALL (0x1 << 2) @@ -96,6 +98,14 @@ #define SST_IMRX_DONE (0x1 << 0) #define SST_BYT_IMRX_REQUEST (0x1 << 1) +/* IMRD / IMD */ +#define SST_IMRD_DONE (0x1 << 0) +#define SST_IMRD_BUSY (0x1 << 1) +#define SST_IMRD_SSP0 (0x1 << 16) +#define SST_IMRD_DMAC0 (0x1 << 21) +#define SST_IMRD_DMAC1 (0x1 << 22) +#define SST_IMRD_DMAC (SST_IMRD_DMAC0 | SST_IMRD_DMAC1) + /* IPCX / IPCC */ #define SST_IPCX_DONE (0x1 << 30) #define SST_IPCX_BUSY (0x1 << 31) @@ -125,6 +135,18 @@ /* HMDC */ #define SST_HMDC_HDDA0(x) (x << 0) #define SST_HMDC_HDDA1(x) (x << 7) +#define SST_HMDC_HDDA_E0_CH0 1 +#define SST_HMDC_HDDA_E0_CH1 2 +#define SST_HMDC_HDDA_E0_CH2 4 +#define SST_HMDC_HDDA_E0_CH3 8 +#define SST_HMDC_HDDA_E1_CH0 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0) +#define SST_HMDC_HDDA_E1_CH1 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1) +#define SST_HMDC_HDDA_E1_CH2 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2) +#define SST_HMDC_HDDA_E1_CH3 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3) +#define SST_HMDC_HDDA_E0_ALLCH (SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \ + SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3) +#define SST_HMDC_HDDA_E1_ALLCH (SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \ + SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3) /* SST Vendor Defined Registers and bits */ @@ -134,11 +156,16 @@ #define SST_VDRTCTL3 0xaC /* VDRTCTL0 */ +#define SST_VDRTCL0_APLLSE_MASK 1 #define SST_VDRTCL0_DSRAMPGE_SHIFT 16 #define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) #define SST_VDRTCL0_ISRAMPGE_SHIFT 6 #define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) +/* PMCS */ +#define SST_PMCS 0x84 +#define SST_PMCS_PS_MASK 0x3 + struct sst_dsp; /* -- cgit v1.2.3-55-g7522 From 81552612501f436f3824f056f95fdc04b8a60e1f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:09:47 +0800 Subject: ASoC: Intel: Add notification trace for reset. Add trace notification of IPC stream reset. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 96373ab46f8c..436c2fa23c80 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -687,7 +687,9 @@ static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg) switch (stream_msg) { case IPC_STR_STAGE_MESSAGE: case IPC_STR_NOTIFICATION: + break; case IPC_STR_RESET: + trace_ipc_notification("stream reset", stream->reply.stream_hw_id); break; case IPC_STR_PAUSE: stream->running = false; -- cgit v1.2.3-55-g7522 From 543ec637e00a9000772c315a8c98fa6ede563c5b Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:11:26 +0800 Subject: ASoC: Intel: Add debug to set DX state Add some debugging info to help with Dx state debug. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 436c2fa23c80..9825d195b8c9 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1646,7 +1646,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx) { u32 header, state_; - int ret; + int ret, item; header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE); state_ = state; @@ -1660,6 +1660,13 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, return ret; } + for (item = 0; item < dx->entries_no; item++) { + dev_dbg(hsw->dev, + "Item[%d] offset[%x] - size[%x] - source[%x]\n", + item, dx->mem_info[item].offset, + dx->mem_info[item].size, + dx->mem_info[item].source); + } dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", dx->entries_no, state); -- cgit v1.2.3-55-g7522 From 30bba67c43c0dd36696d1209c19afc3f25f2f3f3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:18:32 +0800 Subject: ASoC: Intel: Check ops before we derefference pointers. Check ops pointer members before we can derefference them. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 0b715b20a2d7..cd23060a0d86 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -224,19 +224,23 @@ EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64); void sst_dsp_dump(struct sst_dsp *sst) { - sst->ops->dump(sst); + if (sst->ops->dump) + sst->ops->dump(sst); } EXPORT_SYMBOL_GPL(sst_dsp_dump); void sst_dsp_reset(struct sst_dsp *sst) { - sst->ops->reset(sst); + if (sst->ops->reset) + sst->ops->reset(sst); } EXPORT_SYMBOL_GPL(sst_dsp_reset); int sst_dsp_boot(struct sst_dsp *sst) { - sst->ops->boot(sst); + if (sst->ops->boot) + sst->ops->boot(sst); + return 0; } EXPORT_SYMBOL_GPL(sst_dsp_boot); -- cgit v1.2.3-55-g7522 From 19a23a5d76e59f84caafea7a3299c23894ecad63 Mon Sep 17 00:00:00 2001 From: Subhransu S. Prusty Date: Wed, 30 Jul 2014 18:36:00 +0530 Subject: ASoC: Intel: mfld-pcm: Fix to use correct sst_data pointer Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7de87887d9f8..47df05ed3ac3 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -573,7 +573,7 @@ static int sst_platform_probe(struct platform_device *pdev) struct sst_platform_data *pdata = pdev->dev.platform_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); - if (sst == NULL) { + if (drv == NULL) { pr_err("kzalloc failed\n"); return -ENOMEM; } -- cgit v1.2.3-55-g7522 From 2741d43a1edd13c81a50ceb63f4edbf5fedb53ce Mon Sep 17 00:00:00 2001 From: Subhransu S. Prusty Date: Wed, 30 Jul 2014 18:39:05 +0530 Subject: ASoC: Intel: mfld-pcm: Allocate platform data Platform data may be null during platform_device_add. Allocate platform data before using. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 47df05ed3ac3..706212a6a68c 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -570,7 +570,7 @@ static int sst_platform_probe(struct platform_device *pdev) { struct sst_data *drv; int ret; - struct sst_platform_data *pdata = pdev->dev.platform_data; + struct sst_platform_data *pdata; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (drv == NULL) { @@ -578,6 +578,12 @@ static int sst_platform_probe(struct platform_device *pdev) return -ENOMEM; } + pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + if (pdata == NULL) { + pr_err("kzalloc failed for pdata\n"); + return -ENOMEM; + } + pdata->pdev_strm_map = dpcm_strm_map; pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map); drv->pdata = pdata; -- cgit v1.2.3-55-g7522 From 4ebd599e3570f392987df62f361d1742cc62f774 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Fri, 1 Aug 2014 22:54:19 +0800 Subject: ASoC: Intel: Add dependency to DW_DMAC for BDW platform Add dependency to DW_DMAC for broadwell machine, which have built in DW dma engines. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0b305f9da3db..f5b4a9c79cdf 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC select SND_SOC_INTEL_HASWELL select SND_COMPRESS_OFFLOAD select SND_SOC_RT286 -- cgit v1.2.3-55-g7522 From 97cfc751e1f2c300e093a9d2840aeee075db68d4 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 1 Aug 2014 23:08:38 +0800 Subject: ASoC: Intel: Delete message when IPC timeout occurs This fixes a bug where we dont delete the current message when an IPC message timeout occurs. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 9825d195b8c9..1ca71a283761 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -502,6 +502,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg, ipc_shim_dbg(hsw, "message timeout"); trace_ipc_error("error message timeout for", msg->header); + list_del(&msg->list); ret = -ETIMEDOUT; } else { -- cgit v1.2.3-55-g7522 From 94ce33456dbada5cb6ae1e10bd8895f034de731d Mon Sep 17 00:00:00 2001 From: Paweł Piskorski Date: Fri, 1 Aug 2014 23:09:44 +0800 Subject: ASoC: Intel: Don't issue ipc when processing response Make sure we dont issue IPC when we are processing a response. Signed-off-by: Paweł Piskorski Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 1ca71a283761..ae204a6e316b 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work) return; } - /* if the DSP is busy we will TX messages after IRQ */ + /* if the DSP is busy, we will TX messages after IRQ. + * also postpone if we are in the middle of procesing completion irq*/ ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX); - if (ipcx & SST_IPCX_BUSY) { + if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); return; } -- cgit v1.2.3-55-g7522 From d6e08617cb0fd7e5cd9effa6ba51dd00b06a0cf1 Mon Sep 17 00:00:00 2001 From: Paweł Piskorski Date: Fri, 1 Aug 2014 23:10:43 +0800 Subject: ASoC: Intel: update stream only on stream IPC msgs Only update the stream when the IPC message type matches stream type. Signed-off-by: Paweł Piskorski Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index ae204a6e316b..b6291516dbbf 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -782,7 +782,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) } /* update any stream states */ - hsw_stream_update(hsw, msg); + if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE) + hsw_stream_update(hsw, msg); /* wake up and return the error if we have waiters on this message ? */ list_del(&msg->list); -- cgit v1.2.3-55-g7522