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

[alsa-devel] [PATCH 1/2] ASoC: wm8960: correct the min gain value of some PGA
by Zidan Wang 10 Sep '15

10 Sep '15

The min gain is the corresponding gain value when the register value is 0 instead of 1, just correct it. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e3b7d0c..6163474 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -211,11 +211,11 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, return wm8960_set_deemph(codec); } -static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); -static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); -static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); static const struct snd_kcontrol_new wm8960_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, -- 1.9.1

3 10

[alsa-devel] [PATCH] ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform
by John Lin 10 Sep '15

10 Sep '15

Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4972bf3..8e51242 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3205,6 +3205,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, }, + { + .ident = "Google Ultima", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: intel: Fix SSP port configuration after RTD3 resume.
by Liam Girdwood 10 Sep '15

10 Sep '15

Currently the SSP port settings are being clobbered as part of the DSP RTD3 restore logic. make sure we save the correct params and restore them at resume. The FW sadly does not save SSP settings as part of the PM context. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f6efa9d..b27f25f 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -302,6 +302,10 @@ struct sst_hsw { struct sst_hsw_ipc_dx_reply dx; void *dx_context; dma_addr_t dx_context_paddr; + enum sst_hsw_device_id dx_dev; + enum sst_hsw_device_mclk dx_mclk; + enum sst_hsw_device_mode dx_mode; + u32 dx_clock_divider; /* boot */ wait_queue_head_t boot_wait; @@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, trace_ipc_request("set device config", dev); - config.ssp_interface = dev; - config.clock_frequency = mclk; - config.mode = mode; - config.clock_divider = clock_divider; + hsw->dx_dev = config.ssp_interface = dev; + hsw->dx_mclk = config.clock_frequency = mclk; + hsw->dx_mode = config.mode = mode; + hsw->dx_clock_divider = config.clock_divider = clock_divider; if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER) config.channels = 4; else @@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) return -EIO; } - /* Set ADSP SSP port settings */ - ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0, - SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, - SST_HSW_DEVICE_CLOCK_MASTER, 9); + /* Set ADSP SSP port settings - sadly the FW does not store SSP port + settings as part of the PM context. */ + ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk, + hsw->dx_mode, hsw->dx_clock_divider); if (ret < 0) dev_err(dev, "error: SSP re-initialization failed\n"); -- 2.1.4

2 1

[alsa-devel] [PATCH V2 1/1] ASoC: sti: check return of of_property_read
by Arnaud Pouliquen 10 Sep '15

10 Sep '15

Add check on of_property_read to return error when DT required property is not defined. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 14 ++++++++++---- sound/soc/sti/uniperif_reader.c | 6 +++++- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index f6eefe1..843f037 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(pnode, "version", &player->ver); - if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + if (of_property_read_u32(pnode, "version", &player->ver) || + player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { dev_err(dev, "Unknown uniperipheral version "); return -EINVAL; } @@ -998,10 +998,16 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) info->underflow_enabled = 1; - of_property_read_u32(pnode, "uniperiph-id", &info->id); + if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) { + dev_err(dev, "uniperipheral id not defined"); + return -EINVAL; + } /* Read the device mode property */ - of_property_read_string(pnode, "mode", &mode); + if (of_property_read_string(pnode, "mode", &mode)) { + dev_err(dev, "uniperipheral mode not defined"); + return -EINVAL; + } if (strcasecmp(mode, "hdmi") == 0) info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index c502626..f791239 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(node, "version", &reader->ver); + if (of_property_read_u32(node, "version", &reader->ver) || + reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + dev_err(&pdev->dev, "Unknown uniperipheral version "); + return -EINVAL; + } /* Save the info structure */ reader->info = info; -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] ASoC: make dapm cache search depth configurable
by Mark Brown 10 Sep '15

10 Sep '15

On Thu, Sep 10, 2015 at 10:42:50AM +0100, Nikesh wrote: > Hi Mark, As I'm fairly sure I've had to tell you before don't top post. This means that when someone sees something like this: > The reason I though this will be usefull is as below: while reading they know what the "this" you are talking about is. > the widgets in wm5110_dapm_widgets[]) as defined in this macro. But if you > notice the way DSP widgets are created they use a WM_ADSP2 macro which > creates a preloader widget for every DSP widget and we want to skip past > this preloader widget when doing a cache search so increasing the cache > search depth helps here. Why is the solution to this problem to add device specific tuning? Is this extremely expensive so no other device would want it?

1 0

[alsa-devel] [PATCH] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 10 Sep '15

10 Sep '15

From: Libin Yang <libin.yang(a)intel.com> When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- drivers/gpu/drm/i915/intel_audio.c | 56 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 11 ++++++++ 2 files changed, 58 insertions(+), 9 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index e3c32ce..a12bf57 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,8 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + for (i = 0; i < I915_MAX_PORTS; i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7..11999eb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,19 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + * The 4K resolution audio issue should be fixed on CNL + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- 1.9.1

1 0

[alsa-devel] [PATCH 0/9] ASoC: rsnd: Gen3 support
by Kuninori Morimoto 10 Sep '15

10 Sep '15

Hi Mark These are Gen3 support patches for Renesas sound IP. These are based on my previous cleanup patches [PATCH 0/4] ASoC: rsnd: cleanup for Gen3 support Date: Thu, 10 Sep 2015 15:44:13 +0900 Kuninori Morimoto (9): ASoC: rsnd: Add Gen3 initial support ASoC: rsnd: add rsnd_mod_get() macro and use it ASoC: rsnd: ADG uses mod base common method ASoC: rsnd: add common mod confirm method ASoC: rsnd: ADG initial setup on rsnd_adg_ssi_clk_init() ASoC: rsnd: tidyup ADG clock calculate method ASoC: rsnd: tidyup ADG debug message for clock selection ASoC: rsnd: adg: ignore undefined clock error ASoC: rsnd: add AUDIO_CLKOUT support Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 5 ++ sound/soc/sh/rcar/adg.c | 302 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----------------------- sound/soc/sh/rcar/core.c | 12 +++++ sound/soc/sh/rcar/ctu.c | 6 +-- sound/soc/sh/rcar/dvc.c | 6 +-- sound/soc/sh/rcar/mix.c | 6 +-- sound/soc/sh/rcar/rsnd.h | 12 +++++ sound/soc/sh/rcar/src.c | 6 +-- sound/soc/sh/rcar/ssi.c | 98 ++++++++++++++++++---------------- 9 files changed, 326 insertions(+), 127 deletions(-)

1 9

[alsa-devel] [PATCH 0/4] ASoC: rsnd: cleanup for Gen3 support
by Kuninori Morimoto 10 Sep '15

10 Sep '15

Hi Mark These are not so important, but cleanup for preparing of Gen3 support Kuninori Morimoto (4): ASoC: rsnd: tidyup rsnd_dma_to_xxx() position ASoC: rsnd: rename rsnd_src_pcm_new() to rsnd_src_pcm_new_gen2() ASoC: rsnd: remove unneeded sh_clk header ASoC: rsnd: add missing #sound-dai-cells explain on Document Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 2 ++ include/sound/rcar_snd.h | 1 - sound/soc/sh/rcar/adg.c | 1 - sound/soc/sh/rcar/rsnd.h | 3 +-- sound/soc/sh/rcar/src.c | 11 ++--------- 5 files changed, 5 insertions(+), 13 deletions(-)

1 4

[alsa-devel] [PATCH 0/2] ASoC: ak4642: MCKO support
by Kuninori Morimoto 10 Sep '15

10 Sep '15

Hi Mark These patches adds MCKO support for ak4642. Kuninori Morimoto (2): ASoC: ak4642: use *dev on ak4642_i2c_probe() ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT Documentation/devicetree/bindings/sound/ak4642.txt | 22 +++++++++++++++- sound/soc/codecs/ak4642.c | 152 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++-------------------------------------- 2 files changed, 120 insertions(+), 54 deletions(-) Best regards --- Kuninori Morimoto

1 2

[alsa-devel] [PATCH 1/1] ASoC: sti: fix compilation warnings
by Arnaud Pouliquen 09 Sep '15

09 Sep '15

Add check on of_property_read_string return to avoid compilation warning. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 32 +++++++++++++++++++------------- sound/soc/sti/uniperif_reader.c | 6 +++++- 2 files changed, 24 insertions(+), 14 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index f6eefe1..d90a3fb 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(pnode, "version", &player->ver); - if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + if ((of_property_read_u32(pnode, "version", &player->ver)) || + (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN)) { dev_err(dev, "Unknown uniperipheral version "); return -EINVAL; } @@ -998,19 +998,25 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) info->underflow_enabled = 1; - of_property_read_u32(pnode, "uniperiph-id", &info->id); + if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) { + dev_err(dev, "uniperipheral id not defined"); + return -EINVAL; + } /* Read the device mode property */ - of_property_read_string(pnode, "mode", &mode); - - if (strcasecmp(mode, "hdmi") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; - else if (strcasecmp(mode, "pcm") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM; - else if (strcasecmp(mode, "spdif") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF; - else - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE; + if (!of_property_read_string(pnode, "mode", &mode)) { + if (strcasecmp(mode, "hdmi") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; + else if (strcasecmp(mode, "pcm") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM; + else if (strcasecmp(mode, "spdif") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF; + else + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE; + } else { + dev_err(dev, "uniperipheral mode not defined"); + return -EINVAL; + } /* Save the info structure */ player->info = info; diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index c502626..7ddbd14 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(node, "version", &reader->ver); + if ((of_property_read_u32(node, "version", &reader->ver)) || + (reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN)) { + dev_err(&pdev->dev, "Unknown uniperipheral version "); + return -EINVAL; + } /* Save the info structure */ reader->info = info; -- 1.9.1

2 1

[alsa-devel] [PATCH lib 0/5] alsa-lib topology trivial fixes
by Takashi Iwai 09 Sep '15

09 Sep '15

Hi, here is a series of trivial fixes that have been spotted by gcc warnings. Takashi Takashi Iwai (5): topology: builder: Fix possibly uninitialized variable in write_elem_block() topology: ctl: Fix access type checks topology: data: Fix wrong size check in tplg_parse_data_hex() topology: parser: Add missing return value to snd_tplg_set_manifest_data() topology: pcm: Remove unused variables src/topology/builder.c | 9 ++++----- src/topology/ctl.c | 2 +- src/topology/data.c | 4 ++-- src/topology/parser.c | 1 + src/topology/pcm.c | 3 +-- 5 files changed, 9 insertions(+), 10 deletions(-) -- 2.5.1

2 9

[alsa-devel] [PATCH v3 0/9] asoc: Add audio for sti platforms
by Arnaud Pouliquen 09 Sep '15

09 Sep '15

version 3 : - patch dependency: ASoC: simple card: set cpu-dai sysclk with mclk-fs (Arnaud Pouliquen) - Remark not applied: Use Russell King's patch "sound/core: add IEC958 channel status helper". => This allows to create settings based on runtime, but not compatible with iec958 controls exposed to userland (settings overwritten by snd_pcm_create_iec958_consumer) - main corrections: - rework DAI CPU declaration to use generic dma - Supress abstraction layer for uniperiph player/reader - use dapm for DAC powerup/down version 2: corrections after first review - bindings: . suppress useless dts declaration - uniperif_player.c: . Suppress standby mode in uniplayer ( will be part of a dedicated patch) . Deliver IEC and clock adjustement controls in separate patch - uniperif_reader.c: . kept as separate file as behavior and feature not same than uniplayer . minor corrections - codec/sti-sas.c . implement DAPM for DAC . rework registers declaration to separate device and driver part version 1: - First patches submission Arnaud Pouliquen (9): ASoC: sti: add binding for ASoC driver ASoC: sti: Add uniperipheral header file ASoC: sti: Add CPU DAI driver for playback ASoC: sti: Add CPU DAI driver for capture ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: Codec: Add sti platform codec ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control .../devicetree/bindings/sound/st,sti-asoc-card.txt | 155 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sti-sas.c | 627 ++++++++++ sound/soc/sti/Kconfig | 11 + sound/soc/sti/Makefile | 4 + sound/soc/sti/sti_uniperif.c | 254 ++++ sound/soc/sti/uniperif.h | 1229 ++++++++++++++++++++ sound/soc/sti/uniperif_player.c | 1109 ++++++++++++++++++ sound/soc/sti/uniperif_reader.c | 364 ++++++ 12 files changed, 3762 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt create mode 100644 sound/soc/codecs/sti-sas.c create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_uniperif.c create mode 100644 sound/soc/sti/uniperif.h create mode 100644 sound/soc/sti/uniperif_player.c create mode 100644 sound/soc/sti/uniperif_reader.c -- 1.9.1

4 27

[alsa-devel] [PATCH] [alsa-utils.git] Fix a strange regression in aplay's '-d' option
by Grond 09 Sep '15

09 Sep '15

Hello. (J. Random Hacker here) While using aplay/arecord (version 1.0.28) to generate some noise, I ran across a regression in the use of the '-d' option. Specifically: According to the man page: -d, --duration=# Interrupt after # seconds. A value of zero means infinity. The default is zero, so if this option is omitted then the arecord process will run until it is killed. [snip] arecord -d 10 -f cd -t wav -D copy foobar.wav will record foobar.wav as a 10-second, CD-quality wave file, using the PCM "copy" (which might be defined in the user's .asoundrc file as: pcm.copy { type plug slave { pcm hw } route_policy copy } [snip] So the behavior I expect to see (and have experienced in the past) is: $ arecord -d 3 -f cd dump.wav Recording WAVE 'dump.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ ls -l -rw------- 1 grond66 grond66 529244 Sep 5 00:34 dump.wav What actually happens is: $ arecord -d 3 -f cd dump.wav Recording WAVE 'dump.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo [ program hangs here until SIGINT ] ^CAborted by signal Interrupt... arecord: pcm_read:2031: read error: Interrupted system call $ ls -l -rw------- 1 grond66 grond66 526444 Sep 5 00:39 dump-01.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-02.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-03.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-04.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-05.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-06.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-07.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-08.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-09.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-100.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-101.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-102.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-103.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-104.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-105.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-106.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-107.wav [ etcetera... ] $ aplay dump-01.wav [ three seconds of hiss, as expected ] $ less dump-02.wav RIFF$^@^@^@WAVEfmt ^P^@^@^@^A^@^B^@D<AC>^@^@^P<B1>^B^@^D^@^P^@data^@^@^@^@ $ So arecord is recording the requested audio and putting it into the first file, then spamming WAV headers into all the subsequent files. This is not terribly useful. The problem appears to be six lines of code in aplay/aplay.c, in the functions pcm_read() and pcm_readv(): 2006 static ssize_t pcm_read(u_char *data, size_t rcount) 2007 { 2008 ssize_t r; 2009 size_t result = 0; 2010 size_t count = rcount; 2011 )> 2012 if (count != chunk_size) { )> 2013 count = chunk_size; )> 2014 } 2015 2016 while (count > 0 && !in_aborting) { 2045 static ssize_t pcm_readv(u_char **data, unsigned int channels, size_t rcount) 2046 { 2047 ssize_t r; 2048 size_t result = 0; 2049 size_t count = rcount; 2050 )> 2051 if (count != chunk_size) { )> 2052 count = chunk_size; )> 2053 } 2054 2055 while (count > 0 && !in_aborting) { In this case, the 'rcount' arguments are the number of PCM integers to be read from the capture device. In theory. Because what lines 2012-2014 and 2051-2053 do is effectively: count = chunk_size; In other words, the argument is completely disregarded, and 'chunk_size' is used no matter what the functions are passed. But capture() and family use the following logic: "If pcm_read{,v}() returns with a value other than the number integers we told it to read off the wire, assume something went wrong": 2925 static void capture(char *orig_name) 2926 { 2927 int tostdout=0; /* boolean which describes output stream */ 2928 int filecount=0; /* number of files written */ 2929 char *name = orig_name; /* current filename */ 2930 char namebuf[PATH_MAX+1]; 2931 off64_t count, rest; /* number of bytes to capture */ [...] 2964 do { 2965 /* open a file to write */ 2966 if(!tostdout) { 2967 /* upon the second file we start the numbering scheme */ 2968 if (filecount || use_strftime) { 2969 filecount = new_capture_file(orig_name, namebuf, 2970 sizeof(namebuf), 2971 filecount); 2972 name = namebuf; 2973 } 2974 2975 /* open a new file */ 2976 remove(name); 2977 fd = safe_open(name); 2978 f (fd < 0) { 2979 perror(name); 2980 prg_exit(EXIT_FAILURE); 2981 } 2982 filecount++; 2983 } 2984 2985 rest = count; 2986 if (rest > fmt_rec_table[file_type].max_filesize) 2987 rest = fmt_rec_table[file_type].max_filesize; 2988 if (max_file_size && (rest > max_file_size)) 2989 rest = max_file_size; 2990 2991 /* setup sample header */ 2992 if (fmt_rec_table[file_type].start) 2993 fmt_rec_table[file_type].start(fd, rest); 2994 2995 /* capture */ 2996 fdcount = 0; 2997 while (rest > 0 && recycle_capture_file == 0 && !in_aborting) { )> 2998 size_t c = (rest <= (off64_t)chunk_bytes) ? )> 2999 (size_t)rest : chunk_bytes; )> 3000 size_t f = c * 8 / bits_per_frame; )> 3001 if (pcm_read(audiobuf, f) != f) )> 3002 break; 3003 if (write(fd, audiobuf, c) != c) { 3004 perror(name); 3005 prg_exit(EXIT_FAILURE); 3006 } 3007 count -= c; 3008 rest -= c; 3009 fdcount += c; 3010 } 3011 [...] 3027 /* repeat the loop when format is raw without timelimit or 3028 * requested counts of data are recorded 3029 */ 3030 } while ((file_type == FORMAT_RAW && !timelimit) || count > 0); 3031 } These two modes of operation are incompatible. However, if we nix the strange requirement that all calls to pcm_read{,v}() must be for exactly 'chunk_size' samples, the problem goes away. This is the purpose of the patch I've included below.

2 2

[alsa-devel] Getting notification of sound card load
by Mina Almasry 09 Sep '15

09 Sep '15

Hi, I'd like to get a notification when a new sound card becomes available in my system. What's the best way to do that? For example, when I 'cat /proc/asound/cards' I usually see one sound card available. But if I plug in USB speakers to my system and then run 'cat /proc/asound/cards', I see that the USB sound card is ready for the device I just plugged in. What's the best way via C API to get a notification that a new card has been loaded? Is there a better way that looking for changes on /proc/asound/cards? Thanks, Mina

2 2

[alsa-devel] [PATCH] ASoC: dapm: add kcontrol to switch regulator to regulated/bypass state
by Nikesh Oswal 08 Sep '15

08 Sep '15

When regulator is defined with SND_SOC_DAPM_REGULATOR_CONTROL_BYPASS flag, then a kcontrol will be created which can be used to switch regulator to regulated/bypass state. This will help to control the behaviour of the regulator based on a usecase. For example voice call may need a regulated voltage to acheive higher quality whereas voice trigger may need bypass voltage so as to save on power. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 7 +- sound/soc/soc-dapm.c | 177 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 183 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 15717b4..778b847 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -343,7 +343,12 @@ struct device; (e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)) /* regulator widget flags */ -#define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */ +/* bypass when disabled and regulated when enabled */ +#define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 +/* bypass when disabled and regulated when enable by default and a + kcontrol is created to explicitly switch between bypass/regulated */ +#define SND_SOC_DAPM_REGULATOR_CONTROL_BYPASS \ + (SND_SOC_DAPM_REGULATOR_BYPASS | 0x2) struct snd_soc_dapm_widget; enum snd_soc_dapm_type; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 36ab9cb..2d77eb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -965,6 +965,180 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return 0; } +static int snd_soc_dapm_regulator_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = + dapm_kcontrol_get_wlist(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_card *card = widget->dapm->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int new_val, val; + int ret; + bool bypass; + + if (item[0] >= e->items) + return -EINVAL; + + val = dapm_kcontrol_get_value(kcontrol); + new_val = item[0] == 0 ? SND_SOC_DAPM_REGULATOR_BYPASS : 0; + bypass = new_val == SND_SOC_DAPM_REGULATOR_BYPASS ? false : true; + + if (new_val != val) { + mutex_lock_nested(&card->dapm_mutex, + SND_SOC_DAPM_CLASS_RUNTIME); + if (regulator_is_enabled(widget->regulator)) { + ret = regulator_allow_bypass(widget->regulator, bypass); + if (ret != 0) + dev_warn(widget->dapm->dev, + "ASoC: Failed to change bypass %s: %d\n", + widget->name, ret); + } + dapm_kcontrol_set_value(kcontrol, new_val); + widget->on_val = new_val; + mutex_unlock(&card->dapm_mutex); + } + + return 0; +} + +static int snd_soc_dapm_regulator_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int val; + + val = dapm_kcontrol_get_value(kcontrol); + + if (val == SND_SOC_DAPM_REGULATOR_BYPASS) + ucontrol->value.enumerated.item[0] = 0; + else + ucontrol->value.enumerated.item[0] = 1; + + return 0; +} + +static const char * const dapm_regulator_texts[] = { + "Regulated", + "Bypass", +}; + +/* create new dapm regulator control */ +static int dapm_new_regulator(struct snd_soc_dapm_widget *w) +{ + int ret = 0; + struct snd_soc_card *card = w->dapm->card; + unsigned long private_value; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_path *path; + struct soc_enum regulator_enum[] = { + SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(dapm_regulator_texts), + dapm_regulator_texts), + }; + struct snd_kcontrol_new kcontrol_regulator[] = { + SOC_ENUM_EXT(NULL, regulator_enum[0], + snd_soc_dapm_regulator_get, + snd_soc_dapm_regulator_put), + }; + + + /* kcontrol creation is done only if client requests it */ + if (w->on_val != SND_SOC_DAPM_REGULATOR_CONTROL_BYPASS) + return 0; + + + /* create a kcontrol only if somebody is sourcing + from this regulator widget */ + if (list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])) { + dev_err(w->dapm->dev, "ASoC: %s has no sinks\n", w->name); + return -EINVAL; + } + + w->num_kcontrols = 1; + + private_value = (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_regulator[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + w->name); + ret = -ENOMEM; + goto err_out; + } + + kcontrol_regulator[0].private_value = private_value; + + w->kcontrol_news = devm_kmemdup(card->dev, &kcontrol_regulator[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!(w->kcontrol_news)) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + w->name); + ret = -ENOMEM; + goto err_private; + } + + + kcontrol = snd_soc_cnew(&w->kcontrol_news[0], NULL, + w->name, NULL); + + if (!kcontrol) { + ret = -ENOMEM; + goto err_kcontrol_news; + } + + kcontrol->private_free = dapm_kcontrol_free; + + ret = dapm_kcontrol_data_alloc(w, kcontrol); + if (ret) + goto err_kcontrol; + + + ret = snd_ctl_add(card->snd_card, kcontrol); + if (ret < 0) { + dev_err(w->dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[0].name, ret); + goto err_kcontrol; + } + + ret = dapm_kcontrol_add_widget(kcontrol, w); + if (ret) + goto err_kcontrol; + + + /* change on_val to remove the kcontrol creation bit + as kcontrol is already created */ + w->on_val = SND_SOC_DAPM_REGULATOR_BYPASS; + /* update the kcontrol value to reflect the initial value */ + dapm_kcontrol_set_value(kcontrol, w->on_val); + + w->kcontrols = kzalloc(w->num_kcontrols * + sizeof(struct snd_kcontrol *), + GFP_KERNEL); + if (!w->kcontrols) { + ret = -ENOMEM; + goto err_kcontrol; + } + + w->kcontrols[0] = kcontrol; + + snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_IN, path) { + if (path->name) + dapm_kcontrol_add_path(w->kcontrols[0], path); + } + + return 0; + +err_kcontrol: + snd_ctl_free_one(kcontrol); +err_kcontrol_news: + devm_kfree(card->dev, (void *)w->kcontrol_news); +err_private: + devm_kfree(card->dev, (void *)private_value); +err_out: + return ret; +} + /* create new dapm volume control */ static int dapm_new_pga(struct snd_soc_dapm_widget *w) { @@ -2922,6 +3096,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_dai_link: dapm_new_dai_link(w); break; + case snd_soc_dapm_regulator_supply: + dapm_new_regulator(w); + break; default: break; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: make dapm cache search depth configurable
by Nikesh Oswal 08 Sep '15

08 Sep '15

cache search depth will have a default value of 2 and can be modified by the respective component probe function Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 5 +++-- 3 files changed, 6 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5abba03..15717b4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -629,6 +629,7 @@ struct snd_soc_dapm_context { struct snd_soc_dapm_wcache path_sink_cache; struct snd_soc_dapm_wcache path_source_cache; + unsigned int cache_search_depth; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6173d15..1802883 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1595,6 +1595,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->dapm.bias_level = SND_SOC_BIAS_OFF; card->dapm.dev = card->dev; card->dapm.card = card; + card->dapm.cache_search_depth = 2; list_add(&card->dapm.list, &card->dapm_list); #ifdef CONFIG_DEBUG_FS @@ -2665,6 +2666,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->remove = component->driver->remove; dapm = &component->dapm; + dapm->cache_search_depth = 2; dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f4bf21a..36ab9cb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -580,11 +580,12 @@ dapm_wcache_lookup(struct snd_soc_dapm_wcache *wcache, const char *name) { struct snd_soc_dapm_widget *w = wcache->widget; struct list_head *wlist; - const int depth = 2; + int depth; int i = 0; - if (w) { + if (w && w->dapm->cache_search_depth) { wlist = &w->dapm->card->widgets; + depth = w->dapm->cache_search_depth; list_for_each_entry_from(w, wlist, list) { if (!strcmp(name, w->name)) -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] hda-emu: Fix hda_i915 build failure
by David Henningsson 07 Sep '15

07 Sep '15

This allows building hda-emu even after recent i915 -> hda patches. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/drm/i915_component.h | 1 + include/sound/hda_i915.h | 1 + 2 files changed, 2 insertions(+) create mode 100644 include/sound/hda_i915.h diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e69de29..545c915 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -0,0 +1 @@ +#include "../../dist/include/drm/i915_component.h" diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h new file mode 100644 index 0000000..1579430 --- /dev/null +++ b/include/sound/hda_i915.h @@ -0,0 +1 @@ +#include "../../dist/include/sound/hda_i915.h" -- 1.9.1

2 2

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Change internal PCM order
by johan＠oljud.se 07 Sep '15

07 Sep '15

From: Johan Rastén <johan(a)oljud.se> New PCMs will now be added to the end of the chip's PCM list instead of to the front. This changes the way streams are combined so that the first capture stream will now be merged with the first playback stream instead of the last. This fixes a problem with ASUS U7. Cards with one playback stream and cards without capture streams should be unaffected by this change. Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf Signed-off-by: Johan Rastén <johan(a)oljud.se> diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 310a382..01f5a86 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -377,8 +377,15 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, snd_usb_init_substream(as, stream, fp); - list_add(&as->list, &chip->pcm_list); + /* + * Keep using head insertion for M-Audio Audiophile USB (tm) which has a + * fix to swap capture stream order in conf/cards/USB-audio.conf + */ + if (chip->usb_id == USB_ID(0x0763, 0x2003)) + list_add(&as->list, &chip->pcm_list); + else + list_add_tail(&as->list, &chip->pcm_list); + chip->pcm_devs++; snd_usb_proc_pcm_format_add(as); -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Recurse before saving terminal properties
by Julian Scheel 06 Sep '15

06 Sep '15

When the descriptor parser recurses into the clock source descriptor, the terminals properties, like name and type, are set to the values of the clock selector as check_input_term overwrites term->name and others in the recursive call. As we want to only fill missing values using the descriptors higher in the recursion tree the properties taken from the actual descriptor must be applied after recursing. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/usb/mixer.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b3acba..4a49944 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -715,15 +715,16 @@ static int check_input_term(struct mixer_build *state, int id, term->name = d->iTerminal; } else { /* UAC_VERSION_2 */ struct uac2_input_terminal_descriptor *d = p1; - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le32_to_cpu(d->bmChannelConfig); - term->name = d->iTerminal; /* call recursively to get the clock selectors */ err = check_input_term(state, d->bCSourceID, term); if (err < 0) return err; + + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; } return 0; case UAC_FEATURE_UNIT: { -- 2.5.0

4 7

[alsa-devel] [PATCH] ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning
by Axel Lin 06 Sep '15

06 Sep '15

Fix below build warning: sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe': sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable] Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/au1x/psc-i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 38e853a..0bf9d62 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -296,7 +296,6 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) { struct resource *iores, *dmares; unsigned long sel; - int ret; struct au1xpsc_audio_data *wd; wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), -- 2.1.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Enable headphone jack detect on old Fujitsu laptops
by Takashi Iwai 05 Sep '15

05 Sep '15

According to the bug report, FSC Amilo laptops with ALC880 can detect the headphone jack but currently the driver disables it. It's partly intentionally, as non-working jack detect was reported in the past. Let's enable now. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102501 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 764dd5bba9ee..9b43cb95c047 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1135,7 +1135,7 @@ static const struct hda_fixup alc880_fixups[] = { /* override all pins as BIOS on old Amilo is broken */ .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x0121411f }, /* HP */ + { 0x14, 0x0121401f }, /* HP */ { 0x15, 0x99030120 }, /* speaker */ { 0x16, 0x99030130 }, /* bass speaker */ { 0x17, 0x411111f0 }, /* N/A */ @@ -1155,7 +1155,7 @@ static const struct hda_fixup alc880_fixups[] = { /* almost compatible with FUJITSU, but no bass and SPDIF */ .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x0121411f }, /* HP */ + { 0x14, 0x0121401f }, /* HP */ { 0x15, 0x99030120 }, /* speaker */ { 0x16, 0x411111f0 }, /* N/A */ { 0x17, 0x411111f0 }, /* N/A */ -- 2.5.1

1 1

[alsa-devel] No sound with Odys Winpad v10 - byt-rt5640
by Moritz 05 Sep '15

05 Sep '15

Hi, I'm having trouble getting sound to work and it looks like it's a bug in the driver. Full description: When loading the kernel-module for Bay Trail SoC TCO device (Version=3, TCOBASE=0x0460) the kernel throws a kernel trace depending on the firmware-image. First I provided the firmware-image from kernel.org: fw_sst_0f28.bin-48kHz_i2s_master With this image the kernel loads the firmware but no sound-devices are found. /proc/asound/cards reports no cards. dmesg just tells [ 11.196018] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered If I'm loading a firmware-image from Intel directly I get a sound-device in /proc but the module failes later with a kernel trace. The download-link to the intel-firmware: https://downloadcenter.intel.com/download/23903 The relevant parts of my dmesg for this problem: [ 11.120805] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [ 11.214831] iTCO_vendor_support: vendor-support=0 [ 11.215294] iTCO_wdt: Intel TCO WatchDog Timer Driver v1.11 [ 11.215391] iTCO_wdt: Found a Bay Trail SoC TCO device (Version=3, TCOBASE=0x0460) [ 11.215655] iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) [ 11.215997] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [ 11.232805] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.12.02 [ 11.232808] baytrail-pcm-audio baytrail-pcm-audio: Build type: 2 [ 11.232811] baytrail-pcm-audio baytrail-pcm-audio: Build date: Jan 16 2014 18:17:00 [ 11.259456] byt-rt5640 byt-rt5640: rt5640-aif1 <-> baytrail-pcm-audio mapping ok [ 11.970293] audit: type=1305 audit(1441466901.770:2): audit_pid=608 old=0 auid=4294967295 ses=4294967295 res=1 [ 12.291031] irq 6: nobody cared (try booting with the "irqpoll" option) [ 12.294817] CPU: 0 PID: 0 Comm: swapper/0 Tainted: G O 4.2.0-4-desktop #1 [ 12.294822] Hardware name: AXDIA International GmbH WINPAD V10/Type2 - Board Product Name, BIOS AD1012.1.03.017 11/21/2014 [ 12.294826] 00000000 00000000 f582df50 c080ecdd f5b3f500 f582df70 c02abab9 c09c9bfc [ 12.294839] 00000006 00000000 f582df6c f5b3f500 00000006 f582df94 c02abf49 f582df94 [ 12.294850] c05f400c fffb910a 00000000 f5b3f500 f5b3f570 00000000 f582dfd0 c02a9a53 [ 12.294861] Call Trace: [ 12.294874] [<c080ecdd>] dump_stack+0x48/0x69 [ 12.294882] [<c02abab9>] __report_bad_irq+0x29/0xd0 [ 12.294888] [<c02abf49>] note_interrupt+0x1c9/0x210 [ 12.294896] [<c05f400c>] ? add_interrupt_randomness+0x15c/0x190 [ 12.294902] [<c02a9a53>] handle_irq_event_percpu+0xa3/0x1c0 [ 12.294908] [<c02a9ba3>] handle_irq_event+0x33/0x50 [ 12.294914] [<c02ac3b5>] handle_fasteoi_irq+0x75/0x130 [ 12.294919] [<c02ac340>] ? handle_simple_irq+0x70/0x70 [ 12.294925] [<c0204b11>] handle_irq+0x71/0x90 [ 12.294929] <IRQ> [<c081603c>] do_IRQ+0x3c/0xd0 [ 12.294941] [<c08155b3>] common_interrupt+0x33/0x38 [ 12.294948] [<c06e753f>] ? cpuidle_enter_state+0xaf/0x2e0 [ 12.294954] [<c06e77a4>] cpuidle_enter+0x14/0x20 [ 12.294961] [<c02923f3>] call_cpuidle+0x33/0x70 [ 12.294967] [<c029268e>] cpu_startup_entry+0x25e/0x320 [ 12.294973] [<c080743b>] rest_init+0x7b/0x80 [ 12.294980] [<c0b34cd8>] start_kernel+0x456/0x45d [ 12.294986] [<c0b3464e>] ? set_init_arg+0x52/0x52 [ 12.294992] [<c0b3438c>] i386_start_kernel+0x13a/0x13d [ 12.294995] handlers: [ 12.298499] [<f8725c80>] sst_byt_irq [snd_soc_sst_baytrail_pcm] threaded [<f8724360>] sst_byt_irq_thread [snd_soc_sst_baytrail_pcm] [ 12.302073] Disabling IRQ #6 with my limited understanding it looks like the irq isn't handled correctly. System information: Linux klappstulle.XXX 4.2.0-4-desktop #1 SMP PREEMPT Mon Aug 31 12:34:37 UTC 2015 (efc468a) i686 i686 i386 GNU/Linux Modules Loaded fuse dm_crypt drbg ansi_cprng algif_skcipher af_alg loop nf_log_ipv6 af_packet xt_pkttype nf_log_ipv4 nf_log_common xt_LOG xt_limit ip6t_REJECT xt_tcpudp nf_conntrack_ipv6 nf_defrag_ipv6 ip6table_raw ipt_REJECT iptable_raw xt_CT iptable_filter ip6table_mangle nf_conntrack_netbios_ns nf_conntrack_broadcast nf_conntrack_ipv4 nf_defrag_ipv4 ip_tables xt_conntrack nf_conntrack ip6table_filter ip6_tables x_tables xfs libcrc32c nls_iso8859_1 nls_cp437 vfat fat joydev hid_multitouch iTCO_wdt iTCO_vendor_support r8723bs snd_soc_sst_baytrail_pcm snd_soc_sst_ipc snd_soc_sst_dsp dw_dmac_core snd_soc_sst_byt_rt5640_mach intel_powerclamp coretemp kvm_intel kvm crc32_pclmul aesni_intel aes_i586 xts lrw gf128mul ablk_helper cryptd pcspkr efivars cfg80211 snd_intel_sst_acpi snd_intel_sst_core snd_soc_rt5640 snd_soc_sst_mfld_platform snd_soc_rl6231 snd_soc_core snd_compress lpc_ich mfd_core snd_pcm thermal mei_txe xhci_pci mei rfkill_gpio xhci_hcd 8250_fintek jsa1212 industrialio i2c_hid regmap_i2c battery rfkill snd_timer snd acpi_pad int3402_thermal int3403_thermal processor_thermal_device int340x_thermal_zone intel_soc_dts_iosf ac int3400_thermal acpi_thermal_rel soundcore ac97_bus snd_soc_sst_acpi i2c_designware_platform i2c_designware_core 8250_dw pwm_lpss_platform pwm_lpss processor dm_mod efivarfs btrfs xor raid6_pq mmc_block crc32c_intel i915 i2c_algo_bit drm_kms_helper drm button video sdhci_acpi sdhci mmc_core sg Package versions gcc-4.8-7.1.2.i586 make-4.0-2.2.3.i586 binutils-2.24-6.1.7.i586 util-linux-2.25.1-17.1.i586 mount from util-linux 2.25.1 (libmount 2.25.0: selinux, assert, debug) glibc-2.19-16.15.1.i686 linux-glibc-devel-3.16-2.1.7.noarch /proc/cmdline BOOT_IMAGE=/boot/vmlinuz-4.2.0-4-desktop root=UUID=fc064bad-46f0-4947-9364-3a7864bb0b6b ro nomodeset \ sdhci.debug_quirks=0x8000 quiet showopts /proc/cpuinfo processor : *** 0 - 3 *** vendor_id : GenuineIntel cpu family : 6 model : 55 model name : Intel(R) Atom(TM) CPU Z3735F @ 1.33GHz stepping : 8 microcode : 0x82a cpu MHz : 1751.642 cache size : 1024 KB physical id : 0 siblings : 4 core id : 0 cpu cores : 4 apicid : 0 initial apicid : 0 fdiv_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 11 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx rdtscp lm constant_tsc arch_perfmon pebs bts xtopology nonstop_tsc aperfmperf pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr pdcm sse4_1 sse4_2 movbe popcnt tsc_deadline_timer aes rdrand lahf_lm 3dnowprefetch ida arat epb dtherm tpr_shadow vnmi flexpriority ept vpid tsc_adjust smep erms bugs : bogomips : 2662.40 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: /proc/asound/cards 0 [bytrt5640 ]: byt-rt5640 - byt-rt5640 byt-rt5640 /proc/asound/devices 2: [ 0] : control 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 33: : timer Please see attached for full dmesg. Kind regards Moritz

1 0

[alsa-devel] [RFC PATCH (alsa-lib)] pcm: Modify check condition in snd_pcm_sw_params_set_avail_min
by Koro Chen 04 Sep '15

04 Sep '15

When we use a ping-ping buffer in capture, and if hw_ptr reported at IRQ is a little smaller than period_size: |xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------| hw_ptr < period_size This available buffer will not be read since its size is smaller than avail_min (which is set to be period_size), and read thread continues to sleep. If the next hw_ptr is just a little larger than buffer_size, overrun occurs. This could be resolved by setting a small avail_min to kernel, for example, 1, so read thread wakes up and reads every data at IRQ. But current alsa-lib only allows avail_min to be at least period_size. Remove the constraint and only check for zero case. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- src/pcm/pcm.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index f5fc728..8492689 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -5958,12 +5958,8 @@ int snd_pcm_sw_params_set_avail_min(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, #endif { assert(pcm && params); - /* Fix avail_min if it's below period size. The period_size - * defines the minimal wake-up timing accuracy, so it doesn't - * make sense to set below that. - */ - if (val < pcm->period_size) - val = pcm->period_size; + if (!val) + val = 1; params->avail_min = val; return 0; } -- 1.7.9.5

4 7

[alsa-devel] Tascam US-366 and Alsa Firmware
by Burton Samograd 04 Sep '15

04 Sep '15

Hi, I recently bought the Tascam US-366 and forgot to check for Linux support, which it doesn't have (yet! :). Looking through the Alsa firmware sources, there seems to be a firmware loader called usx2yloader which supports a number of Tascam cards, but the US-366 is not listed. It looks like really usx2yloader just needs the firmware and a `prepad` file. I`ve found the firmware for the card (I think) buried in the OSX firmware download package: http://tascam.com/content/downloads/products/789/us-322_us-366_fw_v103_mave… Now I think I just need the prepad file, but there's little documentation regarding exactly what that is (I'm assuming a number of 0s just to pad the firmware image to alignment, but I'm not sure of what alignment). Any ideas if usx2yloader will work at all with the US-366? Does anyone know what the pre-pad file should be/contain? Has anyone tried to get the US-366 working yet and failed? Reference: http://tascam.com/product/us-366/ Thanks. Burton Samograd

2 2

[alsa-devel] reconfigure cpu dai after resume from suspend
by maruthi srinivas 04 Sep '15

04 Sep '15

Hi, I came across a situation, where I need to reconfigure cpu dai ( similar to cpu dai's hw_params()) after system resumed from suspend. Should this be taken care by soc core or do I need to call the same logic(hw_params()) inside snd_soc_dai_driver's resume() ? Regards, Maruthi

1 0

[alsa-devel] [PATCH] drm/i915: Improve kernel-doc for i915_audio_component struct
by David Henningsson 04 Sep '15

04 Sep '15

To make kernel-doc happy, the i915_audio_component_audio_ops struct cannot be nested. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Note that I didn't do the same un-nesting for i915_audio_component_ops. This is to make it easier to merge the pending sync_audio_rate patch set. It applies on top of my just sent patches so should probably be taken through Takashi's tree. Documentation/DocBook/drm.tmpl | 1 + include/drm/i915_component.h | 33 +++++++++++++++++++++++---------- 2 files changed, 24 insertions(+), 10 deletions(-) diff --git a/Documentation/DocBook/drm.tmpl b/Documentation/DocBook/drm.tmpl index 2fb9a54..7554679 100644 --- a/Documentation/DocBook/drm.tmpl +++ b/Documentation/DocBook/drm.tmpl @@ -4048,6 +4048,7 @@ int num_ioctls;</synopsis> <title>High Definition Audio</title> !Pdrivers/gpu/drm/i915/intel_audio.c High Definition Audio over HDMI and Display Port !Idrivers/gpu/drm/i915/intel_audio.c +!Iinclude/drm/i915_component.h </sect2> <sect2> <title>Panel Self Refresh PSR (PSR/SRD)</title> diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index b2d56dd..89d6362 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,6 +24,28 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/** + * struct i915_audio_component_audio_ops - i915 directly calling hda driver + * + * @audio_ptr: Pointer to pass when calling pin_eld_notify. + * @pin_eld_notify: Called from i915 driver, notifying the HDA driver that + * pin sense and/or ELD information has changed. + * + * These functions are implemented by hda driver and called by the i915 + * driver. + */ +struct i915_audio_component_audio_ops { + void *audio_ptr; + void (*pin_eld_notify)(void *audio_ptr, int port); +}; + +/** + * struct i915_audio_component - Used for direct communication between i915 and hda drivers + * + * @dev: i915 device, used as parameter for ops + * @ops: Ops implemented by i915 driver, called by hda driver + * @audio_ops: Ops implemented by hda driver, called by i915 driver + */ struct i915_audio_component { struct device *dev; @@ -35,16 +57,7 @@ struct i915_audio_component { int (*get_cdclk_freq)(struct device *); } *ops; - const struct i915_audio_component_audio_ops { - void *audio_ptr; - /** - * Call from i915 driver, notifying the HDA driver that - * pin sense and/or ELD information has changed. - * @audio_ptr: HDA driver object - * @port: Which port has changed (PORTA / PORTB / PORTC etc) - */ - void (*pin_eld_notify)(void *audio_ptr, int port); - } *audio_ops; + const struct i915_audio_component_audio_ops *audio_ops; }; #endif /* _I915_COMPONENT_H_ */ -- 1.9.1

3 3

[alsa-devel] [PATCH] ALSA: sparc: amd7930: Fix module autoload for OF platform driver
by Luis de Bethencourt 04 Sep '15

04 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- Hello, This patch adds the missing MODULE_DEVICE_TABLE() for OF to eport that information so modules have the correct aliases built-in and autoloading works correctly. A longer explanation by Javier Canillas can be found here: https://lkml.org/lkml/2015/7/30/519 Thanks, Luis sound/sparc/amd7930.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 784ceb8..35c1f6a 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1064,6 +1064,7 @@ static const struct of_device_id amd7930_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, amd7930_match); static struct platform_driver amd7930_sbus_driver = { .driver = { -- 2.4.6

2 1

[alsa-devel] [PATCH] ALSA: hda - Add some FIXUP quirks for white noise on Dell laptop.
by woodrow.shen＠canonical.com 04 Sep '15

04 Sep '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> Dell laptop has a series model to use the same codec but different subsystem ID. At the same time they happens the white noise by login screen and headphone; for fixing them together, I only can add these IDs to FIXUP function ALC292_FIXUP_DISABLE_AAMIX, then try to solve such the similar issues. Codec: Realtek ALC3235 Vendor Id: 0x10ec0293 Subsystem Id: 0x102806dd Subsystem Id: 0x102806df Subsystem Id: 0x102806e0 Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1492132 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4e6b090..764dd5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5189,8 +5189,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.1.4

2 1

[alsa-devel] [PATCH v3 0/9] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 04 Sep '15

04 Sep '15

The purpose of the DMAFLUSHP instruction: - Tell the peripheral to clear its status and control registers. - Send a message to the peripheral to resend its level status. There are 3 timings described in PL330 Technical Reference Manual: - Timing 1: Burst request, can work well without DMAFLUSHP. - Timing 2: Single and burst request, DMAC will ignore the single transfer request. This timing happens if there are single and burst request. - Timing 3: Single transfers for a burst request, DMAC should signals datype to request the peripheral to flush the contents of any control registers. This timing happens if there is not enough MFIFO to places the burst data. A peripheral may signal a DMA request during the execution of DMAFLUSHP instruction, that cause DMA request being ignored by DMAC. But DMAC and all peripherals on RK3X SoCs DO NOT support DMAFLUSHP. It can't send a message to the peripheral to resend DMA request, and the peripheral can't acknowledge a flush request from DMAC. So all DMA requests should NOT be ignored by DMAC, and DMAC will not notify the peripheral to flush. To fix this problem, we need: - Do NOT execute DMAFLUSHP instruction. - Timing 2 and timing 3 should not happen. Because on RK3X SoCs, there are 6 or below channels and 32 MFIFO depth for DMAC_BUS, and 8 channels and 64 MFIFO depth for DMAC_PERI, it is impossible to hit the timing 3 if burst length is equal or less than 4. Since the request type signal by the peripheral can only be set by software. We can set Rockchip Soc's GRF_PERIDMAC_CON0[2:1] to select single or burst request, if it is set b01, all of the peripharals will signal a brust request. So the timing 2 will not happen, too. So DMAC on RK3X can support single or burst transfer, but can't support mixed transfer. Because burst transfer is more efficient than single transfer, this is confirmed by our ASIC team, who strongly suggest to use burst transfer. And this is confirmed by Addy's test on RK3288-Pink2 board, the speed of spi flash burst transfer will increase about two times than single transfer. Also, I have tested dw_mmc with pl330 on RK3188 platform to double confirm the result. That means burst transfer is reansonable. So we need a quirk not to execute DMAFLUSHP instruction and to use burst transfer. Note: - The Rockchip Soc default value of GRF_PERIDMAC_CON0[2:1] is b01. To support brust transfer, these bits should not be changed in bootloader. Changes in v3: - add more rockchip drivers' changes in this patchset - add Reviewed-by: Sonny Rao <sonnyrao(a)chromium.org> Changes in v2: - amend the author - reorder the patches suggested by Doug - add Reviewed-by: Doug Anderson <dianders(a)chromium.org> for rk3288.dtsi patch and arm-pl330.txt patch - amend Olof's mail address Changes in v1: - rename broken-no-flushp to "arm,pl330-broken-no-flushp" suggested by Krzysztof. - add From original author. - remove Sunny's tag Addy Ke (3): DMA: pl330: add quirk for broken no flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3288 platform spi: rockchip: modify DMA max burst to 1 Boojin Kim (1): DMA: pl330: support burst mode for dev-to-mem and mem-to-dev transmit Shawn Lin (4): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp Documentation: spi-rockchip: add description of rockchip,spi-broken-burst-len Documentation: rockchip-i2s: add description of rockchip,i2s-broken-burst-len ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + .../devicetree/bindings/sound/rockchip-i2s.txt | 4 + .../devicetree/bindings/spi/spi-rockchip.txt | 3 +- arch/arm/boot/dts/rk3288.dtsi | 7 ++ arch/arm/boot/dts/rk3xxx.dtsi | 5 + drivers/dma/pl330.c | 101 +++++++++++++++------ drivers/spi/spi-rockchip.c | 16 +++- sound/soc/rockchip/rockchip_i2s.c | 10 +- 8 files changed, 113 insertions(+), 34 deletions(-) -- 2.3.7

2 13

[alsa-devel] [PATCH 00/12] ASoC: Fix module autoload for OF platform drivers
by Luis de Bethencourt 03 Sep '15

03 Sep '15

Hello, This series adds the missing MODULE_DEVICE_TABLE() for OF to eport that information so modules have the correct aliases built-in and autoloading works correctly. A longer explanation by Javier Canillas can be found here: https://lkml.org/lkml/2015/7/30/519 Thanks, Luis Luis de Bethencourt (12): ASoC: atmel_wm8904: Fix module autoload for OF platform driver ASoC: da9055: Fix module autoload for OF platform driver ASoC: wm8510: Fix module autoload for OF platform driver ASoC: wm8580: Fix module autoload for OF platform driver ASoC: wm8523: Fix module autoload for OF platform driver ASoC: fsl-asoc-card: Fix module autoload for OF platform driver ASoC: fsl_sai: Fix module autoload for OF platform driver ASoC: jz4740: Fix module autoload for OF platform driver ASoC: kirkwood: Fix module autoload for OF platform driver ASoC: pxa: Fix module autoload for OF platform drivers ASoC: mop500: Fix module autoload for OF platform driver ASoC: ux500: Fix module autoload for OF platform driver sound/soc/atmel/atmel_wm8904.c | 1 + sound/soc/codecs/da9055.c | 1 + sound/soc/codecs/wm8510.c | 1 + sound/soc/codecs/wm8523.c | 1 + sound/soc/codecs/wm8580.c | 1 + sound/soc/fsl/fsl-asoc-card.c | 1 + sound/soc/fsl/fsl_sai.c | 1 + sound/soc/jz4740/jz4740-i2s.c | 1 + sound/soc/kirkwood/armada-370-db.c | 1 + sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-pcm.c | 1 + sound/soc/ux500/mop500.c | 1 + sound/soc/ux500/ux500_msp_dai.c | 1 + 13 files changed, 13 insertions(+) -- 2.4.6

2 2

[alsa-devel] [PATCH 02/12] ASoC: da9055: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/codecs/da9055.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 66bb446..ab39de6 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1533,6 +1533,7 @@ static const struct of_device_id da9055_of_match[] = { { .compatible = "dlg,da9055-codec", }, { } }; +MODULE_DEVICE_TABLE(of, da9055_of_match); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { -- 2.4.6

2 1

[alsa-devel] [PATCH 12/12] ASoC: ux500: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/ux500/ux500_msp_dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 978f2d7..a57a9b8 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -852,6 +852,7 @@ static const struct of_device_id ux500_msp_i2s_match[] = { { .compatible = "stericsson,ux500-msp-i2s", }, {}, }; +MODULE_DEVICE_TABLE(of, ux500_msp_i2s_match); static struct platform_driver msp_i2s_driver = { .driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 11/12] ASoC: mop500: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/ux500/mop500.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 4e0c0e5..ba9fc09 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -152,6 +152,7 @@ static const struct of_device_id snd_soc_mop500_match[] = { { .compatible = "stericsson,snd-soc-mop500", }, {}, }; +MODULE_DEVICE_TABLE(of, snd_soc_mop500_match); static struct platform_driver snd_soc_mop500_driver = { .driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 10/12] ASoC: pxa: Fix module autoload for OF platform drivers
by Luis de Bethencourt 03 Sep '15

03 Sep '15

These platform drivers have a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-pcm.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fbe2e93..c254a75 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -809,6 +809,7 @@ static const struct of_device_id pxa_ssp_of_ids[] = { { .compatible = "mrvl,pxa-ssp-dai" }, {} }; +MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids); #endif static int asoc_ssp_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index a51c9da..f1aea95 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -138,6 +138,7 @@ static const struct of_device_id snd_soc_pxa_audio_match[] = { { .compatible = "mrvl,pxa-pcm-audio" }, { } }; +MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match); #endif static struct platform_driver pxa_pcm_driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 09/12] ASoC: kirkwood: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/kirkwood/armada-370-db.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c index de7563b..e0304d5 100644 --- a/sound/soc/kirkwood/armada-370-db.c +++ b/sound/soc/kirkwood/armada-370-db.c @@ -130,6 +130,7 @@ static const struct of_device_id a370db_dt_ids[] = { { .compatible = "marvell,a370db-audio" }, { }, }; +MODULE_DEVICE_TABLE(of, a370db_dt_ids); static struct platform_driver a370db_driver = { .driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 08/12] ASoC: jz4740: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/jz4740/jz4740-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index b05fb1c..794a349 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -485,6 +485,7 @@ static const struct of_device_id jz4740_of_matches[] = { { .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 }, { /* sentinel */ } }; +MODULE_DEVICE_TABLE(of, jz4740_of_matches); #endif static int jz4740_i2s_dev_probe(struct platform_device *pdev) -- 2.4.6

1 0

[alsa-devel] [PATCH 07/12] ASoC: fsl_sai: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c73bea..ec14427 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -801,6 +801,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx6sx-sai", }, { /* sentinel */ } }; +MODULE_DEVICE_TABLE(of, fsl_sai_ids); static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, -- 2.4.6

1 0

[alsa-devel] [PATCH 06/12] ASoC: fsl-asoc-card: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..ca8b469 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -580,6 +580,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8962", }, {} }; +MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); static struct platform_driver fsl_asoc_card_driver = { .probe = fsl_asoc_card_probe, -- 2.4.6

1 0

[alsa-devel] [PATCH 05/12] ASoC: wm8523: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/codecs/wm8523.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 43ea8ae..483ec72 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -430,6 +430,7 @@ static const struct of_device_id wm8523_of_match[] = { { .compatible = "wlf,wm8523" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8523_of_match); static const struct regmap_config wm8523_regmap = { .reg_bits = 8, -- 2.4.6

1 0

[alsa-devel] [PATCH 04/12] ASoC: wm8580: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/codecs/wm8580.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 759a792..a9f3900 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -916,6 +916,7 @@ static const struct of_device_id wm8580_of_match[] = { { .compatible = "wlf,wm8580" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8580_of_match); static const struct regmap_config wm8580_regmap = { .reg_bits = 7, -- 2.4.6

1 0

[alsa-devel] [PATCH 03/12] ASoC: wm8510: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/codecs/wm8510.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index dac5beb..0617415 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -598,6 +598,7 @@ static const struct of_device_id wm8510_of_match[] = { { .compatible = "wlf,wm8510" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8510_of_match); static const struct regmap_config wm8510_regmap = { .reg_bits = 7, -- 2.4.6

1 0

[alsa-devel] [PATCH 01/12] ASoC: atmel_wm8904: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/atmel/atmel_wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index aa354e1..1933bcd 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -176,6 +176,7 @@ static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { { .compatible = "atmel,asoc-wm8904", }, { } }; +MODULE_DEVICE_TABLE(of, atmel_asoc_wm8904_dt_ids); #endif static struct platform_driver atmel_asoc_wm8904_driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 0/2] Update v4 -> v5 for i915 -> HDA patch set
by David Henningsson 03 Sep '15

03 Sep '15

Takashi mistakenly merged v4 instead of v5 of the patch set. This updates the code changes. (Not the docbook, I'm still fighting to get that to look right) David Henningsson (2): drm/i915: Drop port_mst_index parameter from pin/eld callback drm/i915: Add locks around audio component bind/unbind drivers/gpu/drm/i915/intel_audio.c | 8 ++++++-- include/drm/i915_component.h | 3 +-- 2 files changed, 7 insertions(+), 4 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH v2 1/1] ASoC: fsl_sai: Add missing MODULE_DEVICE_TABLE
by Alexander Stein 03 Sep '15

03 Sep '15

This enables module auto-loading when using device tree. Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v2: * Added "Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com>" sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c73bea..ec14427 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -801,6 +801,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx6sx-sai", }, { /* sentinel */ } }; +MODULE_DEVICE_TABLE(of, fsl_sai_ids); static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, -- 2.4.6

1 0

[alsa-devel] Linux 4.2 ALSA snd-usb-audio inconsistent lock state warn in PCM nonatomic mode
by Shuah Khan 02 Sep '15

02 Sep '15

Hi Takashi, I am seeing the following inconsistent lock state warning when PCM is run in nonatomic mode. This is on 4.2.0 and with the following change to force PCM on nonatomic mode: diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 310a382..16bbb71 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -370,6 +370,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, pcm->private_data = as; pcm->private_free = snd_usb_audio_pcm_free; pcm->info_flags = 0; + pcm->nonatomic = true; if (chip->pcm_devs > 0) sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs); else The device Bus 003 Device 002: ID 2040:7200 Hauppauge Please let me know if you need more information and any ideas on how to fix this problem. thanks, -- Shuah [ 120.283960] ================================= [ 120.283964] [ INFO: inconsistent lock state ] [ 120.283968] 4.2.0+ #29 Not tainted [ 120.283972] --------------------------------- [ 120.283975] inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage. [ 120.283980] swapper/1/0 [HC1[1]:SC0[0]:HE0:SE1] takes: [ 120.283983] (&(&subs->lock)->rlock){?.+...}, at: [<ffffffffa05ed210>] retire_capture_urb+0x140/0x2b0 [snd_usb_audio] [ 120.284005] {HARDIRQ-ON-W} state was registered at: [ 120.284008] [<ffffffff810af230>] __lock_acquire+0xc50/0x2380 [ 120.284016] [<ffffffff810b1631>] lock_acquire+0xb1/0x130 [ 120.284022] [<ffffffff817fd4a1>] _raw_spin_lock+0x31/0x40 [ 120.284028] [<ffffffffa05ee81d>] snd_usb_pcm_pointer+0x5d/0xc0 [snd_usb_audio] [ 120.284040] [<ffffffffa01c6498>] snd_pcm_update_hw_ptr0+0x38/0x3a0 [snd_pcm] [ 120.284052] [<ffffffffa01c73f0>] snd_pcm_update_hw_ptr+0x10/0x20 [snd_pcm] [ 120.284063] [<ffffffffa01beae5>] snd_pcm_hwsync+0x45/0xa0 [snd_pcm] [ 120.284071] [<ffffffffa01c1347>] snd_pcm_common_ioctl1+0x277/0xce0 [snd_pcm] [ 120.284081] [<ffffffffa01c227e>] snd_pcm_capture_ioctl1+0x1be/0x2d0 [snd_pcm] [ 120.284090] [<ffffffffa01c2444>] snd_pcm_capture_ioctl+0x34/0x40 [snd_pcm] [ 120.284100] [<ffffffff811ed7e1>] do_vfs_ioctl+0x301/0x560 [ 120.284107] [<ffffffff811edab9>] SyS_ioctl+0x79/0x90 [ 120.284112] [<ffffffff817fdf17>] entry_SYSCALL_64_fastpath+0x12/0x6f [ 120.284119] irq event stamp: 823304 [ 120.284122] hardirqs last enabled at (823301): [<ffffffff816707ad>] cpuidle_enter_state+0xed/0x230 [ 120.284129] hardirqs last disabled at (823302): [<ffffffff817fea28>] common_interrupt+0x68/0x6d [ 120.284135] softirqs last enabled at (823304): [<ffffffff81060351>] _local_bh_enable+0x21/0x50 [ 120.284139] softirqs last disabled at (823303): [<ffffffff810611ac>] irq_enter+0x4c/0x70 [ 120.284143] other info that might help us debug this: [ 120.284146] Possible unsafe locking scenario: [ 120.284149] CPU0 [ 120.284150] ---- [ 120.284152] lock(&(&subs->lock)->rlock); [ 120.284155] <Interrupt> [ 120.284157] lock(&(&subs->lock)->rlock); [ 120.284160] *** DEADLOCK *** [ 120.284163] no locks held by swapper/1/0. [ 120.284165] stack backtrace: [ 120.284170] CPU: 1 PID: 0 Comm: swapper/1 Not tainted 4.2.0+ #29 [ 120.284173] Hardware name: Hewlett-Packard HP ProBook 6475b/180F, BIOS 68TTU Ver. F.04 08/03/2012 [ 120.284176] ffffffff828d5630 ffff88023ec83a68 ffffffff817f4ea0 0000000000000007 [ 120.284181] ffff880235a6a500 ffff88023ec83ac8 ffffffff810ad97f 0000000000000000 [ 120.284186] 0000000000000000 ffff880200000001 ffffffff810134df ffffffff827c5390 [ 120.284192] Call Trace: [ 120.284194] <IRQ> [<ffffffff817f4ea0>] dump_stack+0x45/0x57 [ 120.284203] [<ffffffff810ad97f>] print_usage_bug+0x1ff/0x210 [ 120.284209] [<ffffffff810134df>] ? save_stack_trace+0x2f/0x50 [ 120.284214] [<ffffffff810adffe>] mark_lock+0x66e/0x6f0 [ 120.284218] [<ffffffff810aceb0>] ? print_shortest_lock_dependencies+0x1d0/0x1d0 [ 120.284222] [<ffffffff810af3ab>] __lock_acquire+0xdcb/0x2380 [ 120.284226] [<ffffffff81091a2c>] ? __enqueue_entity+0x6c/0x70 [ 120.284230] [<ffffffff810ab8dd>] ? __lock_is_held+0x4d/0x70 [ 120.284234] [<ffffffff810ab8dd>] ? __lock_is_held+0x4d/0x70 [ 120.284238] [<ffffffff810ab8dd>] ? __lock_is_held+0x4d/0x70 [ 120.284242] [<ffffffff810ab8dd>] ? __lock_is_held+0x4d/0x70 [ 120.284246] [<ffffffff810b1631>] lock_acquire+0xb1/0x130 [ 120.284256] [<ffffffffa05ed210>] ? retire_capture_urb+0x140/0x2b0 [snd_usb_audio] [ 120.284261] [<ffffffff817fd61c>] _raw_spin_lock_irqsave+0x3c/0x50 [ 120.284270] [<ffffffffa05ed210>] ? retire_capture_urb+0x140/0x2b0 [snd_usb_audio] [ 120.284276] [<ffffffff81595325>] ? usb_hcd_get_frame_number+0x25/0x30 [ 120.284285] [<ffffffffa05ed210>] retire_capture_urb+0x140/0x2b0 [snd_usb_audio] [ 120.284294] [<ffffffffa05e513c>] snd_complete_urb+0x13c/0x250 [snd_usb_audio] [ 120.284298] [<ffffffff81592cb2>] __usb_hcd_giveback_urb+0x72/0x110 [ 120.284303] [<ffffffff81592e53>] usb_hcd_giveback_urb+0x43/0x140 [ 120.284307] [<ffffffff815d1b02>] xhci_irq+0xd42/0x1fc0 [ 120.284312] [<ffffffff815d2d91>] xhci_msi_irq+0x11/0x20 [ 120.284317] [<ffffffff810c4570>] handle_irq_event_percpu+0x80/0x1a0 [ 120.284322] [<ffffffff810c46da>] handle_irq_event+0x4a/0x70 [ 120.284325] [<ffffffff810c7794>] ? handle_edge_irq+0x24/0x150 [ 120.284329] [<ffffffff810c77f1>] handle_edge_irq+0x81/0x150 [ 120.284333] [<ffffffff81005c85>] handle_irq+0x25/0x40 [ 120.284337] [<ffffffff818008ef>] do_IRQ+0x4f/0xe0 [ 120.284341] [<ffffffff817fea2d>] common_interrupt+0x6d/0x6d [ 120.284343] <EOI> [<ffffffff816707b2>] ? cpuidle_enter_state+0xf2/0x230 [ 120.284351] [<ffffffff816707ad>] ? cpuidle_enter_state+0xed/0x230 [ 120.284355] [<ffffffff81670927>] cpuidle_enter+0x17/0x20 [ 120.284360] [<ffffffff810a3862>] call_cpuidle+0x32/0x60 [ 120.284364] [<ffffffff81670903>] ? cpuidle_select+0x13/0x20 [ 120.284369] [<ffffffff810a3aa6>] cpu_startup_entry+0x216/0x2d0 [ 120.284374] [<ffffffff8103679d>] start_secondary+0x12d/0x150 [ 125.426447] device: 'ep_84': device_unregister [ 125.426511] PM: Removing info for No Bus:ep_84 [ 125.426542] device: 'ep_84': device_add [ 125.426602] PM: Adding info for No Bus:ep_84 -- Shuah Khan Sr. Linux Kernel Developer Open Source Innovation Group Samsung Research America (Silicon Valley) shuahkh(a)osg.samsung.com | (970) 217-8978

3 5

[alsa-devel] [PATCH] ASoC: wm0010: fix memory leak
by Sudip Mukherjee 02 Sep '15

02 Sep '15

We were aborting if the kzalloc of img_swap fails but without freeing the already allocated out. Similarly we were aborting if spi_sync fails without releasing out and img_swap. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/wm0010.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f2c6ad4..8434d45 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -672,8 +672,10 @@ static int wm0010_boot(struct snd_soc_codec *codec) } img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); - if (!img_swap) + if (!img_swap) { + kfree(out); goto abort; + } /* We need to re-order for 0010 */ byte_swap_64((u64 *)&pll_rec, img_swap, len); @@ -690,6 +692,8 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = spi_sync(spi, &m); if (ret != 0) { dev_err(codec->dev, "First PLL write failed: %d\n", ret); + kfree(img_swap); + kfree(out); goto abort; } @@ -697,6 +701,8 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = spi_sync(spi, &m); if (ret != 0) { dev_err(codec->dev, "Second PLL write failed: %d\n", ret); + kfree(img_swap); + kfree(out); goto abort; } -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: hda - Fix missing inline for dummy snd_hdac_set_codec_wakeup()
by Takashi Iwai 02 Sep '15

02 Sep '15

This seems overlooked. Fixes: 98d8fc6c5d36 ('ALSA: hda - Move hda_i915.c from sound/pci/hda to sound/hda') Cc: <stable(a)vger.kernel.org> # v4.2+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_i915.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index a5b5cae03e87..930b41e5acf4 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -14,7 +14,7 @@ int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *); #else -static int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) { return 0; } -- 2.5.1

1 0

[alsa-devel] [PATCH v4 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 02 Sep '15

02 Sep '15

It's been a while since the last patch set iteration, due to me being on vacation. But here's a new set, and I still hope that it can make it into the next merge window. Changes since v3 (with the person suggesting that change within parantheses): * Valleyview now has three pins like all the others (Libin Yang) * Renamed a few references from "hotplug" to "pin_eld" to reduce confusion with hotplug code on the i915 side (Jani Nikula) * Rewrote the dispatching from hda core to codec on the hda side (Takashi Iwai) This iteration has been tested and working on one Skylake machine. David Henningsson (4): drm/i915: Add audio pin sense / ELD callback drm/i915: Call audio pin/ELD notify function ALSA: hda - allow codecs to access the i915 pin/ELD callback ALSA: hda - Wake the codec up on pin/ELD notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 23 ++++++++++++++++++++--- include/drm/i915_component.h | 12 ++++++++++++ include/sound/hda_i915.h | 7 +++++++ sound/hda/hdac_i915.c | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 22 +++++++++++++++++++++- 6 files changed, 71 insertions(+), 4 deletions(-) -- 1.9.1

4 18

[alsa-devel] [PATCH] ASoC: intel: broadwell: Convert to devm_snd_soc_register_card
by Axel Lin 02 Sep '15

02 Sep '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/intel/boards/broadwell.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 8bafaf6..3f8a1e1 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -266,18 +266,11 @@ 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; + return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286); } static struct platform_driver broadwell_audio = { .probe = broadwell_audio_probe, - .remove = broadwell_audio_remove, .driver = { .name = "broadwell-audio", }, -- 2.1.0

1 0

[alsa-devel] [ANNOUNCE] Media Workshop at the Kernel Summit 2015 in Korea
by Mauro Carvalho Chehab 01 Sep '15

01 Sep '15

Hi, As we keep being very active on the development of the Linux Kernel Media subsystem, we'll be doing another workshop this year. This will be the first time we'll be doing it in Asia, so it is a great opportunity for the ones that are there to join us and participate, in order to improve the Linux Kernel support for A/V stream capture, webcam and analog/digital TV. As usual, we'll be using the media-workshop(a)linuxtv.org Mailing List for specific discussions about that, so the ones interested on participate are requested to subscribe it. This time, we'll be doing something different than what we've done on the last years, and use something that other subsystems like ALSA are doing for a while: we'll be using two google docs to track the proposals and the interested parties. If you're interested in participate and wants to be invited for the event, please put your data at this document: https://docs.google.com/spreadsheets/d/1YecCAUo0L1R_i3jPg3S-A5iqX7BYAd2c_Vf… As this is the first time we'll be using Google Docs, I kindly request you to also send an email for media-workshop(a)linuxtv.org if you have any troubles adding things there and/or a backup measure. If you have any proposal for themes, please add it here: https://docs.google.com/spreadsheets/d/1F67VcdhkDlVfRpx-GPhe61adZh81Q26q--_… It is useful if you could also send an email to the workshop ML in order to better explain the theme proposal. Hoping to see you there soon! Mauro

3 3

[alsa-devel] [PATCH] ASoC: sh: siu_dai: Convert to use resource managed APIs
by Axel Lin 01 Sep '15

01 Sep '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/sh/siu_dai.c | 85 ++++++++++++++------------------------------------ 1 file changed, 24 insertions(+), 61 deletions(-) diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index abb0d95..76b2ab8 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -738,7 +738,7 @@ static int siu_probe(struct platform_device *pdev) struct siu_info *info; int ret; - info = kmalloc(sizeof(*info), GFP_KERNEL); + info = devm_kmalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; siu_i2s_data = info; @@ -746,7 +746,7 @@ static int siu_probe(struct platform_device *pdev) ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev); if (ret) - goto ereqfw; + return ret; /* * Loaded firmware is "const" - read only, but we have to modify it in @@ -757,89 +757,52 @@ static int siu_probe(struct platform_device *pdev) release_firmware(fw_entry); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - ret = -ENODEV; - goto egetres; - } + if (!res) + return -ENODEV; - region = request_mem_region(res->start, resource_size(res), - pdev->name); + region = devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name); if (!region) { dev_err(&pdev->dev, "SIU region already claimed\n"); - ret = -EBUSY; - goto ereqmemreg; + return -EBUSY; } - ret = -ENOMEM; - info->pram = ioremap(res->start, PRAM_SIZE); + info->pram = devm_ioremap(&pdev->dev, res->start, PRAM_SIZE); if (!info->pram) - goto emappram; - info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE); + return -ENOMEM; + info->xram = devm_ioremap(&pdev->dev, res->start + XRAM_OFFSET, + XRAM_SIZE); if (!info->xram) - goto emapxram; - info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE); + return -ENOMEM; + info->yram = devm_ioremap(&pdev->dev, res->start + YRAM_OFFSET, + YRAM_SIZE); if (!info->yram) - goto emapyram; - info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) - - REG_OFFSET); + return -ENOMEM; + info->reg = devm_ioremap(&pdev->dev, res->start + REG_OFFSET, + resource_size(res) - REG_OFFSET); if (!info->reg) - goto emapreg; + return -ENOMEM; dev_set_drvdata(&pdev->dev, info); /* register using ARRAY version so we can keep dai name */ - ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component, - &siu_i2s_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &siu_i2s_component, + &siu_i2s_dai, 1); if (ret < 0) - goto edaiinit; + return ret; - ret = snd_soc_register_platform(&pdev->dev, &siu_platform); + ret = devm_snd_soc_register_platform(&pdev->dev, &siu_platform); if (ret < 0) - goto esocregp; + return ret; pm_runtime_enable(&pdev->dev); - return ret; - -esocregp: - snd_soc_unregister_component(&pdev->dev); -edaiinit: - iounmap(info->reg); -emapreg: - iounmap(info->yram); -emapyram: - iounmap(info->xram); -emapxram: - iounmap(info->pram); -emappram: - release_mem_region(res->start, resource_size(res)); -ereqmemreg: -egetres: -ereqfw: - kfree(info); - - return ret; + return 0; } static int siu_remove(struct platform_device *pdev) { - struct siu_info *info = dev_get_drvdata(&pdev->dev); - struct resource *res; - pm_runtime_disable(&pdev->dev); - - snd_soc_unregister_platform(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - - iounmap(info->reg); - iounmap(info->yram); - iounmap(info->xram); - iounmap(info->pram); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res) - release_mem_region(res->start, resource_size(res)); - kfree(info); - return 0; } -- 2.1.0

1 0

[alsa-devel] SSM2518 Device Tree support ?
by Ricard Wanderlof 01 Sep '15

01 Sep '15

We needed DT support in the SSM2518 codec driver, and I noticed that there is already a set of bindings specified in Documentation/devicetree, however, there doesn't seem to be any DT support in the driver itself. I don't know if this is the result of a missing commit ? Documentation/devicetree/bindings/sound/ssm2518.c seems to have been added in May 2013. I've added rudimentary support for at least the compatible property so that the driver can be used in a system with DT configuration (sent as a separate email). /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 4

[alsa-devel] [PATCH v4] OMAPDSS: hdmi: Reconfigure and restart audio when display is enabled
by Jyri Sarha 01 Sep '15

01 Sep '15

Reconfigure and restart audio when display is enabled, if audio playback was active before. This is needed in a situation when an audio+video stream application opens the audio stream before the video. When video stream is opened the display mode may change and that aborts audio playback, because the display is momentarily turned off. The audio configuration is stored when it is successfully applied and a boolean is set when the audio playback is started and unset when stopped. This data is used to reconfigure the audio when display is re-enabled. The audio playback is aborted if the reconfiguration fails. A new spin lock is introduced in order to protect state variables related to audio playback status. This is needed for the transition from display enabled state (when audio start/stop commands can be written to HW) to display disabled state (when audio start/stop commands update only the hdmi.audio_playing variable) to always serialize correctly with the start/stop audio commands. The already existing mutex can not be used, because the audio start and stop commands are executed in atomic context. For example: when display is turned back on we take the spinlock and we can be sure that the audio start/stop status will not change while we update the HW according to hdmi.audio_playing state and set hdmi.display_enabled to true. After releasing the lock hdmi.display_enabled is true and all audio_start and audio_stop commands write their stuff directly to HW. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Fixed one bad sentece in the description. drivers/video/fbdev/omap2/dss/hdmi.h | 9 ++++- drivers/video/fbdev/omap2/dss/hdmi4.c | 66 +++++++++++++++++++++++++----- drivers/video/fbdev/omap2/dss/hdmi5.c | 76 ++++++++++++++++++++++++++++------- 3 files changed, 124 insertions(+), 27 deletions(-) diff --git a/drivers/video/fbdev/omap2/dss/hdmi.h b/drivers/video/fbdev/omap2/dss/hdmi.h index e4a32fe..53616b0 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi.h +++ b/drivers/video/fbdev/omap2/dss/hdmi.h @@ -351,13 +351,20 @@ struct omap_hdmi { struct regulator *vdda_reg; bool core_enabled; - bool display_enabled; struct omap_dss_device output; struct platform_device *audio_pdev; void (*audio_abort_cb)(struct device *dev); int wp_idlemode; + + bool audio_configured; + struct omap_dss_audio audio_config; + + /* This lock should be taken when booleans bellow are touched. */ + spinlock_t audio_playing_lock; + bool audio_playing; + bool display_enabled; }; #endif diff --git a/drivers/video/fbdev/omap2/dss/hdmi4.c b/drivers/video/fbdev/omap2/dss/hdmi4.c index 6d3aa3f..94c8d55 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi4.c +++ b/drivers/video/fbdev/omap2/dss/hdmi4.c @@ -321,9 +321,22 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_enable(&hd->wp, true); + hdmi4_audio_start(&hd->core, &hd->wp); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi4_audio_stop(&hd->core, &hd->wp); + hdmi_wp_audio_enable(&hd->wp, false); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -342,7 +355,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi4_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -354,17 +381,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -568,6 +597,8 @@ static int hdmi_audio_shutdown(struct device *dev) mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + hd->audio_playing = false; mutex_unlock(&hd->lock); return 0; @@ -576,25 +607,34 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi4_audio_start(&hd->core, &hd->wp); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_start_audio_stream(hd); + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi4_audio_stop(&hd->core, &hd->wp); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; + + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -612,7 +652,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi4_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); - + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -657,6 +700,7 @@ static int hdmi4_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); diff --git a/drivers/video/fbdev/omap2/dss/hdmi5.c b/drivers/video/fbdev/omap2/dss/hdmi5.c index 7f87578..b6c767b 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi5.c +++ b/drivers/video/fbdev/omap2/dss/hdmi5.c @@ -349,9 +349,24 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + hdmi_wp_audio_enable(&hd->wp, true); + hdmi_wp_audio_core_req_enable(&hd->wp, true); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_core_req_enable(&hd->wp, false); + hdmi_wp_audio_enable(&hd->wp, false); + REG_FLD_MOD(hd->wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -370,7 +385,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi5_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -382,17 +411,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -596,6 +627,8 @@ static int hdmi_audio_shutdown(struct device *dev) mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + hd->audio_playing = false; mutex_unlock(&hd->lock); return 0; @@ -604,32 +637,35 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - /* No-idle while playing audio, store the old value */ - hd->wp_idlemode = REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + spin_lock_irqsave(&hd->audio_playing_lock, flags); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi_wp_audio_core_req_enable(&hd->wp, true); + if (hd->display_enabled) { + hdmi_start_audio_stream(hd); + } + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_core_req_enable(&hd->wp, false); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; - /* Playback stopped, restore original idlemode */ - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -648,6 +684,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi5_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -678,6 +718,11 @@ static int hdmi_audio_register(struct device *dev) if (IS_ERR(hdmi.audio_pdev)) return PTR_ERR(hdmi.audio_pdev); + hdmi_runtime_get(); + hdmi.wp_idlemode = + REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); + hdmi_runtime_put(); + return 0; } @@ -692,6 +737,7 @@ static int hdmi5_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); -- 1.9.1

2 1

[alsa-devel] Beaglebone Black: Transmit buffer underflows when trying to play audio using an externally attached codec board (simple-card, davinci-mcasp, cs4270)
by Shadi Abdu-Rahman 01 Sep '15

01 Sep '15

Hi all, I'm having trouble using an audio codec board with a custom device tree overlay (see below) with the Beaglebone Black (kernel 4.1.3-bone15). When exporting the overlay, the codecs seem to register fine: dmesg: [ 1375.144463] bone_capemgr bone_capemgr: part_number 'BB-BONE-XA-SK-AU', version 'N/A' [ 1375.152305] bone_capemgr bone_capemgr: slot #4: override [ 1375.157799] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 1375.164864] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-BONE-XA-SK-AU' [ 1375.206351] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-BONE-XA-SK-AU-00A0.dtbo' loaded; overlay id #0 [ 1375.464260] cs4270 2-0048: found device at i2c address 48 [ 1375.469699] cs4270 2-0048: hardware revision 3 [ 1375.485782] cs4270 2-0049: found device at i2c address 49 [ 1375.491222] cs4270 2-0049: hardware revision 3 [ 1375.504744] (NULL device *): Setting gpio_int_masterclk_enable = 0 [ 1375.513334] asoc-simple-card ocp:sound: cs4270-hifi <-> 48038000.mcasp mapping ok [ 1375.521095] (NULL device *): Setting gpio_int_masterclk_enable = 0 [ 1375.529893] asoc-simple-card ocp:sound: cs4270-hifi <-> 48038000.mcasp mapping ok aplay -l: **** List of PLAYBACK Hardware Devices **** card 0: XMOSAudioSlice [XMOS-Audio-Slice], device 0: davinci-mcasp.0-cs4270-hifi-a cs4270-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: XMOSAudioSlice [XMOS-Audio-Slice], device 1: davinci-mcasp.0-cs4270-hifi-b cs4270-hifi-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 lsmod: snd_soc_davinci_mcasp 14391 2 snd_soc_simple_card 8156 0 snd_soc_edma 1150 1 snd_soc_davinci_mcasp snd_soc_cs4270 6875 2 snd_soc_omap 2573 1 snd_soc_davinci_mcasp snd_soc_core 156881 5 snd_soc_davinci_mcasp,snd_soc_edma,snd_soc_omap,snd_soc_simple_card,snd_soc_cs4270 snd_compress 11668 1 snd_soc_core snd_pcm_dmaengine 5061 2 snd_soc_core,snd_soc_omap snd_pcm 77081 4 snd_soc_davinci_mcasp,snd_soc_core,snd_soc_omap,snd_pcm_dmaengine snd_timer 16860 1 snd_pcm snd 56902 4 snd_soc_core,snd_timer,snd_pcm,snd_compress soundcore 6869 1 snd pvrsrvkm 147014 0 omap_aes 13089 0 omap_sham 19190 0 tda998x 11683 1 tilcdc 27919 0 omap_rng 4354 0 rng_core 7270 1 omap_rng drm_kms_helper 106610 3 tda998x,tilcdc uio_pdrv_genirq 3317 0 leds_gpio 3102 0 uio 8330 1 uio_pdrv_genirq|| || | |but when I try to play an audio file I get a stream of transmit buffer underflows and no audio: aplay test128.wav: Playing WAVE 'test128.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo underrun!!! (at least 1.630 ms long) underrun!!! (at least 5.413 ms long) underrun!!! (at least 0.636 ms long) underrun!!! (at least 5.369 ms long) underrun!!! (at least 0.217 ms long) underrun!!! (at least 0.641 ms long) underrun!!! (at least 5.254 ms long) dmesg: [ 1493.190031] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.201647] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.218719] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.229946] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.245328] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.259411] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.270549] davinci-mcasp 48038000.mcasp: Transmit buffer underflow Using a software oscilloscope, I can see that the frame clock starts on mcasp0_fsx frame clock line but stops after a few periods. There’s no activity on the on the mcasp0_axr2 audio out line. I'm not sure if I'm missing something trivial or if this is a general with audio out on the Beaglebone Black and 4.1.x kernels. Any pointers or help would be much appreciated. Additional information: I’m using the v4.1.3-bone15 linux kernel with debian-8.1-minimal-armhf-2015-06-09 compiled as per Robert C Nelsons instructions (https://eewiki.net/display/linuxonarm/BeagleBone+Black). The board I’m using is an XMOS XA-SK-AUDIO slice and consists of two Cirrus Logic cs4270 codecs and an onboard oscillator generating the master clock. I’ve interfaced the board to a XMOS micro controller to make sure the board is in working order (and it is). When connected to the BBB, reset de-assertion and some initial i2c codec configuration (setting the codecs in software mode) are handled by the same XMOS micro controller. The overlay I’m using looks like this: /dts-v1/; /plugin/; / { compatible = "ti,beaglebone", "ti,beaglebone-black"; part-number = "BB-BONE-XA-SK-AU"; version = "00A0"; exclusive-use = /* pin header usage */ "P9.25", /* mcasp0_ahclkx <- MCLK */ "P9.31", /* mcasp0_aclkx <-> BCLK*/ "P9.29", /* mcasp0_fsx <-> LRCLK */ "P9.30", /* mcasp0_axr0 <- ADC_DATA0 */ "P9.41", /* mcasp0_axr1 <- ADC_DATA1 */ "P9.28", /* mcasp0_axr2 -> DAC_DATA0 */ "P9.27", /* mcasp0_axr3 -> DAC_DATA1 */ /* hardware ip usage */ "gpio0_15", /* ext-mclk-sel -> MCLK_FSEL */ "gpio0_20", /* CLKOUT2 disabled */ "gpio1_27", /* internal 24.576 MHz oscillator disabled */ "mcasp0"; fragment@0 { target = <&am33xx_pinmux>; __overlay__ { i2c2_pins: pinmux_i2c2_pins { pinctrl-single,pins = < /* Pullup resistors available on codec board */ 0x17c 0x2b /* i2c2_scl, (PIN_INPUT | MUX_MODE3) */ 0x178 0x2b /* i2c2_sda, (PIN_INPUT | MUX_MODE3) */ >; }; mcasp0_pins: pinmux_mcasp0_pins { pinctrl-single,pins = < 0x1ac 0x20 /* mcasp0_ahclkx, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x190 0x20 /* mcasp0_aclkx, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x194 0x20 /* mcasp0_fsx, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x198 0x20 /* mcasp0_axr0, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x1a8 0x20 /* mcasp0_axr1, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x19c 0x02 /* mcasp0_axr2, (PIN_OUTPUT_PULLDOWN | MUX_MODE2) */ 0x1a4 0x02 /* mcasp0_axr3, (PIN_OUTPUT_PULLDOWN | MUX_MODE2) */ /* disable internal 24.576 MHz oscillator */ 0x06c 0x07 /* gpio1_27, (PIN_OUTPUT_PULLDOWN | MUX_MODE7) */ >; }; gpio_pins: pinmux_gpio_pins { pinctrl-single,pins = < /* ext-mclk-sel */ 0x184 0x17 /* gpio0_15, (PIN_OUTPUT_PULLUP | MUX_MODE7) */ /* disable CLKOUT2 to allow usage of mcasp0_axr1 on P9_41 */ 0x1b4 0x2b /* gpio0_20, (PIN_INPUT | MUX_MODE3) */ >; }; }; }; fragment@1 { target = <&i2c2>; __overlay__ { pinctrl-names = "default"; pinctrl-0 = <&i2c2_pins>; status = "okay"; #address-cells = <1>; #size-cells = <0>; clock-frequency = <100000>; cs4270m: cs4270@48 { compatible = "cirrus,cs4270"; status = "okay"; #sound-dai-cells = <0>; reg = <0x48>; name-prefix = "a"; va-supply = <&ldo4_reg>; vd-supply = <&ldo4_reg>; vlc-supply = <&ldo4_reg>; }; cs4270s: cs4270@49 { compatible = "cirrus,cs4270"; status = "okay"; #sound-dai-cells = <0>; reg = <0x49>; name-prefix = "b"; va-supply = <&ldo4_reg>; vd-supply = <&ldo4_reg>; vlc-supply = <&ldo4_reg>; }; }; }; fragment@2 { target = <&mcasp0>; __overlay__ { pinctrl-names = "default"; pinctrl-0 = <&mcasp0_pins>; status = "okay"; #sound-dai-cells = <0>; op-mode = <0>; // MCASP_IIS_MODE tdm-slots = <2>; num-serializer = <4>; serial-dir = < // 0: INACTIVE, 1: TX, 2: RX 2 2 1 1 >; tx-num-evt = <1>; rx-num-evt = <1>; }; }; fragment@3 { target = <&ocp>; __overlay__ { sound { compatible = "simple-audio-card"; simple-audio-card,name = "XMOS-Audio-Slice"; simple-audio-card,widgets = "Line", "Line In 1", "Line", "Line Out 1", "Line", "Line In 2", "Line", "Line Out 2" ; simple-audio-card,routing = "a AINA", "Line In 1", "a AINB", "Line In 1", "Line Out 1", "a AOUTA", "Line Out 1", "a AOUTB", "b AINA", "Line In 2", "b AINB", "Line In 2", "Line Out 2", "b AOUTA", "Line Out 2", "b AOUTB" ; simple-audio-card,int-masterclk-enable = <&gpio1 27 0>; simple-audio-card,dai-link@1 { format = "i2s"; frame-master = <&cpu_node1>; bitclock-master = <&cpu_node1>; cpu_node1: cpu { sound-dai = <&mcasp0>; system-clock-frequency = <24576000>; }; codec_node1: codec { sound-dai = <&cs4270s>; system-clock-frequency = <24576000>; }; }; simple-audio-card,dai-link@0 { format = "i2s"; frame-master = <&cpu_node0>; bitclock-master = <&cpu_node0>; cpu_node0: cpu { sound-dai = <&mcasp0>; system-clock-frequency = <24576000>; }; codec_node0: codec { sound-dai = <&cs4270m>; system-clock-frequency = <24576000>; }; }; }; }; }; }; I've also made modifications to cs4270.c, simple-card.c, and soc-core.c to add a mechanism to disable the internal BBB master clock (in simple-card.c), as well as to fix issues with missing source/sink widgets (this seems to be a bug in cs4270.c): [ 137.408549] cs4270 2-0048: ASoC: no source widget found for AINA [ 137.414737] cs4270 2-0048: ASoC: Failed to add route AINA -> direct -> Capture [ 137.422016] cs4270 2-0048: ASoC: no source widget found for AINB [ 137.428137] cs4270 2-0048: ASoC: Failed to add route AINB -> direct -> Capture [ 137.435477] cs4270 2-0048: ASoC: no sink widget found for AOUTA [ 137.441445] cs4270 2-0048: ASoC: Failed to add route Playback -> direct -> AOUTA [ 137.448928] cs4270 2-0048: ASoC: no sink widget found for AOUTB [ 137.454922] cs4270 2-0048: ASoC: Failed to add route Playback -> direct -> AOUTB and name collisions, arising from using two identical codecs (by hacking together name-prefix support in simple-card): [ 51.364492] asoc-simple-card ocp:sound: control 2:0:0:Master Playback Volume:0 is already present [ 51.373469] cs4270 2-0049: ASoC: Failed to add Master Playback Volume: -16 ... [ 51.393905] ------------[ cut here ]------------ [ 51.398561] WARNING: CPU: 0 PID: 44 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x5c() [ 51.406281] sysfs: cannot create duplicate filename '/devices/platform/ocp/ocp:sound/davinci-mcasp.0-cs4270-hifi' The modifications are as follows: diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e6d4ff9..4805747 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -140,11 +140,11 @@ struct cs4270_private { }; static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = { -SND_SOC_DAPM_INPUT("AINL"), -SND_SOC_DAPM_INPUT("AINR"), +SND_SOC_DAPM_INPUT("AINA"), +SND_SOC_DAPM_INPUT("AINB"), -SND_SOC_DAPM_OUTPUT("AOUTL"), -SND_SOC_DAPM_OUTPUT("AOUTR"), +SND_SOC_DAPM_OUTPUT("AOUTA"), +SND_SOC_DAPM_OUTPUT("AOUTB"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 33feee9..1383b0f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -28,6 +28,7 @@ struct simple_card_data { struct asoc_simple_dai codec_dai; } *dai_props; unsigned int mclk_fs; + int gpio_int_masterclk_enable; int gpio_hp_det; int gpio_hp_det_invert; int gpio_mic_det; @@ -50,7 +51,7 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) ret = clk_prepare_enable(dai_props->cpu_dai.clk); if (ret) return ret; - + ret = clk_prepare_enable(dai_props->codec_dai.clk); if (ret) clk_disable_unprepare(dai_props->cpu_dai.clk); @@ -167,6 +168,12 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + /* TODO: Logic for enabling internal clock if specified in DT*/ + if (gpio_is_valid(priv->gpio_int_masterclk_enable)) { + dev_info(NULL, "Setting gpio_int_masterclk_enable = 0"); + gpio_set_value(priv->gpio_int_masterclk_enable, 0); + } + if (gpio_is_valid(priv->gpio_hp_det)) { snd_soc_card_jack_new(rtd->card, "Headphones", SND_JACK_HEADPHONE, @@ -195,16 +202,18 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) } static int -asoc_simple_card_sub_parse_of(struct device_node *np, - struct asoc_simple_dai *dai, - struct device_node **p_node, - const char **name, - int *args_count) +asoc_simple_card_sub_parse_of(struct simple_card_data *priv, + struct device_node *np, + struct asoc_simple_dai *dai, + struct device_node **p_node, + const char **name, + int *args_count, bool codec_device, int idx) { struct of_phandle_args args; struct clk *clk; u32 val; int ret; + const char *str = NULL; /* * Get node via "sound-dai = <&phandle port>" @@ -225,6 +234,18 @@ asoc_simple_card_sub_parse_of(struct device_node *np, if (ret < 0) return ret; + /* get name-prefix */ + if (codec_device) { + /* NULL bypasses the device name check in soc-core */ + priv->snd_card.codec_conf[idx].dev_name = NULL; + priv->snd_card.codec_conf[idx].of_node = *p_node; + of_property_read_string(*p_node, "name-prefix", &str); + if (str) + priv->snd_card.codec_conf[idx].name_prefix = str; + else + priv->snd_card.codec_conf[idx].name_prefix = NULL; + } + /* Parse TDM slot */ ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); if (ret) @@ -334,16 +355,16 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_sub_parse_of(cpu, &dai_props->cpu_dai, + ret = asoc_simple_card_sub_parse_of(priv, cpu, &dai_props->cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name, - &cpu_args); + &cpu_args, false, idx); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_sub_parse_of(codec, &dai_props->codec_dai, + ret = asoc_simple_card_sub_parse_of(priv, codec, &dai_props->codec_dai, &dai_link->codec_of_node, - &dai_link->codec_dai_name, NULL); + &dai_link->codec_dai_name, NULL, true, idx); if (ret < 0) goto dai_link_of_err; @@ -357,8 +378,10 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, /* DAI link name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(dai_link->cpu_dai_name) + - strlen(dai_link->codec_dai_name) + 2, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + + (priv->snd_card.codec_conf[idx].name_prefix ? + strlen(priv->snd_card.codec_conf[idx].name_prefix) + 3 : 2), GFP_KERNEL); if (!name) { ret = -ENOMEM; @@ -367,6 +390,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); + if (priv->snd_card.codec_conf[idx].name_prefix) + sprintf(name, "%s-%s", name, + priv->snd_card.codec_conf[idx].name_prefix); dai_link->name = dai_link->stream_name = name; dai_link->ops = &asoc_simple_card_ops; dai_link->init = asoc_simple_card_dai_init; @@ -459,6 +485,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + /* request GPIO to control internal 24.576MHz oscillator */ + priv->gpio_int_masterclk_enable = of_get_named_gpio_flags(node, + "simple-audio-card,int-masterclk-enable", 0, &flags); + + if (priv->gpio_int_masterclk_enable == -EPROBE_DEFER) + return -EPROBE_DEFER; + gpio_request_one(priv->gpio_int_masterclk_enable, GPIOF_OUT_INIT_LOW, + "Internal Clock Enable Pin"); + priv->gpio_hp_det = of_get_named_gpio_flags(node, "simple-audio-card,hp-det-gpio", 0, &flags); priv->gpio_hp_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); @@ -498,13 +533,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; - int num_links, ret; + int num_links = (np && of_get_child_by_name(np, + "simple-audio-card,dai-link")) ? + of_get_child_count(np) : 1; + struct snd_soc_codec_conf codec_conf[num_links]; + int ret; - /* Get the number of DAI links */ + /* Get the number of DAI links if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) num_links = of_get_child_count(np); else num_links = 1; + */ /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, @@ -519,6 +559,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link = priv->dai_link; priv->snd_card.dai_link = dai_link; priv->snd_card.num_links = num_links; + priv->snd_card.codec_conf = codec_conf; + priv->snd_card.num_configs = num_links; priv->gpio_hp_det = -ENOENT; priv->gpio_mic_det = -ENOENT; @@ -589,6 +631,11 @@ static int asoc_simple_card_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + if (gpio_is_valid(priv->gpio_int_masterclk_enable)) { + gpio_set_value(priv->gpio_int_masterclk_enable, 0); + gpio_free(priv->gpio_int_masterclk_enable); + } + if (gpio_is_valid(priv->gpio_hp_det)) snd_soc_jack_free_gpios(&simple_card_hp_jack, 1, &simple_card_hp_jack_gpio); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2373252..4808089 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1102,6 +1102,9 @@ static void soc_set_name_prefix(struct snd_soc_card *card, for (i = 0; i < card->num_configs; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; + + if (!map->name_prefix) + continue; if (map->of_node && component->dev->of_node != map->of_node) continue; if (map->dev_name && strcmp(component->name, map->dev_name)) Another possible bug: Adding name-prefix support to simple-card uncovers a possible bug. You'll notice in the device tree overlay that I'v listed the simple-audio-card dai-links in reverse order, with dai-link@1 before dai-link@0. I've done so because using the normal order the wrong prefix gets assigned to the wrong dai-link. I'm not really sure where this happens, and it's not a huge issue for me, but I thought I'd report it. Kind Regards Shadi Abdu-Rahman

1 0

[alsa-devel] [PATCH RFT] ASoC: pxa: Convert to devm_snd_soc_register_card
by Axel Lin 01 Sep '15

01 Sep '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Only compile tested. sound/soc/pxa/brownstone.c | 9 +-------- sound/soc/pxa/corgi.c | 11 +---------- sound/soc/pxa/e740_wm9705.c | 5 +---- sound/soc/pxa/e750_wm9705.c | 5 +---- sound/soc/pxa/e800_wm9712.c | 5 +---- sound/soc/pxa/hx4700.c | 4 +--- sound/soc/pxa/imote2.c | 11 +---------- sound/soc/pxa/mioa701_wm9713.c | 11 +---------- sound/soc/pxa/palm27x.c | 9 +-------- sound/soc/pxa/poodle.c | 11 +---------- sound/soc/pxa/spitz.c | 5 +---- sound/soc/pxa/tosa.c | 5 +---- sound/soc/pxa/ttc-dkb.c | 12 +----------- 13 files changed, 13 insertions(+), 90 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 2b26318..6147e86 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -116,26 +116,19 @@ static int brownstone_probe(struct platform_device *pdev) int ret; brownstone.dev = &pdev->dev; - ret = snd_soc_register_card(&brownstone); + ret = devm_snd_soc_register_card(&pdev->dev, &brownstone); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int brownstone_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&brownstone); - return 0; -} - static struct platform_driver mmp_driver = { .driver = { .name = "brownstone-audio", .pm = &snd_soc_pm_ops, }, .probe = brownstone_probe, - .remove = brownstone_remove, }; module_platform_driver(mmp_driver); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 3580d10..c97dc13 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -295,28 +295,19 @@ static int corgi_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int corgi_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 corgi_driver = { .driver = { .name = "corgi-audio", .pm = &snd_soc_pm_ops, }, .probe = corgi_probe, - .remove = corgi_remove, }; module_platform_driver(corgi_driver); diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index d72e124..1de8765 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -138,7 +138,7 @@ static int e740_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -149,10 +149,7 @@ static int e740_probe(struct platform_device *pdev) static int e740_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 48f2d7c..b7eb7cd 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -120,7 +120,7 @@ static int e750_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -131,10 +131,7 @@ static int e750_probe(struct platform_device *pdev) static int e750_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 45d4bd4..41bf714 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -119,7 +119,7 @@ static int e800_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -130,10 +130,7 @@ static int e800_probe(struct platform_device *pdev) static int e800_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 9f8be7c..ecbf287 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -193,7 +193,7 @@ static int hx4700_audio_probe(struct platform_device *pdev) return ret; snd_soc_card_hx4700.dev = &pdev->dev; - ret = snd_soc_register_card(&snd_soc_card_hx4700); + ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700); if (ret) gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios)); @@ -203,8 +203,6 @@ static int hx4700_audio_probe(struct platform_device *pdev) static int hx4700_audio_remove(struct platform_device *pdev) { - snd_soc_unregister_card(&snd_soc_card_hx4700); - gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); gpio_set_value(GPIO107_HX4700_SPK_nSD, 0); diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 29fabbf..9d0e407 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -72,28 +72,19 @@ static int imote2_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int imote2_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 imote2_driver = { .driver = { .name = "imote2-audio", .pm = &snd_soc_pm_ops, }, .probe = imote2_probe, - .remove = imote2_remove, }; module_platform_driver(imote2_driver); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index a9615a57..29bc60e8 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -181,7 +181,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev) return -ENODEV; mioa701.dev = &pdev->dev; - rc = snd_soc_register_card(&mioa701); + rc = devm_snd_soc_register_card(&pdev->dev, &mioa701); if (!rc) dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" "lead to overheating and possible destruction of your device." @@ -189,17 +189,8 @@ static int mioa701_wm9713_probe(struct platform_device *pdev) return rc; } -static int mioa701_wm9713_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 mioa701_wm9713_driver = { .probe = mioa701_wm9713_probe, - .remove = mioa701_wm9713_remove, .driver = { .name = "mioa701-wm9713", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index c20bbc0..4e74d95 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -140,22 +140,15 @@ static int palm27x_asoc_probe(struct platform_device *pdev) palm27x_asoc.dev = &pdev->dev; - ret = snd_soc_register_card(&palm27x_asoc); + ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int palm27x_asoc_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&palm27x_asoc); - return 0; -} - static struct platform_driver palm27x_wm9712_driver = { .probe = palm27x_asoc_probe, - .remove = palm27x_asoc_remove, .driver = { .name = "palm27x-asoc", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 80b457a..84d0e2e 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -267,28 +267,19 @@ static int poodle_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int poodle_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 poodle_driver = { .driver = { .name = "poodle-audio", .pm = &snd_soc_pm_ops, }, .probe = poodle_probe, - .remove = poodle_remove, }; module_platform_driver(poodle_driver); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 461123a..b002226 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -305,7 +305,7 @@ static int spitz_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -322,9 +322,6 @@ err1: static int spitz_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); gpio_free(spitz_mic_gpio); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index f59f566..49518dd 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -233,7 +233,7 @@ static int tosa_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -244,10 +244,7 @@ static int tosa_probe(struct platform_device *pdev) static int tosa_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free(TOSA_GPIO_L_MUTE); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 1753c7d..65c20f7 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -128,7 +128,7 @@ static int ttc_dkb_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -136,22 +136,12 @@ static int ttc_dkb_probe(struct platform_device *pdev) return ret; } -static int ttc_dkb_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 ttc_dkb_driver = { .driver = { .name = "ttc-dkb-audio", .pm = &snd_soc_pm_ops, }, .probe = ttc_dkb_probe, - .remove = ttc_dkb_remove, }; module_platform_driver(ttc_dkb_driver); -- 2.1.0

1 0

[alsa-devel] feature request: alsamixer should show alternatives to function key bindings, at least F1/Help
by frederik＠ofb.net 01 Sep '15

01 Sep '15

In alsamixer, many options are bound to function keys. These are used by my window manager, so I cannot use them in my terminal. The alsamixer man page mentions that 'gnome-terminal' cannot use function keys either. I noticed from the "Help" screen that many function keys have non-function-key alternatives, for instance instead of F6 I can press "s". This is great, but there is one remaining problem: The legend (at the top of the alsamixer screen) labels commands with their function-key bindings. This includes "Help" which is labelled with "F1". Seeing this, users are likely to assume that "F1" is the only binding for "Help". But it turns out that one can also access "Help" with "?" and "H". This can be learned by reading the man page or pressing keys experimentally, but it would be nice if "Help" was just labeled with either "?" or "H", as that allows people who are unable to use function keys to easily access the help screen and then see that alternate (non-function-key) bindings exist for other commands as well. As an aside, it seems odd to map "q" to "increase left volume"... (from https://bugs.freedesktop.org/show_bug.cgi?id=91782, edited)

1 0

[alsa-devel] feature request: "mute" indication in alsamixer is not self-evident
by frederik＠ofb.net 01 Sep '15

01 Sep '15

When a user presses 'm' for 'mute' in alsamixer, then a string "MM" appears at the bottom of a volume control. This change makes it obvious to the user that "MM" stands for "Mute". However, when a new user opens alsamixer after some software on his computer has (perhaps by mistake) automatically muted a volume control, the "MM" won't indicate anything obvious. Among other things, this makes it difficult for new users to debug issues with the auto-mute feature. The fact that the volume control can be at maximum, even when the control is muted, contributes to the potential for confusion. The problem would be solved if, for instance, the "Item: " string at the top of the alsamixer screen were changed to contain the word "(muted)" for muted controls. This way, users who don't have sound will see what the problem is when they open alsamixer. (from <https://bugs.freedesktop.org/show_bug.cgi?id=91781>, edited)

1 0

[alsa-devel] bug tracker down?
by frederik＠ofb.net 01 Sep '15

01 Sep '15

Hi guys, For some reason I can't connect to https://bugtrack.alsa-project.org/alsa-bug I want to file two bugs which I mistakenly sent to the Pulseaudio bug tracker, having to do with alsamixer. They are here and here: https://bugs.freedesktop.org/show_bug.cgi?id=91781 https://bugs.freedesktop.org/show_bug.cgi?id=91782 Thanks, Frederick

2 2

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
by Adam Thomson 31 Aug '15

31 Aug '15

Current code incorrectly treats dai format for AC97 as bit mask whereas it's actually an integer value. This causes DAI formats other than AC97 (e.g. DSP_B) to trigger AC97 related code, which is incorrect and breaks functionality. This patch fixes the code to correctly compare values to determine AC97 or not. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c0b940e..ee47041 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -945,7 +946,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.3

1 0

[alsa-devel] [PATCH] reset: Add (devm_)reset_control_get stub functions
by Axel Lin 31 Aug '15

31 Aug '15

So the drivers can be compiled with CONFIG_RESET_CONTROLLER disabled. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- include/linux/reset.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/linux/reset.h b/include/linux/reset.h index da5602b..cd42c1b 100644 --- a/include/linux/reset.h +++ b/include/linux/reset.h @@ -74,6 +74,18 @@ static inline int device_reset_optional(struct device *dev) return -ENOSYS; } +static inline struct reset_control *reset_control_get( + struct device *dev, const char *id) +{ + return ERR_PTR(-ENOSYS); +} + +static inline struct reset_control *devm_reset_control_get( + struct device *dev, const char *id) +{ + return ERR_PTR(-ENOSYS); +} + static inline struct reset_control *reset_control_get_optional( struct device *dev, const char *id) { -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: Document snd-soc-dummy-dai purpose
by Anatol Pomozov 31 Aug '15

31 Aug '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/soc-utils.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 362c69a..53dd085 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec; SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) +/* + * The dummy CODEC is only meant to be used in situations where there is no + * actual hardware. + * + * If there is actual hardware even if it does not have a control bus + * the hardware will still have constraints like supported samplerates, etc. + * which should be modelled. And the data flow graph also should be modelled + * using DAPM. + */ static struct snd_soc_dai_driver dummy_dai = { .name = "snd-soc-dummy-dai", .playback = { -- 2.5.0.457.gab17608

2 1

[alsa-devel] [PATCH 1/4] ASoC: au1x: Convert to devm_snd_soc_register_card
by Axel Lin 31 Aug '15

31 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/au1x/db1000.c | 10 +--------- sound/soc/au1x/db1200.c | 10 +--------- 2 files changed, 2 insertions(+), 18 deletions(-) diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c index 452f404..e97c327 100644 --- a/sound/soc/au1x/db1000.c +++ b/sound/soc/au1x/db1000.c @@ -38,14 +38,7 @@ static int db1000_audio_probe(struct platform_device *pdev) { struct snd_soc_card *card = &db1000_ac97; card->dev = &pdev->dev; - return snd_soc_register_card(card); -} - -static int db1000_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - snd_soc_unregister_card(card); - return 0; + return devm_snd_soc_register_card(&pdev->dev, card); } static struct platform_driver db1000_audio_driver = { @@ -54,7 +47,6 @@ static struct platform_driver db1000_audio_driver = { .pm = &snd_soc_pm_ops, }, .probe = db1000_audio_probe, - .remove = db1000_audio_remove, }; module_platform_driver(db1000_audio_driver); diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 58c3164..638ca0b 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -174,14 +174,7 @@ static int db1200_audio_probe(struct platform_device *pdev) card = db1200_cards[pid->driver_data]; card->dev = &pdev->dev; - return snd_soc_register_card(card); -} - -static int db1200_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - snd_soc_unregister_card(card); - return 0; + return devm_snd_soc_register_card(&pdev->dev, card); } static struct platform_driver db1200_audio_driver = { @@ -191,7 +184,6 @@ static struct platform_driver db1200_audio_driver = { }, .id_table = db1200_pids, .probe = db1200_audio_probe, - .remove = db1200_audio_remove, }; module_platform_driver(db1200_audio_driver); -- 2.1.0

3 5

[alsa-devel] [PATCH 2/2] ASoC: fsl-asoc-card: add AC'97 support
by Maciej S. Szmigiero 31 Aug '15

31 Aug '15

Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 145 +++++++++++++++---- 2 files changed, 120 insertions(+), 35 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 96f55ae..385a9f8 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -404,10 +451,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; - struct platform_device *cpu_pdev; + struct platform_device *cpu_pdev, *codec_pdev; struct fsl_asoc_card_priv *priv; - struct i2c_client *codec_dev; - struct clk *codec_clk; + struct i2c_client *codec_i2c_dev; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,11 +479,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) { + codec_pdev = of_find_device_by_node(codec_np); + codec_i2c_dev = of_find_i2c_device_by_node(codec_np); + } else { + codec_pdev = NULL; + codec_i2c_dev = NULL; } asrc_np = of_parse_phandle(np, "audio-asrc", 0); @@ -446,10 +493,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_pdev) { + struct clk *codec_clk = clk_get(&codec_pdev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,12 +536,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; goto asrc_fail; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_pdev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -508,7 +568,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_i2c_dev ? codec_i2c_dev->name : codec_pdev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -532,8 +594,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -544,6 +624,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -579,14 +661,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", },

1 0

[alsa-devel] [PATCH 1/2] ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device
by Maciej S. Szmigiero 31 Aug '15

31 Aug '15

In case of unknown DT compatible device the ASRC OF node possibly acquired earlier by of_parse_phandle() has to be put before returning from probe method. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl-asoc-card.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed..96f55ae 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); - return -EINVAL; + ret = -EINVAL; + goto asrc_fail; } /* Common settings for corresponding Freescale CPU DAI driver */

2 1

[alsa-devel] [GIT PULL] ASoC updates for v4.3
by Mark Brown 31 Aug '15

31 Aug '15

The following changes since commit c13dcf9f2d6f5f06ef1bf79ec456df614c5e058b: Linux 4.2-rc8 (2015-08-23 20:52:59 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.2-rc8 for you to fetch changes up to 91931320cfbbcc1dd874c2d0aef62e7d90f07f70: Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/xtensa' and 'asoc/topic/zx296702' into asoc-next (2015-08-30 15:59:18 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.3 Not many updates to the core here, but an awful lot of driver updates this time round: - Factoring out of AC'97 reset code into the core - New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek RT298 and ST STI controllers. - Machine drivers for Rockchip systems with MAX98090 and RT5645 and RT5650. - Initial driver support for Intel Skylake devices. - A large number of cleanups for Lars-Peter Clausen and Axel Lin. ---------------------------------------------------------------- Anatol Pomozov (4): ASoC: max98357a: Make 'sdmode-gpios' dts property optional ASoC: max98357a: Use DAI input as a dapm widget ASoC: max98357a: Do not print error message on asoc/gpio errors ASoC: rl6231: Simplify DMIC divider calculation expression Andrew Morton (1): ASoC: cs4349: include gpio/consumer.h Andrzej Hajda (2): ASoC: intel: use kmemdup rather than duplicating its implementation ASoC: topology: use kmemdup rather than duplicating its implementation Antonio Borneo (1): ASoC: wm0010: Remove redundant spi driver bus initialization Arnaud Pouliquen (12): ASoC: simple card: set cpu-dai sysclk with mclk-fs ASoC: sti-sas: Add sti platform codec ASoC: sti: add binding for ASoC driver ASoC: sti: Add uniperipheral header file ASoC: sti: Add CPU DAI driver for playback ASoC: sti-sas: fix x86 compilation issue. ASoC: sti: minor corrections for uniplayer ASoC: sti: Add CPU DAI driver for capture ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control Axel Lin (73): ASoC: gtm601: Fix modalias ASoC: rt5645: Constify dmi_system_id table ASoC: cs42xx8: Move the code checking *regmap argument earlier ASoC: cs42xx8: Setup of_match_table ASoC: 88pm860x: Don't change pm860x->dir setting if pm860x_set_dai_sysclk fails ASoC: ak4642: Fix up max_register setting ASoC: lm49453: Remove fs_rate from struct lm49453_priv ASoC: gtm601: Constify soc_codec_dev_gtm601 ASoC: Constify reg_default tables ASoC: rt286: Constify dmi_system_id table ASoC: wm0010: Remove spurious missing IRQF_ONESHOT ASoC: wm5100: Remove spurious IRQF_ONESHOT flag ASoC: wm8996: Remove spurious IRQF_ONESHOT flag ASoC: sti-sas: Fix checking return value for ERR_PTR ASoC: sti-sas: Staticise local symbols ASoC: Constify snd_soc_dai_ops variables ASoC: da732x: Merge da732x_dai1_ops and da732x_dai2_ops ASoC: sirf-audio-codec: Staticise local symbols ASoC: Constify dev_pm_ops variables ASoC: cs4349: Fix max_register setting for cs4349_regmap ASoC: cs4349: Constify cs4349_regmap ASoC: cs4349: Fix setting digital interface format ASoC: cs4349: Set .writeable_reg for cs4349_regmap ASoC: cs4349: Drop platform data support ASoC: cs4349: Remove unneeded NULL test for cs4349->reset_gpio ASoC: cs4349: Fix up setting PWR_DWN bit ASoC: cs35l32: Fix define for CS35L32_SDOUT_3ST ASoC: cs35l32: Remove unneeded NULL test for cs35l32->reset_gpio ASoC: cs42l73: Fix mask for setting CS42L73_SP_3ST bit ASoC: rt5677: Remove NULL test for desc before gpiod_set_value_cansleep call ASoC: tas2552: Fix off-by-one for max_register setting ASoC: tas5086: Improve the logic for de-emphasis sampling rate selection ASoC: pcm1681: Improve the logic for de-emphasis sampling rate selection ASoC: sta529: Remove redundant I2C_FUNC_SMBUS_BYTE_DATA functionality check ASoC: cs4265: CS4265_INT_STATUS is readable register ASoC: Use params_width() at appropriate places ASoC: rt5677: Return error if devm_gpiod_get_optional return ERR_PTR ASoC: tas2552: Remove unneeded NULL test for tas2552->enable_gpio ASoC: max98088: Get rid of max98088_access table ASoC: max98095: Get rid of max98095_access table ASoC: wm8983: Get rid of wm8983_access_masks table ASoC: max98090: Simplify max98090_readable_register implementation ASoC: wm8741: Simplify wm8741_set_dai_sysclk implementation ASoC: wm8731: Drop wm8731_writeable function ASoC: wm8753: Drop wm8753_writeable function ASoC: topology: Use kcalloc instead of kzalloc for array allocation ASoC: adav80x: Remove .read_flag_mask setting from adav80x_regmap_config ASoC: cs35l32: Use case range at appropriate places ASoC: cs42l52: Use case range at appropriate place ASoC: cs42l56: Use case range at appropriate place ASoC: cs42l73: Use case range at appropriate place ASoC: wm8741: Drop misleading readable_reg callback implementation ASoC: Set missing card owner field ASoC: davinci: Convert to use devm_ioremap_resource ASoC: omap-mcbsp: Convert to use devm_ioremap_resource ASoC: SPEAr: Convert to use devm_ioremap_resource ASoC: tegra: Fix unused variable 'spdif' warning ASoC: tegra: Use devm_ioremap_resource instead of open code ASoC: ics43432: Fix module license ASoC: pxa-ssp: Convert to devm_snd_soc_register_component ASoC: pxa2xx-i2s: Convert to devm_snd_soc_register_component ASoC: sh: ssi: Convert to devm_snd_soc_register_component ASoC: qcom: Constify asoc_qcom_lpass_cpu_dai_ops ASoC: au1x: Convert to devm_snd_soc_register_platform ASoC: blackfin: Convert to devm_snd_soc_register_platform ASoC: nuc900: Convert to devm_snd_soc_register_platform ASoC: pxa: Convert to devm_snd_soc_register_platform ASoC: txx9: Convert to devm_snd_soc_register_platform ASoC: spear_pcm: Use devm_snd_dmaengine_pcm_register to fix resource leak ASoC: sh: dma-sh7760: Convert to devm_snd_soc_register_platform ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource ASoC: sti_uniperif: Ensure component is unregistered when unload module ASoC: zx296702-i2s: Fix resource leak when unload module Bard Liao (4): ASoC: rt5645: move RT5645 muxes to rt5645_specific_dapm_widgets ASoC: add rt298 codec driver ASoC: rt298: remove meanless pr_info ASoC: rl6231: add pll preset table Ben Zhang (3): ASoC: rt5677: Switch to use descriptor-based gpiod API ASoC: rt5677: Switch to use unified device property API ASoC: rt5677: Allow arbitrary block read/write via SPI Charles Keepax (8): ASoC: arizona: Implement stability check for EQ coefficients ASoC: arizona: Implement stability check for LHPF coefficients mfd: wm5110: Add registers for custom write sequence triggers mfd: wm5110: Add register patch for rev E and above ASoC: wm5110: Add special DRE on/off handling for the headphone path ASoC: arizona: Use the more idiomatic params_width ASoC: arizona: Fix error path in codec probe ASoC: arizona: Poll for FLL clock OK rather than use interrupts Dan Carpenter (3): ASoC: fsl: fsl_spdif: signedness bug in fsl_spdif_startup() ASoC: max98925: reading beyond the end of the array ASoC: sti: error handling bug in sti_uniperiph_cpu_dai_of() Fabio Estevam (6): ASoC: fsl: fsl_asrc: Check for clk_prepare_enable() error ASoC: fsl: fsl_spdif: Check for clk_prepare_enable() error ASoC: max98090: Check for clk_prepare_enable() error ASoC: wm8904: Check for clk_prepare_enable() error ASoC: wm8731: Check for clk_prepare_enable() error ASoC: max98095: Check for clk_prepare_enable() error Fang, Yang A (2): ASoC: ts3a227e: do not report jack status when there is i2c read err ASoC: max98090: Fix sequencing when starting additional routes Geert Uytterhoeven (2): ASoC: fsi: Remove obsolete sh_fsi2 platform_device_id entry ASoC: rsnd: Silence DMA slave ID compile warning on 64-bit Hardik T Shah (1): ASoC: Intel: Skylake: Add helpers for SRC and converter modules Harsha Priya (1): ASoC: ssm4567: Added ACPI entry for SSM4567 codec Jarkko Nikula (6): ASoC: rt286: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5640: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5645: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5651: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5670: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5677: Prefix hexadecimal ID register value with 0x in error print Javier Martinez Canillas (1): ASoC: Export OF module alias information in missing codec drivers Jean-Francois Moine (1): ASoC: kirkwood: prevent double streaming Jeeja KP (11): ASoC: Intel: Add Skylake HDA platform driver ASoC: Intel: Add Skylake HDA audio driver ASoC: Intel: Add makefile support for SKL driver ASoC: Intel: Add support for decoupled mode in skl driver ASoC: Intel: Skylake: Add NHLT support to get BE config ASoC: Intel: Skylake: Add dsp and ipc init helpers ASoC: Intel: Skylake: Fix the NHLT rate size ASoC: Intel: Skylake: Use acpi header for NHLT header ASoC: Intel: Skylake: Add helpers for DSP module configuration ASoC: Intel: Skylake: Add DSP module init and binding routines ASoC: Intel: Skylake: Add pipe management helpers Joe Perches (1): ASoC: atmel_ssc_dai: Correct misuse of 0x%<decimal> John Lin (1): ASoC: rt5640: fix line out no sound issue Jonathan Corbet (1): ASoC: Clean up docbook warnings Julia Lawall (1): ASoC: ux500: adjust devm usage Junjie Mao (1): ASoC: Intel: fix runtime pm imbalance on error Jyri Sarha (1): ASoC: omap-hdmi-audio: Set buffer bytes step constraint to 128 Kenneth Westfield (2): ASoC: qcom: move board Kconfig deps to parent config ASoC: qcom: remove Kconfig deps from variant configs Koro Chen (5): ASoC: dpcm: Add checks of playback/capture before dpcm_get_be ASoC: rt5645: Fix missing free_irq ASoC: rt5645: Add regulator support ASoC: mediatek: Add suspend/resume callbacks ASoC: mediatek: Remove AIF widgets for backend DAIs Krzysztof Kozlowski (2): ASoC: mediatek: Drop owner assignment from platform_driver ASoC: drivers: Drop owner assignment from i2c_driver Kuninori Morimoto (36): regmap: add force_write option on _regmap_update_bits() regmap: add regmap_write_bits() regmap: add regmap_fields_force_write() ASoC: rsnd: gen: add rsnd_force_write() ASoC: core: add snd_soc_of_parse_audio_prefix() ASoC: rsrc-card: use snd_soc_of_parse_audio_route/prefix for routing ASoC: rsnd: remove unnecessary 'out of memory' message from SSI ASoC: rsnd: remove unnecessary 'out of memory' message from SRC ASoC: rsnd: remove unnecessary 'out of memory' message from DVC ASoC: rsnd: rename BUSIF_DALIGN to SSI_BUSIF_DALIGN ASoC: rsnd: rename INT_ENABLE to SSI_INT_ENABLE ASoC: rsnd: fixup each module counter on __rsnd_mod_call() ASoC: rsnd: add workaround for SRC sync convert + DVC ASoC: rsnd: rsnd_mod_id() return -1 if mod was NULL ASoC: rsnd: move DVC specific macro into dvc.c ASoC: rsnd: move SRC specific macro into src.c ASoC: rsnd: dvc: make sure DVC soft reset ASoC: rsnd: src: make sure SRC soft reset ASoC: rsnd: enable module multi connection ASoC: rsnd: rename rsnd_path_parse/break() into add/remove ASoC: rsnd: add rsnd_path_parse() for CTU/MIX/DVC route setting ASoC: rsnd: add rsnd_dvc_initialize_lock/unlock() ASoC: rsnd: add rsnd_src_initialize_lock/unlock() ASoC: rsnd: tidyup ADINR function name ASoC: rsnd: add rsnd_get_adinr_chan() ASoC: rsnd: tidyup data align position ASoC: rsnd: show debug message for SSI/SRC/DVC connection ASoC: rsnd: tidyup rsnd_dma_ops definition place ASoC: rsnd: check the Gen1 at the beginning of DVC probe ASoC: rsnd: dma: add DMA name on .ops ASoC: rsnd: add rsnd_io_to_mod() ASoC: rsnd: tidyup SRC position on each code ASoC: rsnd: update Audio DMA path search method ASoC: rsnd: add CTU (Channel Transfer Unit) prototype support ASoC: rsnd: add MIX (Mixer) support ASoC: rsnd: tidyup parameter assignment position Lars-Peter Clausen (83): ASoC: rt5645: Replace direct snd_soc_codec dapm field access ASoC: tas571x: Replace direct snd_soc_codec dapm field access ASoC: omap3pandora: Use card DAPM context to access widgets ASoC: cht_bsw_max98090_ti: Use card DAPM context to access widgets ASoC: dapm_widget_show_codec: Use component instead of CODEC ASoC: Remove snd_soc_codec dapm field ASoC: Prevent components from being bound to multiple cards ASoC: Use card field to indicate whether a component is bound ASoC: uda134x: Move state struct allocation to bus probe ASoC: uda134x: Explicitly handle in-data addresses ASoC: uda134x: Convert to regmap ASoC: uda134x: Use regmap_update_bits() were appropriate ASoC: max98925: Remove control_data initialization ASoC: max9877: Make driver global regmap struct local ASoC: max9877: Convert to component ASoC: max9768: Use managed gpio request ASoC: max9768: Convert to component ASoC: cs4349: Fix suspend/resume ASoC: cs4349: Don't use rtd->codec ASoC: dapm: Add helper function to free a widget ASoC: dapm: Avoid duplicating immutable strings ALSA: ac97: Add helper function to reset the AC97 device ASoC: ac97: Add support for resetting device before registration ASoC: ad1980: Use core AC'97 reset helper ASoC: stac9766: Use core reset helper ASoC: wm9705: Use core AC'97 reset helper ASoC: wm9712: Use core AC'97 reset helper ASoC: wm9713: Use core AC'97 reset helper ASoC: dapm: Drop always true checks ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets() ASoC: dapm: Add widget path iterators ASoC: wm8804: Drop duplicate const ASoC: lpass-ipq806x: Staticise local symbols ASoC: rcar ctu: Staticise local symbols ASoC: uniperf: Add missing __iomem annotation ASoC: uniperf: Staticise local symbols ASoC: xtfpga-i2s: Add missing __rcu annotation ASoC: 88pm860x: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: ab8500: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: adau1373: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: alc5623: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: alc5632: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l52: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l56: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l73: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da7210: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da7213: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da9055: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: jz4740: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9768: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98088: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98090: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98095: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9850: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9877: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5631: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5640: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5645: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5651: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5670: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5677: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: sgtl5000: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: ssm2602: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: tpa6130a2: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: twl4030: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: uda1380: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8350: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8400: Don't use range container for TLV with one entry ASoC: wm8737: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8753: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8961: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8962: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8990: Don't use range container for TLV with one entry ASoC: wm8991: Don't use range container for TLVs with one entry ASoC: wm8993: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9081: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9090: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9713: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm_hubs: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: dapm: dapm_dai_get_connected_widgets: Fix missing mutex unlock ASoC: dapm: Consolidate input and output path handling ASoC: dapm: Consolidate path trace events ASoC: ssm4567: Add sense support Luis de Bethencourt (1): ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers Maciej S. Szmigiero (6): ASoC: fsl_ssi: enable IPG clock during AC'97 reg access ASoC: fsl_ssi: AC'97 DAI driver needs probe method too ASoC: fsl_ssi: enable AC'97 asymmetric rates ASoC: fsl_ssi: add AC'97 ops setting check and cleanup ASoC: fsl_ssi: instantiate AC'97 CODEC ASoC: fsl_ssi: adjust set DAI format in AC'97 mode Marek Belisko (2): ASoC: Add gtm601 codec driver Documentation: vendor-prefixes: Add option prefix Mark Brown (43): Merge branch 'topic/codec-dapm' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge tag 'regmap-force' into asoc-rcar ASoC: sti-sas: Remove spurious dependency on SND_SOC_STI Merge branches 'topic/rt298', 'topic/ts3a227e' and 'topic/cs42xx8' of git://git.kernel.org/.../broonie/sound into asoc-owner Merge branches 'topic/owner' and 'topic/reg-default' of git://git.kernel.org/.../broonie/sound into asoc-max9877 Merge branch 'topic/of-name' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge tag 'regmap-seq-delay-api' of git://git.kernel.org/.../broonie/regmap into asoc-wm5110 Merge tag 'regmap-seq-delay-api' of git://git.kernel.org/.../broonie/regmap into asoc-reg-default Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 ASoC: sti-uniperf: Fix implicit inclusion of pinctrl Merge branch 'fix/dapm' of git://git.kernel.org/.../broonie/sound into asoc-dapm Merge branch 'topic/owner' of git://git.kernel.org/.../broonie/sound into asoc-ssm4567 Merge branch 'fix/ssm4567' of git://git.kernel.org/.../broonie/sound into asoc-ssm4567 Merge tag 'v4.2-rc6' into asoc-topology Merge branch 'topic/gtm601' of git://git.kernel.org/.../broonie/sound into asoc-ics43432 Merge branch 'topic/ics43432' of git://git.kernel.org/.../broonie/sound into asoc-gtm601 Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/adav80x', 'asoc/fix/arizona', 'asoc/fix/max98090', 'asoc/fix/rt5640', 'asoc/fix/samsung' and 'asoc/fix/wm8994' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branch 'asoc/topic/ssm4567' into asoc-next Merge remote-tracking branches 'asoc/topic/88pm860x', 'asoc/topic/ac97', 'asoc/topic/ak4542', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/au1x', 'asoc/topic/bcm2835', 'asoc/topic/blackfin' and 'asoc/topic/card' into asoc-next Merge remote-tracking branches 'asoc/topic/const', 'asoc/topic/cs35l32', 'asoc/topic/cs4265' and 'asoc/topic/cs42l52' into asoc-next Merge remote-tracking branches 'asoc/topic/cs42l56', 'asoc/topic/cs42l73', 'asoc/topic/cs4349' and 'asoc/topic/da732x' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/davinci-vcif', 'asoc/topic/doc' and 'asoc/topic/dpcm' into asoc-next Merge remote-tracking branches 'asoc/topic/fsi', 'asoc/topic/fsl', 'asoc/topic/fsl-asrc', 'asoc/topic/fsl-card' and 'asoc/topic/fsl-sai' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/fsl-ssi', 'asoc/topic/gtm601', 'asoc/topic/ics43432' and 'asoc/topic/ids' into asoc-next Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/kirkwood', 'asoc/topic/lm49453', 'asoc/topic/max9768' and 'asoc/topic/max98088' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/max98357a', 'asoc/topic/max9877' and 'asoc/topic/max98925' into asoc-next Merge remote-tracking branches 'asoc/topic/mediatek', 'asoc/topic/mtk', 'asoc/topic/nuc900', 'asoc/topic/of-name' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/pm-ops', 'asoc/topic/pxa' and 'asoc/topic/qcom' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/reg-default', 'asoc/topic/rl6231', 'asoc/topic/rockchip' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt298', 'asoc/topic/rt5640', 'asoc/topic/rt5651' and 'asoc/topic/rt5670' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5677', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf-codec' and 'asoc/topic/spear' into asoc-next Merge remote-tracking branches 'asoc/topic/ssm2518', 'asoc/topic/sta529', 'asoc/topic/sti' and 'asoc/topic/sti-sas' into asoc-next Merge remote-tracking branches 'asoc/topic/tas2552', 'asoc/topic/tas5086', 'asoc/topic/tegra', 'asoc/topic/tlv' and 'asoc/topic/topology' into asoc-next Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/uda134x', 'asoc/topic/ux500' and 'asoc/topic/width' into asoc-next Merge remote-tracking branches 'asoc/topic/wm0010', 'asoc/topic/wm5100', 'asoc/topic/wm5110', 'asoc/topic/wm8004' and 'asoc/topic/wm8731' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8741', 'asoc/topic/wm8753', 'asoc/topic/wm8904', 'asoc/topic/wm8960' and 'asoc/topic/wm8983' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/xtensa' and 'asoc/topic/zx296702' into asoc-next Masanari Iida (2): ASoC: Fix warning while make xmldocs caused by soc-core.c ASoC: Fix warning caused by a typo in comments of snd_soc_add_platform() Mengdong Lin (4): ASoC: topology: Change pass number of DAI smaller than graph ASoC: topology: Bind vendor specific kcontrol handlers before standard ones ASoC: topology: Reduce arguments of soc_tplg_kcontrol_bind_io() ASoC: topology: Add support for TLV bytes controls Nariman Poushin (2): regmap: Use reg_sequence for multi_reg_write / register_patch ASoC: wm5110: Use reg_sequence for multi_reg_write/register_patch Nicolas Boichat (5): ASoC: rt5645: Remove irq_jack_detection function ASoC: rt5645: Update dapm pins when the card is not instantiated yet ASoC: rt5645: Simplify rt5645_enable_push_button_irq ASoC: rt5645: Remove unused rt5645 variable ASoC: rt5645: Remove return value in jack detect work Nicolin Chen (1): ASoC: fsl-asoc-card: Specify codec_dai_name for DAI links Nik Nyby (1): ASoC: codecs: max98088: fix typo in constant Nikesh Oswal (1): ASoC: arizona: Fix gain settings of FLL in free-run mode Oder Chiou (7): ASoC: Add function "rl6231_get_pre_div" to correct the dmic clock calculation ASoC: rt5645: Modify the headphone depop and calibration function to prevent the pop sound in the booting time ASoC: rt5645: Modify the jack detection function to prevent the pop sound while the jack plug in ASoC: rt5645: Remove the incorrect setting of the JD mode ASoC: rt5645: Add the register RT5645_CHARGE_PUMP to readable check function ASoC: rt5645: Add struct dmi_system_id "Google Celes" for chrome platform ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning Omair M Abdullah (1): ASoC: Intel: Skylake: Print error code in IPC for SKL Ricard Wanderlof (3): ASoC: ssm2518: Add explicit device tree support ASoC: ics43432: Add codec driver for InvenSense ICS-43432 ASoC: ics43432: Fix build errors Sergey Kiselev (1): ASoC: wm8731: initialize the hardware when loading the codec driver Shengjiu Wang (2): ASoC: fsl: Add dedicated DMA buffer size for each cpu dai ASoC: core: Don't probe the component which is dummy Subhransu S. Prusty (9): ASoC: Intel: Define SKL ADSP IPC and general purpose registers ASoC: Intel: Reorganize the common dsp structure ASoC: Intel: Add helper to poll register for DSP status ASoC: Intel: Add helper to update register bits with attr RWC ASoC: Intel: Add Skylake IPC library ASoC: Intel: Add DSP init and boot up functionality for SKL ASoC: Intel: Skylake: Add dsp loader ops ASoC: Intel: Skylake: Process code loader DMA interrupt ASoC: Intel: Skylake: Add SKL DSP initialization Tim Howe (1): ASoC: cs4349: Add support for Cirrus Logic CS4349 Vaishali Thakkar (7): ASoC: xtensa: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: tegra: Use devm_clk_get ASoC: tegra: Convert to managed resources ASoC: davinci-vcif: Use devm_snd_soc_register_component ASoC: tegra: Revert part of "ASoC: tegra: Convert to managed resources" ASoC: rockchip: i2s: Adjust devm usage ASoC: samsung: Remove redundant arndale_audio_remove Vinod Koul (7): ASoC: intel: kconfig - Move DW_DMAC_CORE dependency to machines ASoC: intel - kconfig: remove SND_SOC_INTEL_SST prompt ASoC: intel - atom: statify sst_slot_enum_info ASoC: intel - atom: fix sst_platform_compr_ops declaration ASoC: Intel: Skylake: Add code loader DMA registers ASoC: Intel: Skylake: add code loader DMA operations ASoC: Intel: Skylake: Add code loader DMA APIs Xing Zheng (1): ASoC: rockchip: fix a misjudgement by return Xiubo Li (1): ASoC: fsl: fix typos for sound/soc/fsl/* Zidan Wang (4): ASoC: wm8960: update pll and clock setting function ASoC: fsl-sai: add 32 bit word length support ASoC: fsl-asoc-card: add wm8960 support ASoC: wm8994: fix add dynamic path error kbuild test robot (8): ASoC: fix platform_no_drv_owner.cocci warnings ASoC: gtm601_dai can be static ASoC: sti-sas: fix platform_no_drv_owner.cocci warnings ASoC: sti: sti_uniperiph_dai_create_ctrl() can be static ASoC: cs4349: fix platform_no_drv_owner.cocci warnings ASoC: rockchip: fix platform_no_drv_owner.cocci warnings ASoC: rockchip: fix platform_no_drv_owner.cocci warnings ASoC: ics43432: fix platform_no_drv_owner.cocci warnings zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec Documentation/DocBook/alsa-driver-api.tmpl | 2 +- Documentation/devicetree/bindings/sound/cs4349.txt | 19 + .../devicetree/bindings/sound/ics43432.txt | 17 + .../devicetree/bindings/sound/max98357a.txt | 6 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 22 + .../bindings/sound/renesas,rsrc-card.txt | 7 + .../bindings/sound/rockchip-max98090.txt | 19 + .../devicetree/bindings/sound/rockchip-rt5645.txt | 17 + .../devicetree/bindings/sound/st,sti-asoc-card.txt | 155 +++ .../devicetree/bindings/vendor-prefixes.txt | 2 + drivers/base/regmap/internal.h | 2 +- drivers/base/regmap/regmap.c | 73 +- drivers/gpu/drm/i2c/adv7511.c | 2 +- drivers/input/misc/drv260x.c | 6 +- drivers/input/misc/drv2665.c | 2 +- drivers/input/misc/drv2667.c | 4 +- drivers/mfd/arizona-core.c | 2 +- drivers/mfd/twl6040.c | 2 +- drivers/mfd/wm5102-tables.c | 6 +- drivers/mfd/wm5110-tables.c | 22 +- drivers/mfd/wm8994-core.c | 8 +- drivers/mfd/wm8997-tables.c | 2 +- include/linux/mfd/arizona/registers.h | 37 + include/linux/regmap.h | 28 +- include/sound/ac97_codec.h | 2 + include/sound/rcar_snd.h | 14 + include/sound/rt298.h | 20 + include/sound/soc-dapm.h | 84 +- include/sound/soc-topology.h | 13 +- include/sound/soc.h | 29 +- include/trace/events/asoc.h | 53 +- sound/ac97_bus.c | 62 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/au1x/dbdma2.c | 11 +- sound/soc/au1x/dma.c | 11 +- sound/soc/au1x/psc-i2s.c | 16 +- sound/soc/bcm/bcm2835-i2s.c | 2 + sound/soc/blackfin/bf5xx-ac97-pcm.c | 10 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 10 +- sound/soc/blackfin/bfin-eval-adau1x61.c | 1 + sound/soc/codecs/88pm860x-codec.c | 32 +- sound/soc/codecs/Kconfig | 25 + sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 7 +- sound/soc/codecs/ad1980.c | 36 +- sound/soc/codecs/adau1373.c | 22 +- sound/soc/codecs/adau1701.c | 1 - sound/soc/codecs/adau1761-i2c.c | 1 - sound/soc/codecs/adau1781-i2c.c | 1 - sound/soc/codecs/adau1977-i2c.c | 1 - sound/soc/codecs/adav803.c | 1 - sound/soc/codecs/adav80x.c | 3 +- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 34 +- sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/alc5623.c | 8 +- sound/soc/codecs/alc5632.c | 10 +- sound/soc/codecs/arizona.c | 129 +- sound/soc/codecs/arizona.h | 20 +- sound/soc/codecs/cs35l32.c | 60 +- sound/soc/codecs/cs35l32.h | 2 +- sound/soc/codecs/cs4265.c | 19 +- sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs4271-i2c.c | 1 - sound/soc/codecs/cs42l51-i2c.c | 1 - sound/soc/codecs/cs42l52.c | 65 +- sound/soc/codecs/cs42l56.c | 71 +- sound/soc/codecs/cs42l73.c | 115 +- sound/soc/codecs/cs42xx8-i2c.c | 4 +- sound/soc/codecs/cs42xx8.c | 19 +- sound/soc/codecs/cs42xx8.h | 1 + sound/soc/codecs/cs4349.c | 392 ++++++ sound/soc/codecs/cs4349.h | 136 +++ sound/soc/codecs/da7210.c | 29 +- sound/soc/codecs/da7213.c | 18 +- sound/soc/codecs/da732x.c | 15 +- sound/soc/codecs/da9055.c | 19 +- sound/soc/codecs/gtm601.c | 95 ++ sound/soc/codecs/ics43432.c | 76 ++ sound/soc/codecs/isabelle.c | 11 +- sound/soc/codecs/jz4740.c | 7 +- sound/soc/codecs/lm4857.c | 1 - sound/soc/codecs/lm49453.c | 19 +- sound/soc/codecs/max9768.c | 71 +- sound/soc/codecs/max98088.c | 329 +---- sound/soc/codecs/max98088.h | 2 +- sound/soc/codecs/max98090.c | 171 ++- sound/soc/codecs/max98090.h | 1 + sound/soc/codecs/max98095.c | 354 +----- sound/soc/codecs/max98357a.c | 25 +- sound/soc/codecs/max9850.c | 8 +- sound/soc/codecs/max9877.c | 33 +- sound/soc/codecs/max98925.c | 6 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/ml26124.c | 3 +- sound/soc/codecs/pcm1681.c | 14 +- sound/soc/codecs/pcm512x-i2c.c | 1 - sound/soc/codecs/pcm512x.c | 4 +- sound/soc/codecs/rl6231.c | 104 +- sound/soc/codecs/rl6231.h | 1 + sound/soc/codecs/rt286.c | 11 +- sound/soc/codecs/rt298.c | 1271 ++++++++++++++++++++ sound/soc/codecs/rt298.h | 206 ++++ sound/soc/codecs/rt5631.c | 8 +- sound/soc/codecs/rt5640.c | 59 +- sound/soc/codecs/rt5645.c | 443 +++---- sound/soc/codecs/rt5645.h | 27 +- sound/soc/codecs/rt5651.c | 19 +- sound/soc/codecs/rt5670.c | 21 +- sound/soc/codecs/rt5677-spi.c | 233 +++- sound/soc/codecs/rt5677-spi.h | 8 +- sound/soc/codecs/rt5677.c | 159 +-- sound/soc/codecs/rt5677.h | 5 +- sound/soc/codecs/sgtl5000.c | 8 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/sirf-audio-codec.c | 4 +- sound/soc/codecs/ssm2518.c | 10 +- sound/soc/codecs/ssm2602-i2c.c | 1 - sound/soc/codecs/ssm2602.c | 7 +- sound/soc/codecs/ssm4567.c | 34 +- sound/soc/codecs/sta32x.c | 1 - sound/soc/codecs/sta350.c | 1 - sound/soc/codecs/sta529.c | 4 - sound/soc/codecs/stac9766.c | 57 +- sound/soc/codecs/sti-sas.c | 628 ++++++++++ sound/soc/codecs/tas2552.c | 20 +- sound/soc/codecs/tas2552.h | 2 +- sound/soc/codecs/tas5086.c | 11 +- sound/soc/codecs/tas571x.c | 2 +- sound/soc/codecs/tfa9879.c | 3 +- sound/soc/codecs/tlv320aic31xx.c | 3 +- sound/soc/codecs/tlv320aic32x4.c | 1 - sound/soc/codecs/tlv320aic3x.c | 3 +- sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/tpa6130a2.c | 15 +- sound/soc/codecs/ts3a227e.c | 48 +- sound/soc/codecs/twl4030.c | 7 +- sound/soc/codecs/uda134x.c | 180 ++- sound/soc/codecs/uda134x.h | 2 - sound/soc/codecs/uda1380.c | 15 +- sound/soc/codecs/wm0010.c | 3 +- sound/soc/codecs/wm1250-ev1.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm2200.c | 9 +- sound/soc/codecs/wm5100.c | 13 +- sound/soc/codecs/wm5102.c | 27 +- sound/soc/codecs/wm5110.c | 315 ++++- sound/soc/codecs/wm8350.c | 7 +- sound/soc/codecs/wm8400.c | 5 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8728.c | 1 - sound/soc/codecs/wm8731.c | 88 +- sound/soc/codecs/wm8737.c | 8 +- sound/soc/codecs/wm8741.c | 63 +- sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 14 +- sound/soc/codecs/wm8776.c | 5 +- sound/soc/codecs/wm8804-i2c.c | 1 - sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 5 +- sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 221 +++- sound/soc/codecs/wm8960.h | 1 + sound/soc/codecs/wm8961.c | 8 +- sound/soc/codecs/wm8962.c | 21 +- sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8983.c | 81 +- sound/soc/codecs/wm8985.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 6 +- sound/soc/codecs/wm8991.c | 53 +- sound/soc/codecs/wm8993.c | 12 +- sound/soc/codecs/wm8994.c | 18 +- sound/soc/codecs/wm8995.c | 1 - sound/soc/codecs/wm8996.c | 11 +- sound/soc/codecs/wm8997.c | 20 +- sound/soc/codecs/wm9081.c | 10 +- sound/soc/codecs/wm9090.c | 22 +- sound/soc/codecs/wm9705.c | 40 +- sound/soc/codecs/wm9712.c | 45 +- sound/soc/codecs/wm9713.c | 55 +- sound/soc/codecs/wm9713.h | 2 - sound/soc/codecs/wm_hubs.c | 7 +- sound/soc/davinci/davinci-i2s.c | 25 +- sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-vcif.c | 14 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 16 +- sound/soc/fsl/fsl_asrc.c | 25 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_sai.h | 15 +- sound/soc/fsl/fsl_spdif.c | 25 +- sound/soc/fsl/fsl_ssi.c | 68 +- sound/soc/fsl/imx-pcm-dma.c | 25 +- sound/soc/fsl/imx-pcm.h | 9 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/generic/simple-card.c | 9 + sound/soc/intel/Kconfig | 29 +- sound/soc/intel/Makefile | 1 + sound/soc/intel/atom/sst-atom-controls.c | 6 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 - sound/soc/intel/atom/sst-mfld-platform.h | 1 + sound/soc/intel/atom/sst/sst_drv_interface.c | 9 +- sound/soc/intel/atom/sst/sst_ipc.c | 3 +- sound/soc/intel/boards/byt-max98090.c | 1 + sound/soc/intel/boards/byt-rt5640.c | 1 + sound/soc/intel/boards/bytcr_rt5640.c | 1 + sound/soc/intel/boards/cht_bsw_max98090_ti.c | 19 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 + sound/soc/intel/boards/cht_bsw_rt5672.c | 1 + sound/soc/intel/common/sst-dsp-priv.h | 23 +- sound/soc/intel/common/sst-dsp.c | 71 ++ sound/soc/intel/common/sst-dsp.h | 6 + sound/soc/intel/skylake/Makefile | 9 + sound/soc/intel/skylake/skl-messages.c | 884 ++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 140 +++ sound/soc/intel/skylake/skl-nhlt.h | 106 ++ sound/soc/intel/skylake/skl-pcm.c | 916 ++++++++++++++ sound/soc/intel/skylake/skl-sst-cldma.c | 327 +++++ sound/soc/intel/skylake/skl-sst-cldma.h | 251 ++++ sound/soc/intel/skylake/skl-sst-dsp.c | 342 ++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 145 +++ sound/soc/intel/skylake/skl-sst-ipc.c | 771 ++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 125 ++ sound/soc/intel/skylake/skl-sst.c | 280 +++++ sound/soc/intel/skylake/skl-topology.h | 286 +++++ sound/soc/intel/skylake/skl-tplg-interface.h | 88 ++ sound/soc/intel/skylake/skl.c | 536 +++++++++ sound/soc/intel/skylake/skl.h | 84 ++ sound/soc/kirkwood/kirkwood-dma.c | 4 + sound/soc/mediatek/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 8 - sound/soc/mediatek/mtk-afe-pcm.c | 89 +- sound/soc/nuc900/nuc900-pcm.c | 9 +- sound/soc/omap/mcbsp.c | 20 +- sound/soc/omap/omap-hdmi-audio.c | 10 +- sound/soc/omap/omap3pandora.c | 6 +- sound/soc/pxa/mmp-pcm.c | 9 +- sound/soc/pxa/pxa-ssp.c | 11 +- sound/soc/pxa/pxa2xx-i2s.c | 11 +- sound/soc/pxa/pxa2xx-pcm.c | 9 +- sound/soc/qcom/Kconfig | 7 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass.h | 2 +- sound/soc/rockchip/Kconfig | 19 + sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_i2s.c | 8 +- sound/soc/rockchip/rockchip_max98090.c | 236 ++++ sound/soc/rockchip/rockchip_rt5645.c | 225 ++++ sound/soc/samsung/arndale_rt5631.c | 11 +- sound/soc/samsung/snow.c | 1 + sound/soc/sh/dma-sh7760.c | 9 +- sound/soc/sh/fsi.c | 1 - sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 195 ++- sound/soc/sh/rcar/ctu.c | 171 +++ sound/soc/sh/rcar/dma.c | 128 +- sound/soc/sh/rcar/dvc.c | 73 +- sound/soc/sh/rcar/gen.c | 33 +- sound/soc/sh/rcar/mix.c | 200 +++ sound/soc/sh/rcar/rsnd.h | 111 +- sound/soc/sh/rcar/rsrc-card.c | 22 +- sound/soc/sh/rcar/src.c | 117 +- sound/soc/sh/rcar/ssi.c | 4 +- sound/soc/sh/ssi.c | 12 +- sound/soc/soc-ac97.c | 30 +- sound/soc/soc-core.c | 86 +- sound/soc/soc-dapm.c | 503 ++++---- sound/soc/soc-pcm.c | 16 +- sound/soc/soc-topology.c | 145 +-- sound/soc/spear/spdif_in.c | 20 +- sound/soc/spear/spear_pcm.c | 2 +- sound/soc/sti/Kconfig | 11 + sound/soc/sti/Makefile | 4 + sound/soc/sti/sti_uniperif.c | 254 ++++ sound/soc/sti/uniperif.h | 1229 +++++++++++++++++++ sound/soc/sti/uniperif_player.c | 1110 +++++++++++++++++ sound/soc/sti/uniperif_reader.c | 362 ++++++ sound/soc/tegra/tegra20_das.c | 23 +- sound/soc/tegra/tegra20_i2s.c | 23 +- sound/soc/tegra/tegra20_spdif.c | 47 +- sound/soc/tegra/tegra30_ahub.c | 80 +- sound/soc/tegra/tegra30_i2s.c | 23 +- sound/soc/txx9/txx9aclc.c | 10 +- sound/soc/ux500/ux500_msp_dai.c | 29 +- sound/soc/xtensa/xtfpga-i2s.c | 8 +- sound/soc/zte/zx296702-i2s.c | 6 +- 301 files changed, 16003 insertions(+), 3744 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-rt5645.txt create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt create mode 100644 include/sound/rt298.h create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h create mode 100644 sound/soc/codecs/gtm601.c create mode 100644 sound/soc/codecs/ics43432.c create mode 100644 sound/soc/codecs/rt298.c create mode 100644 sound/soc/codecs/rt298.h create mode 100644 sound/soc/codecs/sti-sas.c create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-messages.c create mode 100644 sound/soc/intel/skylake/skl-nhlt.c create mode 100644 sound/soc/intel/skylake/skl-nhlt.h create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c create mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h create mode 100644 sound/soc/intel/skylake/skl-sst.c create mode 100644 sound/soc/intel/skylake/skl-topology.h create mode 100644 sound/soc/intel/skylake/skl-tplg-interface.h create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c create mode 100644 sound/soc/sh/rcar/ctu.c create mode 100644 sound/soc/sh/rcar/mix.c create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_uniperif.c create mode 100644 sound/soc/sti/uniperif.h create mode 100644 sound/soc/sti/uniperif_player.c create mode 100644 sound/soc/sti/uniperif_reader.c

2 6

[alsa-devel] [PATCH] ASoC: Make SND_SOC_STI_SAS depend on RESET_CONTROLLER
by Axel Lin 31 Aug '15

31 Aug '15

(devm_)reset_control_get function is available only when RESET_CONTROLLER is set. Make SND_SOC_STI_SAS depend on RESET_CONTROLLER to fix build error if !RESET_CONTROLLER. Reported-by: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..f8e0022 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -106,7 +106,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS - select SND_SOC_STI_SAS + select SND_SOC_STI_SAS if RESET_CONTROLLER select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C @@ -634,6 +634,7 @@ config SND_SOC_STAC9766 config SND_SOC_STI_SAS tristate "codec Audio support for STI SAS codec" + depends on RESET_CONTROLLER config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: ics43432: Fix build errors
by Ricard Wanderlof 31 Aug '15

31 Aug '15

Fix build errors Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- Not sure how the incorrect name or the misspelled license string got through my local build, perhaps because I've been developing this for an ARM system? Have now set up a tree to build for x86 as well, where the errors manifest themselves, to avoid this type of stuff happening in the future. sound/soc/codecs/ics43432.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index d217fed..465d591 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -57,7 +57,7 @@ static const struct of_device_id ics43432_ids[] = { { .compatible = "invensense,ics43432", }, { } }; -MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +MODULE_DEVICE_TABLE(of, ics43432_ids); #endif static struct platform_driver ics43432_driver = { @@ -73,4 +73,4 @@ module_platform_driver(ics43432_driver); MODULE_DESCRIPTION("ASoC ICS43432 driver"); MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 3

[alsa-devel] [PATCH] Revert part of "ASoC: tegra: Convert to managed resources"
by Vaishali Thakkar 31 Aug '15

31 Aug '15

Revert the problematic part of commit 470805eb9f31 ("ASoC: tegra: Convert to managed resources"). Before this commit, PM cleanup was performed after the component was unregistered. But returning directly will skip PM cleanup. So, to be on safe side it is better to use snd_soc_register_component instead of devm_snd_soc_register_component. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/tegra/tegra20_spdif.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index a814da0..66237bc 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -329,9 +329,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = devm_snd_soc_register_component(&pdev->dev, - &tegra20_spdif_component, - &tegra20_spdif_dai, 1); + ret = snd_soc_register_component(&pdev->dev, &tegra20_spdif_component, + &tegra20_spdif_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -341,11 +340,13 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - return ret; + goto err_unregister_component; } return 0; +err_unregister_component: + snd_soc_unregister_component(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_spdif_runtime_suspend(&pdev->dev); @@ -364,6 +365,7 @@ static int tegra20_spdif_platform_remove(struct platform_device *pdev) tegra20_spdif_runtime_suspend(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: tegra: Use devm_ioremap_resource instead of open code
by Axel Lin 31 Aug '15

31 Aug '15

Use devm_ioremap_resource() to simplify the code. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/tegra/tegra20_das.c | 23 ++++------------------ sound/soc/tegra/tegra20_i2s.c | 23 ++++------------------ sound/soc/tegra/tegra20_spdif.c | 22 ++++----------------- sound/soc/tegra/tegra30_ahub.c | 43 +++++++---------------------------------- sound/soc/tegra/tegra30_i2s.c | 23 ++++------------------ 5 files changed, 23 insertions(+), 111 deletions(-) diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index f52600b..89add13 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -133,7 +133,7 @@ static const struct regmap_config tegra20_das_regmap_config = { static int tegra20_das_probe(struct platform_device *pdev) { - struct resource *res, *region; + struct resource *res; void __iomem *regs; int ret = 0; @@ -149,24 +149,9 @@ static int tegra20_das_probe(struct platform_device *pdev) das->dev = &pdev->dev; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err; - } - - region = devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name); - if (!region) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err; - } - - regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); goto err; } diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 05f1c6e..14106fa 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -339,7 +339,7 @@ static const struct regmap_config tegra20_i2s_regmap_config = { static int tegra20_i2s_platform_probe(struct platform_device *pdev) { struct tegra20_i2s *i2s; - struct resource *mem, *memregion; + struct resource *mem; void __iomem *regs; int ret; @@ -362,24 +362,9 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev) } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; + regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); goto err_clk_put; } diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 0809b1e..a0c3640 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -265,7 +265,7 @@ static const struct regmap_config tegra20_spdif_regmap_config = { static int tegra20_spdif_platform_probe(struct platform_device *pdev) { struct tegra20_spdif *spdif; - struct resource *mem, *memregion, *dmareq; + struct resource *mem, *dmareq; void __iomem *regs; int ret; @@ -285,10 +285,9 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - return -ENODEV; - } + regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(regs)) + return PTR_ERR(regs); dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { @@ -296,19 +295,6 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) return -ENODEV; } - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - return -EBUSY; - } - - regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - return -ENOMEM; - } - spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs, &tegra20_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 989b1e8..fef3b9a 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -521,7 +521,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) const struct tegra30_ahub_soc_data *soc_data; struct reset_control *rst; int i; - struct resource *res0, *res1, *region; + struct resource *res0, *res1; void __iomem *regs_apbif, *regs_ahub; int ret = 0; @@ -584,26 +584,12 @@ static int tegra30_ahub_probe(struct platform_device *pdev) } res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res0) { - dev_err(&pdev->dev, "No apbif memory resource\n"); - return -ENODEV; - } + regs_apbif = devm_ioremap_resource(&pdev->dev, res0); + if (IS_ERR(regs_apbif)) + return PTR_ERR(regs_apbif); - region = devm_request_mem_region(&pdev->dev, res0->start, - resource_size(res0), DRV_NAME); - if (!region) { - dev_err(&pdev->dev, "request region apbif failed\n"); - return -EBUSY; - } ahub->apbif_addr = res0->start; - regs_apbif = devm_ioremap(&pdev->dev, res0->start, - resource_size(res0)); - if (!regs_apbif) { - dev_err(&pdev->dev, "ioremap apbif failed\n"); - return -ENOMEM; - } - ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif, &tegra30_ahub_apbif_regmap_config); if (IS_ERR(ahub->regmap_apbif)) { @@ -614,24 +600,9 @@ static int tegra30_ahub_probe(struct platform_device *pdev) regcache_cache_only(ahub->regmap_apbif, true); res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); - if (!res1) { - dev_err(&pdev->dev, "No ahub memory resource\n"); - return -ENODEV; - } - - region = devm_request_mem_region(&pdev->dev, res1->start, - resource_size(res1), DRV_NAME); - if (!region) { - dev_err(&pdev->dev, "request region ahub failed\n"); - return -EBUSY; - } - - regs_ahub = devm_ioremap(&pdev->dev, res1->start, - resource_size(res1)); - if (!regs_ahub) { - dev_err(&pdev->dev, "ioremap ahub failed\n"); - return -ENOMEM; - } + regs_ahub = devm_ioremap_resource(&pdev->dev, res1); + if (IS_ERR(regs_ahub)) + return PTR_ERR(regs_ahub); ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub, &tegra30_ahub_ahub_regmap_config); diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index fe36375..8e55583 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -379,7 +379,7 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) struct tegra30_i2s *i2s; const struct of_device_id *match; u32 cif_ids[2]; - struct resource *mem, *memregion; + struct resource *mem; void __iomem *regs; int ret; @@ -419,24 +419,9 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; + regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); goto err_clk_put; } -- 2.1.0

3 2

[alsa-devel] [asoc:topic/au1x 2/2] sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret'
by kbuild test robot 30 Aug '15

30 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/au1x head: fd63542fc5492b86158dbd53ce5de764f171b1b6 commit: fd63542fc5492b86158dbd53ce5de764f171b1b6 [2/2] ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource config: mips-db1xxx_defconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout fd63542fc5492b86158dbd53ce5de764f171b1b6 # save the attached .config to linux build tree make.cross ARCH=mips All warnings (new ones prefixed by >>): sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe': >> sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable] int ret; ^ vim +/ret +299 sound/soc/au1x/psc-i2s.c 0f83d639 Manuel Lauss 2009-10-31 283 .rates = AU1XPSC_I2S_RATES, 0f83d639 Manuel Lauss 2009-10-31 284 .formats = AU1XPSC_I2S_FMTS, 0f83d639 Manuel Lauss 2009-10-31 285 .channels_min = 2, 0f83d639 Manuel Lauss 2009-10-31 286 .channels_max = 8, /* 2 without external help */ 0f83d639 Manuel Lauss 2009-10-31 287 }, 0f83d639 Manuel Lauss 2009-10-31 288 .ops = &au1xpsc_i2s_dai_ops, 0f83d639 Manuel Lauss 2009-10-31 289 }; 0f83d639 Manuel Lauss 2009-10-31 290 4edf87f5 Kuninori Morimoto 2013-03-21 291 static const struct snd_soc_component_driver au1xpsc_i2s_component = { 4edf87f5 Kuninori Morimoto 2013-03-21 292 .name = "au1xpsc-i2s", 4edf87f5 Kuninori Morimoto 2013-03-21 293 }; 4edf87f5 Kuninori Morimoto 2013-03-21 294 5c658be0 Bill Pemberton 2012-12-07 295 static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) 0f83d639 Manuel Lauss 2009-10-31 296 { 226d0f22 Julia Lawall 2011-10-18 297 struct resource *iores, *dmares; 4a161d23 Manuel Lauss 2008-07-09 298 unsigned long sel; 4a161d23 Manuel Lauss 2008-07-09 @299 int ret; 0f83d639 Manuel Lauss 2009-10-31 300 struct au1xpsc_audio_data *wd; 4a161d23 Manuel Lauss 2008-07-09 301 aa4079c1 Julia Lawall 2011-12-29 302 wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), aa4079c1 Julia Lawall 2011-12-29 303 GFP_KERNEL); 0f83d639 Manuel Lauss 2009-10-31 304 if (!wd) 4a161d23 Manuel Lauss 2008-07-09 305 return -ENOMEM; 4a161d23 Manuel Lauss 2008-07-09 306 226d0f22 Julia Lawall 2011-10-18 307 iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); :::::: The code at line 299 was first introduced by commit :::::: 4a161d235b68eb7234f40106560c488a1bdb3851 ALSA: ASoC: Au12x0/Au1550 PSC Audio support :::::: TO: Manuel Lauss <mano(a)roarinelk.homelinux.net> :::::: CC: Jaroslav Kysela <perex(a)perex.cz> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: spear_pcm: Use devm_snd_dmaengine_pcm_register to fix resource leak
by Axel Lin 30 Aug '15

30 Aug '15

All the callers assume devm_spear_pcm_platform_register is a devm_ API, so use devm_snd_dmaengine_pcm_register in devm_spear_pcm_platform_register. Fixes: e1771bcf99b0 ("ASoC: SPEAr: remove custom DMA alloc compat function") Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/spear/spear_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index a7dc3c5..e8476da 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -44,7 +44,7 @@ int devm_spear_pcm_platform_register(struct device *dev, *config = spear_dmaengine_pcm_config; config->compat_filter_fn = filter; - return snd_dmaengine_pcm_register(dev, config, + return devm_snd_dmaengine_pcm_register(dev, config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: sh: dma-sh7760: Convert to devm_snd_soc_register_platform
by Axel Lin 30 Aug '15

30 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/sh/dma-sh7760.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index fd11404..8fad444 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -327,13 +327,7 @@ static struct snd_soc_platform_driver sh7760_soc_platform = { static int sh7760_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &sh7760_soc_platform); -} - -static int sh7760_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; + return devm_snd_soc_register_platform(&pdev->dev, &sh7760_soc_platform); } static struct platform_driver sh7760_pcm_driver = { @@ -342,7 +336,6 @@ static struct platform_driver sh7760_pcm_driver = { }, .probe = sh7760_soc_platform_probe, - .remove = sh7760_soc_platform_remove, }; module_platform_driver(sh7760_pcm_driver); -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource
by Axel Lin 30 Aug '15

30 Aug '15

Use devm_ioremap_resource() instead of open code. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/au1x/psc-i2s.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index e742ef6..38e853a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -305,19 +305,9 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) return -ENOMEM; iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) - return -ENODEV; - - ret = -EBUSY; - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: sti_uniperif: Ensure component is unregistered when unload module
by Axel Lin 30 Aug '15

30 Aug '15

Use devm_snd_soc_register_component to ensure component is unregistered when unload the module. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/sti/sti_uniperif.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index dffabf3..39bcefe 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -224,9 +224,9 @@ static int sti_uniperiph_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, priv); - ret = snd_soc_register_component(&pdev->dev, - &sti_uniperiph_dai_component, - priv->dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &sti_uniperiph_dai_component, + priv->dai, 1); if (ret < 0) return ret; -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: zx296702-i2s: Fix resource leak when unload module
by Axel Lin 30 Aug '15

30 Aug '15

Use devm_* API to fix leaks in current code. 1. Use devm_kzalloc to fix memory leak for zx_i2s when unload the module. 2. Use devm_snd_soc_register_component to ensure component is unregistered when unload the module. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/zte/zx296702-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c index 1930c42..1cad93d 100644 --- a/sound/soc/zte/zx296702-i2s.c +++ b/sound/soc/zte/zx296702-i2s.c @@ -380,7 +380,7 @@ static int zx_i2s_probe(struct platform_device *pdev) struct zx_i2s_info *zx_i2s; int ret; - zx_i2s = kzalloc(sizeof(*zx_i2s), GFP_KERNEL); + zx_i2s = devm_kzalloc(&pdev->dev, sizeof(*zx_i2s), GFP_KERNEL); if (!zx_i2s) return -ENOMEM; @@ -401,8 +401,8 @@ static int zx_i2s_probe(struct platform_device *pdev) writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL); platform_set_drvdata(pdev, zx_i2s); - ret = snd_soc_register_component(&pdev->dev, &zx_i2s_component, - &zx_i2s_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &zx_i2s_component, + &zx_i2s_dai, 1); if (ret) { dev_err(&pdev->dev, "Register DAI failed: %d\n", ret); return ret; -- 2.1.0

3 2

[alsa-devel] [PATCH 1/3] ASoC: pxa-ssp: Convert to devm_snd_soc_register_component
by Axel Lin 30 Aug '15

30 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/pxa/pxa-ssp.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fbe2e93..3da485e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -813,14 +813,8 @@ static const struct of_device_id pxa_ssp_of_ids[] = { static int asoc_ssp_probe(struct platform_device *pdev) { - return snd_soc_register_component(&pdev->dev, &pxa_ssp_component, - &pxa_ssp_dai, 1); -} - -static int asoc_ssp_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - return 0; + return devm_snd_soc_register_component(&pdev->dev, &pxa_ssp_component, + &pxa_ssp_dai, 1); } static struct platform_driver asoc_ssp_driver = { @@ -830,7 +824,6 @@ static struct platform_driver asoc_ssp_driver = { }, .probe = asoc_ssp_probe, - .remove = asoc_ssp_remove, }; module_platform_driver(asoc_ssp_driver); -- 2.1.0

4 7

[alsa-devel] [PATCH 1/5] ASoC: au1x: Convert to devm_snd_soc_register_platform
by Axel Lin 30 Aug '15

30 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/au1x/dbdma2.c | 11 ++--------- sound/soc/au1x/dma.c | 11 ++--------- 2 files changed, 4 insertions(+), 18 deletions(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index dd94fea..5741c0a 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -344,14 +344,8 @@ static int au1xpsc_pcm_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, dmadata); - return snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); -} - -static int au1xpsc_pcm_drvremove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - - return 0; + return devm_snd_soc_register_platform(&pdev->dev, + &au1xpsc_soc_platform); } static struct platform_driver au1xpsc_pcm_driver = { @@ -359,7 +353,6 @@ static struct platform_driver au1xpsc_pcm_driver = { .name = "au1xpsc-pcm", }, .probe = au1xpsc_pcm_drvprobe, - .remove = au1xpsc_pcm_drvremove, }; module_platform_driver(au1xpsc_pcm_driver); diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 24cc7f4..fcf5a9ad 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -312,14 +312,8 @@ static int alchemy_pcm_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); - return snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform); -} - -static int alchemy_pcm_drvremove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - - return 0; + return devm_snd_soc_register_platform(&pdev->dev, + &alchemy_pcm_soc_platform); } static struct platform_driver alchemy_pcmdma_driver = { @@ -327,7 +321,6 @@ static struct platform_driver alchemy_pcmdma_driver = { .name = "alchemy-pcm-dma", }, .probe = alchemy_pcm_drvprobe, - .remove = alchemy_pcm_drvremove, }; module_platform_driver(alchemy_pcmdma_driver); -- 2.1.0

3 10

Re: [alsa-devel] [pulseaudio-discuss] sound is distorted when pulseaudio system timer scheduling is used (tsched=1)
by Raymond Yau 29 Aug '15

29 Aug '15

> > I am having issue with pulseaudiosink, > I used following command to play a wav file > gst-launch filesrc location=<wav file path> ! wavparse ! pulseaudiosink > > if tsched is enabled, then play of wav file via pulseaudiosink gives me lots of Underrun, and the sound is distorted, following is the pulseaudio log when --log-level=3 > > I: [pulseaudio] protocol-native.c: Got credentials: uid=0 gid=0 success=1 > I: [pulseaudio] sink-input.c: Trying to change sample rate > I: [pulseaudio] alsa-sink.c: Updating rate for device hw:0, new rate is 48000 > I: [pulseaudio] source.c: Changed sampling rate successfully > I: [pulseaudio] sink.c: Changed sampling rate successfully > I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Trying resume... > I: [alsa-sink-HiFi wm8962-0] alsa-util.c: cannot disable ALSA period wakeups > I: [alsa-sink-HiFi wm8962-0] alsa-util.c: ALSA period wakeups were not disabled > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Time scheduling watermark is 18.38ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Resumed successfully... > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Starting playback. > I: [pulseaudio] sink-input.c: Created input 0 "pulsesink probe" on alsa_output.platform-sound.analog-s > tereo with sample spec s16le 2ch 48000Hz and channel map front-left,front-right > I: [pulseaudio] sink-input.c: media.name = "pulsesink probe" > I: [pulseaudio] sink-input.c: application.name = "gst-launch-0.10" > I: [pulseaudio] sink-input.c: native-protocol.peer = "UNIX socket client" > I: [pulseaudio] sink-input.c: native-protocol.version = "29" > I: [pulseaudio] sink-input.c: application.process.id = "985" > I: [pulseaudio] sink-input.c: application.process.user = "root" > I: [pulseaudio] sink-input.c: application.process.host = "mx6q" > I: [pulseaudio] sink-input.c: application.process.binary = "gst-launch-0.10" > I: [pulseaudio] sink-input.c: application.language = "C" > I: [pulseaudio] sink-input.c: application.process.machine_id = "0aa7a9566c734f608cbd36f187de8645" > I: [pulseaudio] sink-input.c: module-stream-restore.id = "sink-input-by-application-name:gst-launc > h-0.10" > I: [pulseaudio] protocol-native.c: Requested tlength=250.00 ms, minreq=20.00 ms > I: [pulseaudio] protocol-native.c: Final latency 250.00 ms = 105.00 ms + 2*20.00 ms + 105.00 ms > I: [pulseaudio] sink-input.c: Freeing input 0 "pulsesink probe" > I: [pulseaudio] sink-input.c: Created input 1 "Playback Stream" on alsa_output.platform-sound.analog-s > tereo with sample spec s16le 2ch 48000Hz and channel map front-left,front-right > I: [pulseaudio] sink-input.c: media.name = "Playback Stream" > I: [pulseaudio] sink-input.c: application.name = "gst-launch-0.10" > I: [pulseaudio] sink-input.c: native-protocol.peer = "UNIX socket client" > I: [pulseaudio] sink-input.c: native-protocol.version = "29" > I: [pulseaudio] sink-input.c: application.process.id = "985" > I: [pulseaudio] sink-input.c: application.process.user = "root" > I: [pulseaudio] sink-input.c: application.process.host = "mx6q" > I: [pulseaudio] sink-input.c: application.process.binary = "gst-launch-0.10" > I: [pulseaudio] sink-input.c: application.language = "C" > I: [pulseaudio] sink-input.c: application.process.machine_id = "0aa7a9566c734f608cbd36f187de8645" > I: [pulseaudio] sink-input.c: module-stream-restore.id = "sink-input-by-application-name:gst-launc > h-0.10" > I: [pulseaudio] protocol-native.c: Requested tlength=200.00 ms, minreq=10.00 ms > I: [pulseaudio] protocol-native.c: Final latency 200.00 ms = 90.00 ms + 2*10.00 ms + 90.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 28.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 38.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 48.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 58.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 68.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 78.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 80.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 1.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 2.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 4.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 8.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 16.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 26.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 36.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 46.00 ms > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: ALSA woke us up to write new data to the device, but there w > as actually nothing to write! > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Most likely this is a bug in the ALSA driver '(null)'. Pleas > e report this issue to the ALSA developers. > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent sn > d_pcm_avail() returned 0 or another value < min_avail. > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 56.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 66.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 76.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 86.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 96.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 86.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 106.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 96.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 116.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 106.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 126.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 116.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 136.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 126.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Underrun! > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 146.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Underrun! > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 136.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Underrun! > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 156.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 146.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 166.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 156.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 176.00 ms > I: [pulseaudio] sink-input.c: Freeing input 1 "Playback Stream" > I: [pulseaudio] client.c: Freed 4 "gst-launch-0.10" > I: [pulseaudio] protocol-native.c: Connection died. > > from the log it seems like, when system timer based scheduling is used, then pulseaudio will try to keep minimum watermark/latency, only when Underrun occurs, it will increase watermark/latency. > > Is this the expected behaviour of pulseaudio 'glict free' feature? > And from the log there is some error message complain about ALSA driver > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: ALSA woke us up to write new data to the device, but there w > as actually nothing to write! > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Most likely this is a bug in the ALSA driver '(null)'. Pleas > e report this issue to the ALSA developers. > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent sn > d_pcm_avail() returned 0 or another value < min_avail. > > does this mean, there is some bug in ALSA driver, or it just indicates the ALSA driver doesn't support glitch free feature very well The timer base scheduling require sound card driver provide accurate hw_ptr position at any time. So far only two sound card drivers snd-hda-intel and snd-oxygen can provide accurate hw_ptr position from hardware register static snd_pcm_uframes_t dmaengine_pcm_pointer( struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_dmaengine_pcm_pointer_no_residue(substream); else return snd_dmaengine_pcm_pointer(substream); } http://lxr.free-electrons.com/ident?i=dmaengine_tx_status Seem none of soc sound drivers can report position accurate than period size Is there any reason to enable timer based scheduling by default ? http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081501.… You can try alexandra 's program to find out how accuate hw_ptr can your sound card driver can report ?

1 0

[alsa-devel] [PATCH] ALSA: fireworks/bebob/dice/oxfw: fix substreams counting at vmalloc failure
by Takashi Sakamoto 29 Aug '15

29 Aug '15

In PCM core, when hw_params() in each driver returns error, the state of PCM substream is kept as 'open'. In this case, current drivers for sound units on IEEE 1394 bus doesn't decrement substream counter in hw_free() correctly. This causes these drivers to keep streams even if not required. This commit fixes this bug. When snd_pcm_lib_alloc_vmalloc_buffer() fails, hw_params function in each driver returns without incrementing the counter. Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_pcm.c | 20 ++++++++++++++++---- sound/firewire/dice/dice-pcm.c | 18 ++++++++++++++---- sound/firewire/fireworks/fireworks_pcm.c | 18 ++++++++++++++---- sound/firewire/oxfw/oxfw-pcm.c | 17 +++++++++++++---- 4 files changed, 57 insertions(+), 16 deletions(-) diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 7a2c1f5..c0f018a 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -211,26 +211,38 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_bebob *bebob = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&bebob->substreams_counter); amdtp_stream_set_pcm_format(&bebob->tx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + + return 0; } static int pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_bebob *bebob = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&bebob->substreams_counter); amdtp_stream_set_pcm_format(&bebob->rx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + + return 0; } static int diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index f7771451..4e67b1d 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -230,6 +230,12 @@ static int capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { mutex_lock(&dice->mutex); @@ -240,13 +246,18 @@ static int capture_hw_params(struct snd_pcm_substream *substream, amdtp_stream_set_pcm_format(&dice->tx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { mutex_lock(&dice->mutex); @@ -257,8 +268,7 @@ static int playback_hw_params(struct snd_pcm_substream *substream, amdtp_stream_set_pcm_format(&dice->rx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int capture_hw_free(struct snd_pcm_substream *substream) diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 8a34753..c30b2ff 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -244,25 +244,35 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_efw *efw = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&efw->capture_substreams); amdtp_stream_set_pcm_format(&efw->tx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_efw *efw = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&efw->playback_substreams); amdtp_stream_set_pcm_format(&efw->rx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int pcm_capture_hw_free(struct snd_pcm_substream *substream) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 67ade07..9c73930 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -231,7 +231,12 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_oxfw *oxfw = substream->private_data; + int err; + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { mutex_lock(&oxfw->mutex); @@ -241,13 +246,18 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, amdtp_stream_set_pcm_format(&oxfw->tx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_oxfw *oxfw = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { mutex_lock(&oxfw->mutex); @@ -257,8 +267,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int pcm_capture_hw_free(struct snd_pcm_substream *substream) -- 2.1.4

3 2

[alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants
by Takashi Sakamoto 29 Aug '15

29 Aug '15

This patchset for Linux 4.3 updates my previous one: [alsa-devel] [PATCH 00/25] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096338.html As the same as the previous, the second patch is too large to be blasted by mailing list server. I request server administrators to grant it. Changes: * Enable to select as kernel module or the others. * Add a comment for memory barriers in firewire-lib. * Add a comment about using jiffies in firewire-lib. * Use jiffies_64 helpers instead of jiffies helpers in firewire-lib. Rest of work: * Using proper clocksource for MIDI throttle in asynchronous transaction helper. * Handling control/status messages for TASCAM driver. * Supporting MIDI exclusive message in outgoing packets for TASCAM driver. * Researching a mechanism to de-synchronize sometimes for Digi00x family. You can see this patchset as a part in this branch. https://github.com/takaswie/sound/tree/amdtp-variants Takashi Sakamoto (25): ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' for functional separation ALSA: firewire-lib: functional separation for packet transmission layer and data processing layer ALSA: firewire-lib: add helper functions for asynchronous MIDI port ALSA: firewire-lib: add a restriction for a transaction at once ALSA: firewire-lib: schedule tasklet again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error ALSA: firewire-digi00x: add skeleton for Digi 002/003 family ALSA: firewire-digi00x: add data processing layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add MIDI functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging ALSA: firewire-digi00x: add support for MIDI ports for machine control ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add data processing layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add transaction functionality ALSA: firewire-tascam: add MIDI functionality ALSA: firewire-tascam: add hwdep interface include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 9 + sound/firewire/Kconfig | 27 ++ sound/firewire/Makefile | 4 +- sound/firewire/amdtp-am824.c | 451 ++++++++++++++++++++++++ sound/firewire/amdtp-am824.h | 47 +++ sound/firewire/{amdtp.c => amdtp-stream.c} | 375 ++++---------------- sound/firewire/{amdtp.h => amdtp-stream.h} | 113 ++---- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 16 +- sound/firewire/dice/dice-pcm.c | 12 +- sound/firewire/dice/dice-stream.c | 41 ++- sound/firewire/dice/dice.h | 2 +- sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 436 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++ sound/firewire/digi00x/digi00x-midi.c | 198 +++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 353 +++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 ++++++ sound/firewire/digi00x/digi00x-stream.c | 422 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 137 ++++++++ sound/firewire/digi00x/digi00x.c | 174 ++++++++++ sound/firewire/digi00x/digi00x.h | 157 +++++++++ sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 16 +- sound/firewire/fireworks/fireworks_pcm.c | 10 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/lib.c | 147 ++++++++ sound/firewire/lib.h | 52 +++ sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 10 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- sound/firewire/tascam/Makefile | 4 + sound/firewire/tascam/amdtp-tascam.c | 248 +++++++++++++ sound/firewire/tascam/tascam-hwdep.c | 202 +++++++++++ sound/firewire/tascam/tascam-midi.c | 162 +++++++++ sound/firewire/tascam/tascam-pcm.c | 298 ++++++++++++++++ sound/firewire/tascam/tascam-proc.c | 88 +++++ sound/firewire/tascam/tascam-stream.c | 497 +++++++++++++++++++++++++++ sound/firewire/tascam/tascam-transaction.c | 291 ++++++++++++++++ sound/firewire/tascam/tascam.c | 230 +++++++++++++ sound/firewire/tascam/tascam.h | 140 ++++++++ 48 files changed, 5295 insertions(+), 498 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h rename sound/firewire/{amdtp.c => amdtp-stream.c} (70%) rename sound/firewire/{amdtp.h => amdtp-stream.h} (77%) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam-transaction.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

2 35

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 29 Aug '15

29 Aug '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 14 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 860 +++++++++++++++++++++ sound/soc/codecs/nau8825.h | 258 +++++++ 5 files changed, 1138 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..0334be3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,14 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Example: + + codec: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7fcbc22..1a3639c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,6 +78,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -892,6 +893,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b46c8f..4874881 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -262,6 +263,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..1c99308b --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,860 @@ +/* + * NAU8825 audio codec driver + * + * Copyright 2015 Nuvoton Technology Corp. + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/input.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "nau8825.h" + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + int irq; + struct snd_soc_jack *jack; + int button_pressed; +}; + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); + +static const struct snd_kcontrol_new nau8825_snd_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Sidetone Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Gain", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), +}; + +static int nau8825_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, NAU8825_JAMNODCLOW); + } else { + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, 0); + } + + return 0; +} + +static int nau8825_output_driver_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* Power Up L and R Output Drivers */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3c, + 0x3c); + /* Power up Main Output Drivers (The main driver must be turned + on after the predriver to avoid pops) */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3, + 0x3); + } else { + /* Power Down L and R Output Drivers */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3f, + 0x0); + } + + return 0; +} + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, + &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, + &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Change Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("DACL Power", NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, + 0), + SND_SOC_DAPM_SUPPLY("DACR Power", NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, + 0), + + SND_SOC_DAPM_OUT_DRV_E("Output Driver", SND_SOC_NOPM, 0, 0, NULL, 0, + nau8825_output_driver_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"Capture", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DDACL", NULL, "DACL Power"}, + {"DDACR", NULL, "DACR Power"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "Change Pump"}, + {"HP amp R", NULL, "Change Pump"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver", NULL, "ADACL"}, + {"Output Driver", NULL, "ADACR"}, + {"HPOL", NULL, "Output Driver"}, + {"HPOR", NULL, "Output Driver"}, +}; + +static int nau8825_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 nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int reg_val; + + reg_val = mute ? NAU8825_HP_MUTE : 0; + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HP_MUTE, reg_val); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .digital_mute = nau8825_mute, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + nau8825->jack = jack; + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons; + + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_info(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_info(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + lower 8 bits - for long pressed buttons */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct regmap *regmap) +{ + /* Setup Button Detect (Debounce, number of buttons, and Hysteresis) */ + regmap_write(regmap, NAU8825_REG_KEYDET_CTRL, 0x7311); + + /* Setup 4 buttons impedane according to Android specification + * https://source.android.com/accessories/headset-spec.html + * Button 0 - 0-70 Ohm + * Button 1 - 110-180 Ohm + * Button 2 - 210-290 Ohm + * Button 3 - 360-680 Ohm + */ + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, 0x0f1f); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, 0x325f); +} + +static void nau8825_init_regs(struct regmap *regmap) +{ + /* Power down DACs for power savings */ + regmap_write(regmap, NAU8825_REG_CHARGE_PUMP, 0x0300); + /* DAC Zero Crossing Enable */ + regmap_write(regmap, NAU8825_REG_MUTE_CTRL, 0x1000); + /* VMID Enable and Tieoff */ + regmap_write(regmap, NAU8825_REG_BIAS_ADJ, 0x0060); + /* Analog Bias Enable, Disable Boost Driver, + Automatic Short circuit protection enable */ + regmap_write(regmap, NAU8825_REG_BOOST, 0x3140); + /* Ground HP Outputs[1:0], + Enable Automatic Mic/Gnd switching reading on insert interrupt[6] */ + regmap_write(regmap, NAU8825_REG_HSD_CTRL, 0x004f); + + regmap_write(regmap, NAU8825_REG_JACK_DET_CTRL, 0x01e0); + /* IRQ Output Enable, Mask Insert Interrupt */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_MASK, 0x0801); + /* Jack Detect pull up (High=eject, Low=insert) */ + regmap_write(regmap, NAU8825_REG_GPIO12_CTRL, 0x0800); + /* Setup SAR ADC */ + regmap_write(regmap, NAU8825_REG_SAR_CTRL, 0x0280); + nau8825_setup_buttons(regmap); + + /* Setup ADC x128 OSR */ + regmap_write(regmap, NAU8825_REG_ADC_RATE, 0x0002); + /* Setup DAC x128 OSR */ + regmap_write(regmap, NAU8825_REG_DAC_CTRL1, 0x0082); + /* DAC bias settings */ + regmap_write(regmap, NAU8825_REG_ANALOG_CONTROL_2, 0x1003); + /* ADC Bias Settings */ + regmap_write(regmap, NAU8825_REG_ANALOG_ADC_2, 0x0260); + /* Maximize Jack Insert Debounce */ + regmap_write(regmap, NAU8825_REG_JACK_DET_CTRL, 0x00e0); + + /* Setup clocks */ + regmap_write(regmap, NAU8825_REG_ENA_CTRL, 0x0416); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + as we cannot guarantee one will be provided by the system. Turning + master mode on then off enables us to generate that FSCLK cycle + with a minimum of contention on the clock bus. + Enables master mode and correctly divides BCLK and FSCLK from MCLK. + */ + regmap_write(regmap, NAU8825_REG_I2S_PCM_CTRL2, 0x2019); + regmap_write(regmap, NAU8825_REG_I2S_PCM_CTRL2, 0x0010); + + /* Mask jack insert interruption */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0x0011); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + + nau8825->dapm = &codec->dapm; + + nau8825_init_regs(nau8825->regmap); + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "DDACR"); + snd_soc_dapm_sync(&codec->dapm); + + if (nau8825->irq) { + int ret = devm_request_threaded_irq(dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + } + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + + .controls = nau8825_snd_controls, + .num_controls = ARRAY_SIZE(nau8825_snd_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825; + int ret, value; + + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->irq = i2c->irq; + nau8825->dev = dev; + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..e1eb769 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,258 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * 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 __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +#define NAU8825_SPKR_DWN1R_SHORT (1 << 1) +#define NAU8825_SPKR_DWN1L_SHORT (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SINC4 (1 << 4) +#define NAU8825_ADC_SINC_DOWN_MASK 3 +#define NAU8825_ADC_SINC_DOWN_32 0 +#define NAU8825_ADC_SINC_DOWN_64 1 +#define NAU8825_ADC_SINC_DOWN_128 2 +#define NAU8825_ADC_SINC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_MASK 7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_MASK (3 << 4) +#define NAU8825_BIAS_VMID_SEL_OPEN (0 << 4) +#define NAU8825_BIAS_VMID_SEL_25KOHM (1 << 4) +#define NAU8825_BIAS_VMID_SEL_125KOHM (2 << 4) +#define NAU8825_BIAS_VMID_SEL_2_5KOHM (3 << 4) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 12) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_INTEGR_R (1 << 5) + +/* CHARGE_PUMP_AND_POWER_DOWN_CONTROL (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.5.0.rc2.392.g76e840b

2 12

[alsa-devel] [PATCH] ASoC: qcom: Constify asoc_qcom_lpass_cpu_dai_ops
by Axel Lin 28 Aug '15

28 Aug '15

asoc_qcom_lpass_cpu_dai_ops is exported and used by multiple drivers, make it const to prevent modifying it at run time. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/lpass.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 23f3d59..97bc202 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -235,7 +235,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } -struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { +const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index d6e86c1..0b63e2e 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -93,6 +93,6 @@ int asoc_qcom_lpass_platform_register(struct platform_device *); int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); -extern struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; #endif /* __LPASS_H__ */ -- 2.1.0

2 1

[alsa-devel] Applied "sound: Clean up docbook warnings" to the asoc tree
by Mark Brown 28 Aug '15

28 Aug '15

The patch sound: Clean up docbook warnings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 55b1faa33d485ace99a5a9e58bb1007782ae8ec9 Mon Sep 17 00:00:00 2001 From: Jonathan Corbet <corbet(a)lwn.net> Date: Tue, 25 Aug 2015 01:14:48 -0600 Subject: [PATCH] sound: Clean up docbook warnings A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet(a)lwn.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/DocBook/alsa-driver-api.tmpl | 2 +- include/sound/soc.h | 11 +++++++---- sound/soc/soc-core.c | 13 +++++++++---- sound/soc/soc-dapm.c | 2 +- 4 files changed, 18 insertions(+), 10 deletions(-) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 71f9246..e94a10b 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -108,7 +108,7 @@ <sect1><title>ASoC Core API</title> !Iinclude/sound/soc.h !Esound/soc/soc-core.c -!Esound/soc/soc-cache.c + !Esound/soc/soc-devres.c !Esound/soc/soc-io.c !Esound/soc/soc-pcm.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf..4537e81 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -619,6 +619,7 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported + * @list: internal list entry */ struct snd_soc_jack_pin { struct list_head list; @@ -635,7 +636,7 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container + * @list: internal list entry */ struct snd_soc_jack_zone { unsigned int min_mv; @@ -651,12 +652,12 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debounce time in ms * @wake: enable as wake source * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an @@ -672,11 +673,13 @@ struct snd_soc_jack_gpio { int debounce_time; bool wake; + /* private: */ struct snd_soc_jack *jack; struct delayed_work work; struct gpio_desc *desc; void *data; + /* public: */ int (*jack_status_check)(void *data); }; @@ -1319,7 +1322,7 @@ static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm( /** * snd_soc_dapm_init_bias_level() - Initialize CODEC DAPM bias level - * @dapm: The CODEC for which to initialize the DAPM bias level + * @codec: The CODEC for which to initialize the DAPM bias level * @level: The DAPM level to initialize to * * Initializes the CODEC DAPM bias level. See snd_soc_dapm_init_bias_level(). diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 90d6335..3224251 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2798,6 +2798,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core * + * @dev: The device to unregister */ void snd_soc_unregister_component(struct device *dev) { @@ -2877,7 +2878,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: The device for the platform + * @platform_drv: The driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -2938,7 +2940,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -3029,7 +3031,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: The parent device for this codec + * @codec_drv: Codec driver + * @dai_drv: The associated DAI driver + * @num_dai: Number of DAIs */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -3128,7 +3133,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..c4e3720 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2911,7 +2911,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card to be checked for new dapm widgets * * Checks the codec for any new dapm widgets and creates them if found. * -- 2.5.0

1 0

[alsa-devel] [PATCH] sound: Clean up docbook warnings
by Jonathan Corbet 28 Aug '15

28 Aug '15

A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet(a)lwn.net> --- NOTE that it would be good if somebody more knowledgeable than me checked my work, since I had to fill in a couple of things. Documentation/DocBook/alsa-driver-api.tmpl | 2 +- include/sound/soc.h | 11 +++++++---- sound/soc/soc-core.c | 17 +++++++++++------ sound/soc/soc-dapm.c | 2 +- 4 files changed, 20 insertions(+), 12 deletions(-) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 71f9246127ec..e94a10bb4a9e 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -108,7 +108,7 @@ <sect1><title>ASoC Core API</title> !Iinclude/sound/soc.h !Esound/soc/soc-core.c -!Esound/soc/soc-cache.c + !Esound/soc/soc-devres.c !Esound/soc/soc-io.c !Esound/soc/soc-pcm.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf9d54a..4537e81eeeda 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -619,6 +619,7 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported + * @list: internal list entry */ struct snd_soc_jack_pin { struct list_head list; @@ -635,7 +636,7 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container + * @list: internal list entry */ struct snd_soc_jack_zone { unsigned int min_mv; @@ -651,12 +652,12 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debounce time in ms * @wake: enable as wake source * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an @@ -672,11 +673,13 @@ struct snd_soc_jack_gpio { int debounce_time; bool wake; + /* private: */ struct snd_soc_jack *jack; struct delayed_work work; struct gpio_desc *desc; void *data; + /* public: */ int (*jack_status_check)(void *data); }; @@ -1319,7 +1322,7 @@ static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm( /** * snd_soc_dapm_init_bias_level() - Initialize CODEC DAPM bias level - * @dapm: The CODEC for which to initialize the DAPM bias level + * @codec: The CODEC for which to initialize the DAPM bias level * @level: The DAPM level to initialize to * * Initializes the CODEC DAPM bias level. See snd_soc_dapm_init_bias_level(). diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4a5c0e3f97..32242512d828 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2127,7 +2127,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -2798,6 +2798,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core * + * @dev: The device to unregister */ void snd_soc_unregister_component(struct device *dev) { @@ -2838,7 +2839,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -2877,7 +2878,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: The device for the platform + * @platform_drv: The driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -2938,7 +2940,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -3029,7 +3031,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: The parent device for this codec + * @codec_drv: Codec driver + * @dai_drv: The associated DAI driver + * @num_dai: Number of DAIs */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -3128,7 +3133,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c92480c..c4e3720bea41 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2911,7 +2911,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card to be checked for new dapm widgets * * Checks the codec for any new dapm widgets and creates them if found. * -- 2.4.3

3 8

[alsa-devel] Applied "ASoC: ics43432: fix platform_no_drv_owner.cocci warnings" to the asoc tree
by Mark Brown 28 Aug '15

28 Aug '15

The patch ASoC: ics43432: fix platform_no_drv_owner.cocci warnings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 11d932416ecada27ca7576dca7e74e23477df7ef Mon Sep 17 00:00:00 2001 From: kbuild test robot <fengguang.wu(a)intel.com> Date: Fri, 28 Aug 2015 10:04:03 +0800 Subject: [PATCH] ASoC: ics43432: fix platform_no_drv_owner.cocci warnings sound/soc/codecs/ics43432.c:66:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Ricard Wanderlof <ricard.wanderlof(a)axis.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ics43432.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index 4f202c1..d217fed 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -63,7 +63,6 @@ MODULE_DEVICE_TABLE(of, ics43432_dt_ids); static struct platform_driver ics43432_driver = { .driver = { .name = "ics43432", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(ics43432_ids), }, .probe = ics43432_probe, -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: ics43432: Fix module license
by Axel Lin 28 Aug '15

28 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ics43432.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index 4f202c1..382a004 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -6,7 +6,7 @@ * * Copyright (c) 2015 Axis Communications AB * - * Licensed under GPL2. + * Licensed under GPL v2. */ #include <linux/module.h> @@ -74,4 +74,4 @@ module_platform_driver(ics43432_driver); MODULE_DESCRIPTION("ASoC ICS43432 driver"); MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- 2.1.0

2 1

[alsa-devel] Applied "ASoC: ics43432: Fix build errors" to the asoc tree
by Mark Brown 28 Aug '15

28 Aug '15

The patch ASoC: ics43432: Fix build errors has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b37bfdaac914f4886b8be803703b12177eb873f6 Mon Sep 17 00:00:00 2001 From: Ricard Wanderlof <ricard.wanderlof(a)axis.com> Date: Fri, 28 Aug 2015 12:11:48 +0200 Subject: [PATCH] ASoC: ics43432: Fix build errors Fix build errors Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ics43432.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index 7c35c98..dd850b9 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -57,7 +57,7 @@ static const struct of_device_id ics43432_ids[] = { { .compatible = "invensense,ics43432", }, { } }; -MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +MODULE_DEVICE_TABLE(of, ics43432_ids); #endif static struct platform_driver ics43432_driver = { -- 2.5.0

1 0

[alsa-devel] [PATCH v3] OMAPDSS: hdmi: Reconfigure and restart audio when display is enabled
by Jyri Sarha 28 Aug '15

28 Aug '15

Reconfigure and restart audio when display is enabled, if audio playback was active before. This is needed in a situation when an audio+video stream application opens the audio stream before the video. When video stream is opened the display mode may change and that would aborted audio playback, because the display is momentarily turned off. The audio configuration is stored when it is successfully applied and a boolean is set when the audio playback is started and unset when stopped. This data is used to reconfigure the audio when display is re-enabled. The audio playback is aborted if the reconfiguration fails. A new spin lock is introduced in order to protect state variables related to audio playback status. This is needed for the transition from display enabled state (when audio start/stop commands can be written to HW) to display disabled state (when audio start/stop commands update only the hdmi.audio_playing variable) to always serialize correctly with the start/stop audio commands. The already existing mutex can not be used, because the audio start and stop commands are executed in atomic context. For example: when display is turned back on we take the spinlock and we can be sure that the audio start/stop status will not change while we update the HW according to hdmi.audio_playing state and set hdmi.display_enabled to true. After releasing the lock hdmi.display_enabled is true and all audio_start and audio_stop commands write their stuff directly to HW. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- drivers/video/fbdev/omap2/dss/hdmi.h | 9 ++++- drivers/video/fbdev/omap2/dss/hdmi4.c | 66 +++++++++++++++++++++++++----- drivers/video/fbdev/omap2/dss/hdmi5.c | 76 ++++++++++++++++++++++++++++------- 3 files changed, 124 insertions(+), 27 deletions(-) diff --git a/drivers/video/fbdev/omap2/dss/hdmi.h b/drivers/video/fbdev/omap2/dss/hdmi.h index e4a32fe..53616b0 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi.h +++ b/drivers/video/fbdev/omap2/dss/hdmi.h @@ -351,13 +351,20 @@ struct omap_hdmi { struct regulator *vdda_reg; bool core_enabled; - bool display_enabled; struct omap_dss_device output; struct platform_device *audio_pdev; void (*audio_abort_cb)(struct device *dev); int wp_idlemode; + + bool audio_configured; + struct omap_dss_audio audio_config; + + /* This lock should be taken when booleans bellow are touched. */ + spinlock_t audio_playing_lock; + bool audio_playing; + bool display_enabled; }; #endif diff --git a/drivers/video/fbdev/omap2/dss/hdmi4.c b/drivers/video/fbdev/omap2/dss/hdmi4.c index 6d3aa3f..94c8d55 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi4.c +++ b/drivers/video/fbdev/omap2/dss/hdmi4.c @@ -321,9 +321,22 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_enable(&hd->wp, true); + hdmi4_audio_start(&hd->core, &hd->wp); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi4_audio_stop(&hd->core, &hd->wp); + hdmi_wp_audio_enable(&hd->wp, false); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -342,7 +355,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi4_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -354,17 +381,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -568,6 +597,8 @@ static int hdmi_audio_shutdown(struct device *dev) mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + hd->audio_playing = false; mutex_unlock(&hd->lock); return 0; @@ -576,25 +607,34 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi4_audio_start(&hd->core, &hd->wp); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_start_audio_stream(hd); + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi4_audio_stop(&hd->core, &hd->wp); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; + + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -612,7 +652,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi4_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); - + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -657,6 +700,7 @@ static int hdmi4_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); diff --git a/drivers/video/fbdev/omap2/dss/hdmi5.c b/drivers/video/fbdev/omap2/dss/hdmi5.c index 7f87578..b6c767b 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi5.c +++ b/drivers/video/fbdev/omap2/dss/hdmi5.c @@ -349,9 +349,24 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + hdmi_wp_audio_enable(&hd->wp, true); + hdmi_wp_audio_core_req_enable(&hd->wp, true); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_core_req_enable(&hd->wp, false); + hdmi_wp_audio_enable(&hd->wp, false); + REG_FLD_MOD(hd->wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -370,7 +385,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi5_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -382,17 +411,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -596,6 +627,8 @@ static int hdmi_audio_shutdown(struct device *dev) mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + hd->audio_playing = false; mutex_unlock(&hd->lock); return 0; @@ -604,32 +637,35 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - /* No-idle while playing audio, store the old value */ - hd->wp_idlemode = REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + spin_lock_irqsave(&hd->audio_playing_lock, flags); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi_wp_audio_core_req_enable(&hd->wp, true); + if (hd->display_enabled) { + hdmi_start_audio_stream(hd); + } + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_core_req_enable(&hd->wp, false); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; - /* Playback stopped, restore original idlemode */ - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -648,6 +684,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi5_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -678,6 +718,11 @@ static int hdmi_audio_register(struct device *dev) if (IS_ERR(hdmi.audio_pdev)) return PTR_ERR(hdmi.audio_pdev); + hdmi_runtime_get(); + hdmi.wp_idlemode = + REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); + hdmi_runtime_put(); + return 0; } @@ -692,6 +737,7 @@ static int hdmi5_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); -- 1.9.1

1 0

[alsa-devel] [PATCH v2] OMAPDSS: hdmi: Reconfigure and restart audio when display is enabled
by Jyri Sarha 28 Aug '15

28 Aug '15

Reconfigure and restart audio when display is enabled, if audio playback was active before. The audio configuration is stored when is is successfully applied and a boolean is set when playback has started and unset when stopped. This data is used to reconfigure the audio when display is re-enabled. Abort audio playback if reconfiguration fails. A new spin lock is introduced in order to protect state variables related to audio playback status. This is needed for the transitions from display enabled state (when audio start/stop commands can be written to HW) to display disabled state (when audio start/stop commands update only the hdmi.audio_playing variable) to always serialize correctly with the start/stop audio commands. For example: when display is turned back on we take the spinlock and we can be sure that the audio start/stop status won't change while we update the HW according to hdmi.audio_playing state and set hdmi.display_enabled to true. After releasing the lock hdmi.display_enabled is true and all audio_start and audio_stop commands write their stuff directly to HW. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- I dropped the ASoC maintainers from the recipient list as this patch hardly concerns them. drivers/video/fbdev/omap2/dss/hdmi.h | 9 +++- drivers/video/fbdev/omap2/dss/hdmi4.c | 69 +++++++++++++++++++++++++----- drivers/video/fbdev/omap2/dss/hdmi5.c | 79 ++++++++++++++++++++++++++++------- 3 files changed, 130 insertions(+), 27 deletions(-) diff --git a/drivers/video/fbdev/omap2/dss/hdmi.h b/drivers/video/fbdev/omap2/dss/hdmi.h index e4a32fe..e48aefd 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi.h +++ b/drivers/video/fbdev/omap2/dss/hdmi.h @@ -351,13 +351,20 @@ struct omap_hdmi { struct regulator *vdda_reg; bool core_enabled; - bool display_enabled; struct omap_dss_device output; struct platform_device *audio_pdev; void (*audio_abort_cb)(struct device *dev); int wp_idlemode; + + bool audio_configured; + struct omap_dss_audio audio_config; + + /* This lock should be taken when booleas bellow is touched. */ + spinlock_t audio_playing_lock; + bool audio_playing; + bool display_enabled; }; #endif diff --git a/drivers/video/fbdev/omap2/dss/hdmi4.c b/drivers/video/fbdev/omap2/dss/hdmi4.c index 6d3aa3f..6b9817a 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi4.c +++ b/drivers/video/fbdev/omap2/dss/hdmi4.c @@ -321,9 +321,22 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_enable(&hd->wp, true); + hdmi4_audio_start(&hd->core, &hd->wp); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi4_audio_stop(&hd->core, &hd->wp); + hdmi_wp_audio_enable(&hd->wp, false); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -342,7 +355,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi4_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -354,17 +381,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -565,9 +594,14 @@ out: static int hdmi_audio_shutdown(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + spin_lock_irqsave(&hd->audio_playing_lock, flags); + hd->audio_playing = false; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); mutex_unlock(&hd->lock); return 0; @@ -576,25 +610,34 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi4_audio_start(&hd->core, &hd->wp); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_start_audio_stream(hd); + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi4_audio_stop(&hd->core, &hd->wp); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; + + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -612,7 +655,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi4_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); - + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -657,6 +703,7 @@ static int hdmi4_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); diff --git a/drivers/video/fbdev/omap2/dss/hdmi5.c b/drivers/video/fbdev/omap2/dss/hdmi5.c index 7f87578..85e70af 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi5.c +++ b/drivers/video/fbdev/omap2/dss/hdmi5.c @@ -349,9 +349,24 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + hdmi_wp_audio_enable(&hd->wp, true); + hdmi_wp_audio_core_req_enable(&hd->wp, true); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_core_req_enable(&hd->wp, false); + hdmi_wp_audio_enable(&hd->wp, false); + REG_FLD_MOD(hd->wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -370,7 +385,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi5_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -382,17 +411,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -593,9 +624,14 @@ out: static int hdmi_audio_shutdown(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + spin_lock_irqsave(&hd->audio_playing_lock, flags); + hd->audio_playing = false; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); mutex_unlock(&hd->lock); return 0; @@ -604,32 +640,35 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - /* No-idle while playing audio, store the old value */ - hd->wp_idlemode = REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + spin_lock_irqsave(&hd->audio_playing_lock, flags); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi_wp_audio_core_req_enable(&hd->wp, true); + if (hd->display_enabled) { + hdmi_start_audio_stream(hd); + } + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_core_req_enable(&hd->wp, false); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; - /* Playback stopped, restore original idlemode */ - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -648,6 +687,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi5_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -678,6 +721,11 @@ static int hdmi_audio_register(struct device *dev) if (IS_ERR(hdmi.audio_pdev)) return PTR_ERR(hdmi.audio_pdev); + hdmi_runtime_get(); + hdmi.wp_idlemode = + REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); + hdmi_runtime_put(); + return 0; } @@ -692,6 +740,7 @@ static int hdmi5_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); -- 1.9.1

2 3

[alsa-devel] [PATCH 0/2] OMAPDSS: Couple of HDMI audio fixes
by Jyri Sarha 28 Aug '15

28 Aug '15

The first fix is a hairy one, but I think the locking is fool proof now. It is needed because there is no telling in which order user space starts an audio and video stream playback. If the audio is started first and the video mode is changed when video playback starts the audio setup needs to survive display turning off and back on again. After this patch the audio streams should survive a suspend-resume cycle too. The second one is a trivial work-around to a problem in ALSA constraint resolver code. The two fixes are totally independent and the video and audio side patches applied separately trough their own trees. Jyri Sarha (2): OMAPDSS: hdmi: Reconfigure and restart audio when display is enabled ASoC: omap-hdmi-audio: Set buffer bytes step constraint to 128 drivers/video/fbdev/omap2/dss/hdmi.h | 10 +++- drivers/video/fbdev/omap2/dss/hdmi4.c | 65 ++++++++++++++++++++----- drivers/video/fbdev/omap2/dss/hdmi5.c | 89 +++++++++++++++++++++++++++++------ sound/soc/omap/omap-hdmi-audio.c | 10 +++- 4 files changed, 146 insertions(+), 28 deletions(-) -- 1.9.1

3 7

[alsa-devel] Fwd: [PATCH 1/1] SPI : spi-pxa2xx : fix spi init of WM510205 codec via ACPI (resend)
by Christian Hartmann 28 Aug '15

28 Aug '15

Hi Mika, Dan et al. @Dan : thank you, I will use get_maintainer.pl as of now. @Mika: you mean I should prepare a new DSDT table and in this case overwrite the SPI value there with the 0 and load it (the DSDT table ) into a custom kernel build ?? I will do this (cause all what I want is to enable the codec and to hear sound) currently I have in the extracted DSDT table from this device the SPI1 controller for the "AUDI" device as seen in this snippet : Scope (_SB) { Device (LPEA) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "80860F28" /* Intel SST Audio DSP */) // _HID: Hardware ID Name (_CID, "80860F28" /* Intel SST Audio DSP */) // _CID: Compatible ID Name (_DDN, "Intel(R) Low Power Audio Controller - 80860F28") // _DDN: DOS Device Name Name (_SUB, "17AA7004") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (_DEP, Package (0x01) // _DEP: Dependencies { ^SPI1.AUDI }) Name (_PR0, Package (0x01) // _PR0: Power Resources for D0 { PLPE }) .... some lines later follows the AUDI device with Device (AUDI) { Name (_HID, "WM510205") // _HID: Hardware ID Name (_CID, "WM510205") // _CID: Compatible ID Name (_DDN, "Wolfson Microelectronics Audio WM5102") // _DDN: DOS Device Name Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (SBUF, ResourceTemplate () { SpiSerialBus (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI1", 0x00, ResourceConsumer, , ) GpioInt (Edge, ActiveLow, ExclusiveAndWake, PullNone, 0x0000, "\\_SB.GPO2", 0x00, ResourceConsumer, , ) { // Pin list 0x0004 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.I2C7.PMIC", 0x00, ResourceConsumer, , ) { // Pin list 0x0003 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO1", 0x00, ResourceConsumer, , ) { // Pin list 0x0017 } }) Return (SBUF) /* \_SB_.SPI1.AUDI._CRS.SBUF */ } .... I t makes sense to me, to change the current value(s) in the DSDT and I have read about it to use a custom DSDT table but I am not 100% sure yet (until I have tested it). If I understand you correct, than all I have to do is to change the 'SPI1' string in this dsdt to SPI0 like ^SPI1.AUDI to ^SPI0.AUDI and this block here SpiSerialBus (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI1", 0x00, ResourceConsumer, , becomes SpiSerialBus (0x0000, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI0", 0x00, ResourceConsumer, , For this case, could you please have a look in the whole DSDT table that I extract from this device? If nothing else seems relevant and the current system dsdt has to be changed as suggested, I will try this out as soon as possible @Jarkko : as you suggested , I attach the current system DSDT as a binary. I have also disassembled, and if someone needs it, I can also paste it somewhere else as a reference After all I will assembly the new dsdt and include it in my custom kernel builld as documented, but will wait for some new infos etc. thank you for you answers @all cheers chris 2015-06-25 9:36 GMT+02:00 Mika Westerberg <mika.westerberg(a)linux.intel.com>: > On Thu, Jun 25, 2015 at 09:29:31AM +0200, Christian Hartmann wrote: >> Hello Maintainer(s)s of SPI, >> >> hope I got all of you maintainers now. This email was already sent out >> last week to the alsa-devel list and I resent it here to you. >> >> I have one patch that enables the spi initialization of the WM510205 audio codec >> Without the patch, this message indicates that the codec chip fails >> with the current chipselect number in spi-pxa2xx: >> >> >> [ 0.296256] pxa2xx-spi 80860F0E:00: cs1 >= max 1 >> [ 0.296270] spi_master spi32766: failed to add SPI device >> WM510205:00 from ACPI >> >> So raise num_chipselect by one to enable the codec. The WM510202 is on >> chipselect = 1. > > The hardware only has one chipselect. > > You can fix this by specifying the correct chipselect in the ACPI > SpiSerialBus connector: > > SpiSerialBus (0, PolarityLow, FourWireMode, 8, ControllerInitiated, .. > > First parameter is the chipselect.

6 66

[alsa-devel] [PATCH] ALSA: usb-audio: correct the value cache check.
by Yao-Wen Mao 28 Aug '15

28 Aug '15

The check of cval->cached should be zero-based (including master channel). Signed-off-by: Yao-Wen Mao <yaowen(a)google.com> --- sound/usb/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index fd5c49e..f494dce 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2540,7 +2540,7 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) for (c = 0; c < MAX_CHANNELS; c++) { if (!(cval->cmask & (1 << c))) continue; - if (cval->cached & (1 << c)) { + if (cval->cached & (1 << (c + 1))) { err = snd_usb_set_cur_mix_value(cval, c + 1, idx, cval->cache_val[idx]); if (err < 0) -- 2.5.0.457.gab17608

2 1

[alsa-devel] [PATCH v2] ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning
by Oder Chiou 28 Aug '15

28 Aug '15

Add i2c shutdown function to prevent the pop sound of the headphone while the system is rebooting or shutdowning. It de-initials the jack detection function, and it cannot be turned off in _BIAS_OFF. If we don't de-initial it, the pop sound will be heard in the situation of powering off. And replace the related register settings from magic number to meaningful defined name. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++++++++----------- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 25 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 50f797c..4972bf3 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2806,13 +2806,13 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } regmap_write(rt5645->regmap, RT5645_JD_CTRL3, 0x00f0); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x1000); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, - 0x0004); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); msleep(100); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, 0); msleep(600); regmap_read(rt5645->regmap, RT5645_IN1_CTRL3, &val); @@ -2836,10 +2836,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } else { /* jack out */ rt5645->jack_type = 0; - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, 0x1000, - 0x1000); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, - 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, 0); if (rt5645->en_button_func) rt5645_enable_push_button_irq(codec, false); @@ -3458,13 +3458,26 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) return 0; } +static void rt5645_i2c_shutdown(struct i2c_client *i2c) +{ + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_RING2_SLEEVE_GND, RT5645_RING2_SLEEVE_GND); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, + RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, + 0); +} + static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, - .remove = rt5645_i2c_remove, + .remove = rt5645_i2c_remove, + .shutdown = rt5645_i2c_shutdown, .id_table = rt5645_i2c_id, }; module_i2c_driver(rt5645_i2c_driver); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index dffdda2..0e4cfc6 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2115,6 +2115,7 @@ enum { #define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) #define RT5645_MICINDET_MANU (0x1 << 7) +#define RT5645_RING2_SLEEVE_GND (0x1 << 5) /* Vendor ID (0xfd) */ #define RT5645_VER_C 0x2 -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] Applied "ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers" to the asoc tree
by Mark Brown 27 Aug '15

27 Aug '15

The patch ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 054bc835d27b558393541c32a209c01d89cda75a Mon Sep 17 00:00:00 2001 From: Luis de Bethencourt <luis(a)debethencourt.com> Date: Thu, 27 Aug 2015 17:04:01 +0200 Subject: [PATCH] ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers These platform drivers have a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/bcm/bcm2835-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 03fa1cbf8ec1..8c435beb263d 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -862,6 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = { {}, }; +MODULE_DEVICE_TABLE(of, bcm2835_i2s_of_match); + static struct platform_driver bcm2835_i2s_driver = { .probe = bcm2835_i2s_probe, .driver = { -- 2.5.0

1 0

[alsa-devel] [PATCH v5] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 27 Aug '15

27 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Tested on a setup with a single ICS-43432 (the device itself supports stereo operation using a hardware pin controlling left vs. right channel output). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V5: Not a lot left on the code now... Removed set_dai_fmt as it only checked the formats and never did any useful work. V4: Updated commit message that the driver has been tested with real hardware, after off-list comment on V3. V3: Fixed MODULE_LICENSE string. V2: Update after comments from Lars-Peter: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. I've left the checks in dai_set_fmt, as (IIUC) there currently doesn't seem to be a way for the ALSA framework to handle format constraints using the codec configuration struct. .../devicetree/bindings/sound/ics43432.txt | 17 +++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 77 ++++++++++++++++++++ 5 files changed, 101 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..4f202c1 --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,77 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPLv2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] Applied "ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning" to the asoc tree
by Mark Brown 27 Aug '15

27 Aug '15

The patch ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f2988afedf2c19880a3c65d79dfc7939e1b53d8a Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Thu, 27 Aug 2015 15:14:51 +0800 Subject: [PATCH] ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning Add i2c shutdown function to prevent the pop sound of the headphone while the system is rebooting or shutdowning. It de-initials the jack detection function, and it cannot be turned off in _BIAS_OFF. If we don't de-initial it, the pop sound will be heard in the situation of powering off. And replace the related register settings from magic number to meaningful defined name. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++++++++----------- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 25 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 0a4cb6bc3433..db50b035b00b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2806,13 +2806,13 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } regmap_write(rt5645->regmap, RT5645_JD_CTRL3, 0x00f0); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x1000); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, - 0x0004); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); msleep(100); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, 0); msleep(600); regmap_read(rt5645->regmap, RT5645_IN1_CTRL3, &val); @@ -2836,10 +2836,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } else { /* jack out */ rt5645->jack_type = 0; - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, 0x1000, - 0x1000); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, - 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, 0); if (rt5645->en_button_func) rt5645_enable_push_button_irq(codec, false); @@ -3449,6 +3449,18 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) return 0; } +static void rt5645_i2c_shutdown(struct i2c_client *i2c) +{ + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_RING2_SLEEVE_GND, RT5645_RING2_SLEEVE_GND); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, + RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, + 0); +} + static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", @@ -3456,7 +3468,8 @@ static struct i2c_driver rt5645_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, - .remove = rt5645_i2c_remove, + .remove = rt5645_i2c_remove, + .shutdown = rt5645_i2c_shutdown, .id_table = rt5645_i2c_id, }; module_i2c_driver(rt5645_i2c_driver); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 199b22fe1bf3..1987eb825e41 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2111,6 +2111,7 @@ enum { #define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) #define RT5645_MICINDET_MANU (0x1 << 7) +#define RT5645_RING2_SLEEVE_GND (0x1 << 5) /* Vendor ID (0xfd) */ #define RT5645_VER_C 0x2 -- 2.5.0

1 0

[alsa-devel] [PATCH 1/2] mfd: arizona: Add register bits to support the ANC block
by Charles Keepax 27 Aug '15

27 Aug '15

Some Arizona devices have a hardware ANC block present. This patch adds the registers necessary to configure this hardware block. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 185 +++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/registers.h | 70 +++++++++++++ 2 files changed, 255 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index dae04dd..71b84cd 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -1637,6 +1637,184 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000EF8, 0x0000 }, /* R3832 - ISRC 3 CTRL 3 */ { 0x00000F00, 0x0000 }, /* R3840 - Clock Control */ { 0x00000F01, 0x0000 }, /* R3841 - ANC_SRC */ + { 0x00000F08, 0x001c }, /* R3848 - ANC Coefficient */ + { 0x00000F09, 0x0000 }, /* R3849 - ANC Coefficient */ + { 0x00000F0B, 0x0000 }, /* R3851 - ANC Coefficient */ + { 0x00000F0C, 0x0000 }, /* R3852 - ANC Coefficient */ + { 0x00000F0D, 0x0000 }, /* R3853 - ANC Coefficient */ + { 0x00000F0E, 0x0000 }, /* R3854 - ANC Coefficient */ + { 0x00000F0F, 0x0000 }, /* R3855 - ANC Coefficient */ + { 0x00000F10, 0x0000 }, /* R3856 - ANC Coefficient */ + { 0x00000F11, 0x0000 }, /* R3857 - ANC Coefficient */ + { 0x00000F12, 0x0000 }, /* R3858 - ANC Coefficient */ + { 0x00000F15, 0x0000 }, /* R3861 - FCL Filter Control */ + { 0x00000F17, 0x0004 }, /* R3863 - FCL ADC Reformatter Control */ + { 0x00000F18, 0x0004 }, /* R3864 - ANC Coefficient */ + { 0x00000F19, 0x0002 }, /* R3865 - ANC Coefficient */ + { 0x00000F1A, 0x0000 }, /* R3866 - ANC Coefficient */ + { 0x00000F1B, 0x0010 }, /* R3867 - ANC Coefficient */ + { 0x00000F1C, 0x0000 }, /* R3868 - ANC Coefficient */ + { 0x00000F1D, 0x0000 }, /* R3869 - ANC Coefficient */ + { 0x00000F1E, 0x0000 }, /* R3870 - ANC Coefficient */ + { 0x00000F1F, 0x0000 }, /* R3871 - ANC Coefficient */ + { 0x00000F20, 0x0000 }, /* R3872 - ANC Coefficient */ + { 0x00000F21, 0x0000 }, /* R3873 - ANC Coefficient */ + { 0x00000F22, 0x0000 }, /* R3874 - ANC Coefficient */ + { 0x00000F23, 0x0000 }, /* R3875 - ANC Coefficient */ + { 0x00000F24, 0x0000 }, /* R3876 - ANC Coefficient */ + { 0x00000F25, 0x0000 }, /* R3877 - ANC Coefficient */ + { 0x00000F26, 0x0000 }, /* R3878 - ANC Coefficient */ + { 0x00000F27, 0x0000 }, /* R3879 - ANC Coefficient */ + { 0x00000F28, 0x0000 }, /* R3880 - ANC Coefficient */ + { 0x00000F29, 0x0000 }, /* R3881 - ANC Coefficient */ + { 0x00000F2A, 0x0000 }, /* R3882 - ANC Coefficient */ + { 0x00000F2B, 0x0000 }, /* R3883 - ANC Coefficient */ + { 0x00000F2C, 0x0000 }, /* R3884 - ANC Coefficient */ + { 0x00000F2D, 0x0000 }, /* R3885 - ANC Coefficient */ + { 0x00000F2E, 0x0000 }, /* R3886 - ANC Coefficient */ + { 0x00000F2F, 0x0000 }, /* R3887 - ANC Coefficient */ + { 0x00000F30, 0x0000 }, /* R3888 - ANC Coefficient */ + { 0x00000F31, 0x0000 }, /* R3889 - ANC Coefficient */ + { 0x00000F32, 0x0000 }, /* R3890 - ANC Coefficient */ + { 0x00000F33, 0x0000 }, /* R3891 - ANC Coefficient */ + { 0x00000F34, 0x0000 }, /* R3892 - ANC Coefficient */ + { 0x00000F35, 0x0000 }, /* R3893 - ANC Coefficient */ + { 0x00000F36, 0x0000 }, /* R3894 - ANC Coefficient */ + { 0x00000F37, 0x0000 }, /* R3895 - ANC Coefficient */ + { 0x00000F38, 0x0000 }, /* R3896 - ANC Coefficient */ + { 0x00000F39, 0x0000 }, /* R3897 - ANC Coefficient */ + { 0x00000F3A, 0x0000 }, /* R3898 - ANC Coefficient */ + { 0x00000F3B, 0x0000 }, /* R3899 - ANC Coefficient */ + { 0x00000F3C, 0x0000 }, /* R3900 - ANC Coefficient */ + { 0x00000F3D, 0x0000 }, /* R3901 - ANC Coefficient */ + { 0x00000F3E, 0x0000 }, /* R3902 - ANC Coefficient */ + { 0x00000F3F, 0x0000 }, /* R3903 - ANC Coefficient */ + { 0x00000F40, 0x0000 }, /* R3904 - ANC Coefficient */ + { 0x00000F41, 0x0000 }, /* R3905 - ANC Coefficient */ + { 0x00000F42, 0x0000 }, /* R3906 - ANC Coefficient */ + { 0x00000F43, 0x0000 }, /* R3907 - ANC Coefficient */ + { 0x00000F44, 0x0000 }, /* R3908 - ANC Coefficient */ + { 0x00000F45, 0x0000 }, /* R3909 - ANC Coefficient */ + { 0x00000F46, 0x0000 }, /* R3910 - ANC Coefficient */ + { 0x00000F47, 0x0000 }, /* R3911 - ANC Coefficient */ + { 0x00000F48, 0x0000 }, /* R3912 - ANC Coefficient */ + { 0x00000F49, 0x0000 }, /* R3913 - ANC Coefficient */ + { 0x00000F4A, 0x0000 }, /* R3914 - ANC Coefficient */ + { 0x00000F4B, 0x0000 }, /* R3915 - ANC Coefficient */ + { 0x00000F4C, 0x0000 }, /* R3916 - ANC Coefficient */ + { 0x00000F4D, 0x0000 }, /* R3917 - ANC Coefficient */ + { 0x00000F4E, 0x0000 }, /* R3918 - ANC Coefficient */ + { 0x00000F4F, 0x0000 }, /* R3919 - ANC Coefficient */ + { 0x00000F50, 0x0000 }, /* R3920 - ANC Coefficient */ + { 0x00000F51, 0x0000 }, /* R3921 - ANC Coefficient */ + { 0x00000F52, 0x0000 }, /* R3922 - ANC Coefficient */ + { 0x00000F53, 0x0000 }, /* R3923 - ANC Coefficient */ + { 0x00000F54, 0x0000 }, /* R3924 - ANC Coefficient */ + { 0x00000F55, 0x0000 }, /* R3925 - ANC Coefficient */ + { 0x00000F56, 0x0000 }, /* R3926 - ANC Coefficient */ + { 0x00000F57, 0x0000 }, /* R3927 - ANC Coefficient */ + { 0x00000F58, 0x0000 }, /* R3928 - ANC Coefficient */ + { 0x00000F59, 0x0000 }, /* R3929 - ANC Coefficient */ + { 0x00000F5A, 0x0000 }, /* R3930 - ANC Coefficient */ + { 0x00000F5B, 0x0000 }, /* R3931 - ANC Coefficient */ + { 0x00000F5C, 0x0000 }, /* R3932 - ANC Coefficient */ + { 0x00000F5D, 0x0000 }, /* R3933 - ANC Coefficient */ + { 0x00000F5E, 0x0000 }, /* R3934 - ANC Coefficient */ + { 0x00000F5F, 0x0000 }, /* R3935 - ANC Coefficient */ + { 0x00000F60, 0x0000 }, /* R3936 - ANC Coefficient */ + { 0x00000F61, 0x0000 }, /* R3937 - ANC Coefficient */ + { 0x00000F62, 0x0000 }, /* R3938 - ANC Coefficient */ + { 0x00000F63, 0x0000 }, /* R3939 - ANC Coefficient */ + { 0x00000F64, 0x0000 }, /* R3940 - ANC Coefficient */ + { 0x00000F65, 0x0000 }, /* R3941 - ANC Coefficient */ + { 0x00000F66, 0x0000 }, /* R3942 - ANC Coefficient */ + { 0x00000F67, 0x0000 }, /* R3943 - ANC Coefficient */ + { 0x00000F68, 0x0000 }, /* R3944 - ANC Coefficient */ + { 0x00000F69, 0x0000 }, /* R3945 - ANC Coefficient */ + { 0x00000F70, 0x0000 }, /* R3952 - FCR Filter Control */ + { 0x00000F72, 0x0004 }, /* R3954 - FCR ADC Reformatter Control */ + { 0x00000F73, 0x0004 }, /* R3955 - ANC Coefficient */ + { 0x00000F74, 0x0002 }, /* R3956 - ANC Coefficient */ + { 0x00000F75, 0x0000 }, /* R3957 - ANC Coefficient */ + { 0x00000F76, 0x0010 }, /* R3958 - ANC Coefficient */ + { 0x00000F77, 0x0000 }, /* R3959 - ANC Coefficient */ + { 0x00000F78, 0x0000 }, /* R3960 - ANC Coefficient */ + { 0x00000F79, 0x0000 }, /* R3961 - ANC Coefficient */ + { 0x00000F7A, 0x0000 }, /* R3962 - ANC Coefficient */ + { 0x00000F7B, 0x0000 }, /* R3963 - ANC Coefficient */ + { 0x00000F7C, 0x0000 }, /* R3964 - ANC Coefficient */ + { 0x00000F7D, 0x0000 }, /* R3965 - ANC Coefficient */ + { 0x00000F7E, 0x0000 }, /* R3966 - ANC Coefficient */ + { 0x00000F7F, 0x0000 }, /* R3967 - ANC Coefficient */ + { 0x00000F80, 0x0000 }, /* R3968 - ANC Coefficient */ + { 0x00000F81, 0x0000 }, /* R3969 - ANC Coefficient */ + { 0x00000F82, 0x0000 }, /* R3970 - ANC Coefficient */ + { 0x00000F83, 0x0000 }, /* R3971 - ANC Coefficient */ + { 0x00000F84, 0x0000 }, /* R3972 - ANC Coefficient */ + { 0x00000F85, 0x0000 }, /* R3973 - ANC Coefficient */ + { 0x00000F86, 0x0000 }, /* R3974 - ANC Coefficient */ + { 0x00000F87, 0x0000 }, /* R3975 - ANC Coefficient */ + { 0x00000F88, 0x0000 }, /* R3976 - ANC Coefficient */ + { 0x00000F89, 0x0000 }, /* R3977 - ANC Coefficient */ + { 0x00000F8A, 0x0000 }, /* R3978 - ANC Coefficient */ + { 0x00000F8B, 0x0000 }, /* R3979 - ANC Coefficient */ + { 0x00000F8C, 0x0000 }, /* R3980 - ANC Coefficient */ + { 0x00000F8D, 0x0000 }, /* R3981 - ANC Coefficient */ + { 0x00000F8E, 0x0000 }, /* R3982 - ANC Coefficient */ + { 0x00000F8F, 0x0000 }, /* R3983 - ANC Coefficient */ + { 0x00000F90, 0x0000 }, /* R3984 - ANC Coefficient */ + { 0x00000F91, 0x0000 }, /* R3985 - ANC Coefficient */ + { 0x00000F92, 0x0000 }, /* R3986 - ANC Coefficient */ + { 0x00000F93, 0x0000 }, /* R3987 - ANC Coefficient */ + { 0x00000F94, 0x0000 }, /* R3988 - ANC Coefficient */ + { 0x00000F95, 0x0000 }, /* R3989 - ANC Coefficient */ + { 0x00000F96, 0x0000 }, /* R3990 - ANC Coefficient */ + { 0x00000F97, 0x0000 }, /* R3991 - ANC Coefficient */ + { 0x00000F98, 0x0000 }, /* R3992 - ANC Coefficient */ + { 0x00000F99, 0x0000 }, /* R3993 - ANC Coefficient */ + { 0x00000F9A, 0x0000 }, /* R3994 - ANC Coefficient */ + { 0x00000F9B, 0x0000 }, /* R3995 - ANC Coefficient */ + { 0x00000F9C, 0x0000 }, /* R3996 - ANC Coefficient */ + { 0x00000F9D, 0x0000 }, /* R3997 - ANC Coefficient */ + { 0x00000F9E, 0x0000 }, /* R3998 - ANC Coefficient */ + { 0x00000F9F, 0x0000 }, /* R3999 - ANC Coefficient */ + { 0x00000FA0, 0x0000 }, /* R4000 - ANC Coefficient */ + { 0x00000FA1, 0x0000 }, /* R4001 - ANC Coefficient */ + { 0x00000FA2, 0x0000 }, /* R4002 - ANC Coefficient */ + { 0x00000FA3, 0x0000 }, /* R4003 - ANC Coefficient */ + { 0x00000FA4, 0x0000 }, /* R4004 - ANC Coefficient */ + { 0x00000FA5, 0x0000 }, /* R4005 - ANC Coefficient */ + { 0x00000FA6, 0x0000 }, /* R4006 - ANC Coefficient */ + { 0x00000FA7, 0x0000 }, /* R4007 - ANC Coefficient */ + { 0x00000FA8, 0x0000 }, /* R4008 - ANC Coefficient */ + { 0x00000FA9, 0x0000 }, /* R4009 - ANC Coefficient */ + { 0x00000FAA, 0x0000 }, /* R4010 - ANC Coefficient */ + { 0x00000FAB, 0x0000 }, /* R4011 - ANC Coefficient */ + { 0x00000FAC, 0x0000 }, /* R4012 - ANC Coefficient */ + { 0x00000FAD, 0x0000 }, /* R4013 - ANC Coefficient */ + { 0x00000FAE, 0x0000 }, /* R4014 - ANC Coefficient */ + { 0x00000FAF, 0x0000 }, /* R4015 - ANC Coefficient */ + { 0x00000FB0, 0x0000 }, /* R4016 - ANC Coefficient */ + { 0x00000FB1, 0x0000 }, /* R4017 - ANC Coefficient */ + { 0x00000FB2, 0x0000 }, /* R4018 - ANC Coefficient */ + { 0x00000FB3, 0x0000 }, /* R4019 - ANC Coefficient */ + { 0x00000FB4, 0x0000 }, /* R4020 - ANC Coefficient */ + { 0x00000FB5, 0x0000 }, /* R4021 - ANC Coefficient */ + { 0x00000FB6, 0x0000 }, /* R4022 - ANC Coefficient */ + { 0x00000FB7, 0x0000 }, /* R4023 - ANC Coefficient */ + { 0x00000FB8, 0x0000 }, /* R4024 - ANC Coefficient */ + { 0x00000FB9, 0x0000 }, /* R4025 - ANC Coefficient */ + { 0x00000FBA, 0x0000 }, /* R4026 - ANC Coefficient */ + { 0x00000FBB, 0x0000 }, /* R4027 - ANC Coefficient */ + { 0x00000FBC, 0x0000 }, /* R4028 - ANC Coefficient */ + { 0x00000FBD, 0x0000 }, /* R4029 - ANC Coefficient */ + { 0x00000FBE, 0x0000 }, /* R4030 - ANC Coefficient */ + { 0x00000FBF, 0x0000 }, /* R4031 - ANC Coefficient */ + { 0x00000FC0, 0x0000 }, /* R4032 - ANC Coefficient */ + { 0x00000FC1, 0x0000 }, /* R4033 - ANC Coefficient */ + { 0x00000FC2, 0x0000 }, /* R4034 - ANC Coefficient */ + { 0x00000FC3, 0x0000 }, /* R4035 - ANC Coefficient */ + { 0x00000FC4, 0x0000 }, /* R4036 - ANC Coefficient */ { 0x00001100, 0x0010 }, /* R4352 - DSP1 Control 1 */ { 0x00001101, 0x0000 }, /* R4353 - DSP1 Clocking 1 */ { 0x00001200, 0x0010 }, /* R4608 - DSP2 Control 1 */ @@ -2716,6 +2894,13 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_CLOCK_CONTROL: case ARIZONA_ANC_SRC: case ARIZONA_DSP_STATUS: + case ARIZONA_ANC_COEFF_START ... ARIZONA_ANC_COEFF_END: + case ARIZONA_FCL_FILTER_CONTROL: + case ARIZONA_FCL_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCL_COEFF_START ... ARIZONA_FCL_COEFF_END: + case ARIZONA_FCR_FILTER_CONTROL: + case ARIZONA_FCR_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCR_COEFF_START ... ARIZONA_FCR_COEFF_END: case ARIZONA_DSP1_CONTROL_1: case ARIZONA_DSP1_CLOCKING_1: case ARIZONA_DSP1_STATUS_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 11affb3..fdb12bf 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -1050,6 +1050,16 @@ #define ARIZONA_CLOCK_CONTROL 0xF00 #define ARIZONA_ANC_SRC 0xF01 #define ARIZONA_DSP_STATUS 0xF02 +#define ARIZONA_ANC_COEFF_START 0xF08 +#define ARIZONA_ANC_COEFF_END 0xF12 +#define ARIZONA_FCL_FILTER_CONTROL 0xF15 +#define ARIZONA_FCL_ADC_REFORMATTER_CONTROL 0xF17 +#define ARIZONA_FCL_COEFF_START 0xF18 +#define ARIZONA_FCL_COEFF_END 0xF69 +#define ARIZONA_FCR_FILTER_CONTROL 0xF70 +#define ARIZONA_FCR_ADC_REFORMATTER_CONTROL 0xF72 +#define ARIZONA_FCR_COEFF_START 0xF73 +#define ARIZONA_FCR_COEFF_END 0xFC4 #define ARIZONA_DSP1_CONTROL_1 0x1100 #define ARIZONA_DSP1_CLOCKING_1 0x1101 #define ARIZONA_DSP1_STATUS_1 0x1104 @@ -7823,6 +7833,66 @@ #define ARIZONA_ISRC3_NOTCH_ENA_WIDTH 1 /* ISRC3_NOTCH_ENA */ /* + * R3840 (0xF00) - Clock Control + */ +#define ARIZONA_EXT_NG_SEL_CLR 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_MASK 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_SHIFT 7 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_WIDTH 1 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_SET 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_MASK 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_SHIFT 6 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_WIDTH 1 /* EXT_NG_SEL_SET */ +#define ARIZONA_CLK_R_ENA_CLR 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_MASK 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_SHIFT 5 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_WIDTH 1 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_SET 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_MASK 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_SHIFT 4 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_WIDTH 1 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_CLR 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_MASK 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_SHIFT 3 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_WIDTH 1 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_SET 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_MASK 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_SHIFT 2 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_WIDTH 1 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_L_ENA_CLR 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_MASK 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_SHIFT 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_WIDTH 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_SET 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_MASK 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_SHIFT 0 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_WIDTH 1 /* CLK_L_ENA_SET */ + +/* + * R3841 (0xF01) - ANC SRC + */ +#define ARIZONA_IN_RXANCR_SEL_MASK 0x0070 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_SHIFT 4 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_WIDTH 3 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCL_SEL_MASK 0x0007 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_SHIFT 0 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_WIDTH 3 /* IN_RXANCL_SEL - [0:2] */ + +/* + * R3863 (0xF17) - FCL ADC Reformatter Control + */ +#define ARIZONA_FCL_MIC_MODE_SEL 0x000C /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_SHIFT 2 /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_WIDTH 2 /* FCL_MIC_MODE_SEL - [2:3] */ + +/* + * R3954 (0xF72) - FCR ADC Reformatter Control + */ +#define ARIZONA_FCR_MIC_MODE_SEL 0x000C /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_SHIFT 2 /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_WIDTH 2 /* FCR_MIC_MODE_SEL - [2:3] */ + +/* * R4352 (0x1100) - DSP1 Control 1 */ #define ARIZONA_DSP1_RATE_MASK 0x7800 /* DSP1_RATE - [14:11] */ -- 1.7.2.5

2 3

[alsa-devel] [PATCH v2 1/2] mfd: arizona: Add register bits to support the ANC block
by Charles Keepax 27 Aug '15

27 Aug '15

Some Arizona devices have a hardware ANC block present. This patch adds the registers necessary to configure this hardware block. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 185 +++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/registers.h | 70 +++++++++++++ 2 files changed, 255 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index dae04dd..71b84cd 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -1637,6 +1637,184 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000EF8, 0x0000 }, /* R3832 - ISRC 3 CTRL 3 */ { 0x00000F00, 0x0000 }, /* R3840 - Clock Control */ { 0x00000F01, 0x0000 }, /* R3841 - ANC_SRC */ + { 0x00000F08, 0x001c }, /* R3848 - ANC Coefficient */ + { 0x00000F09, 0x0000 }, /* R3849 - ANC Coefficient */ + { 0x00000F0B, 0x0000 }, /* R3851 - ANC Coefficient */ + { 0x00000F0C, 0x0000 }, /* R3852 - ANC Coefficient */ + { 0x00000F0D, 0x0000 }, /* R3853 - ANC Coefficient */ + { 0x00000F0E, 0x0000 }, /* R3854 - ANC Coefficient */ + { 0x00000F0F, 0x0000 }, /* R3855 - ANC Coefficient */ + { 0x00000F10, 0x0000 }, /* R3856 - ANC Coefficient */ + { 0x00000F11, 0x0000 }, /* R3857 - ANC Coefficient */ + { 0x00000F12, 0x0000 }, /* R3858 - ANC Coefficient */ + { 0x00000F15, 0x0000 }, /* R3861 - FCL Filter Control */ + { 0x00000F17, 0x0004 }, /* R3863 - FCL ADC Reformatter Control */ + { 0x00000F18, 0x0004 }, /* R3864 - ANC Coefficient */ + { 0x00000F19, 0x0002 }, /* R3865 - ANC Coefficient */ + { 0x00000F1A, 0x0000 }, /* R3866 - ANC Coefficient */ + { 0x00000F1B, 0x0010 }, /* R3867 - ANC Coefficient */ + { 0x00000F1C, 0x0000 }, /* R3868 - ANC Coefficient */ + { 0x00000F1D, 0x0000 }, /* R3869 - ANC Coefficient */ + { 0x00000F1E, 0x0000 }, /* R3870 - ANC Coefficient */ + { 0x00000F1F, 0x0000 }, /* R3871 - ANC Coefficient */ + { 0x00000F20, 0x0000 }, /* R3872 - ANC Coefficient */ + { 0x00000F21, 0x0000 }, /* R3873 - ANC Coefficient */ + { 0x00000F22, 0x0000 }, /* R3874 - ANC Coefficient */ + { 0x00000F23, 0x0000 }, /* R3875 - ANC Coefficient */ + { 0x00000F24, 0x0000 }, /* R3876 - ANC Coefficient */ + { 0x00000F25, 0x0000 }, /* R3877 - ANC Coefficient */ + { 0x00000F26, 0x0000 }, /* R3878 - ANC Coefficient */ + { 0x00000F27, 0x0000 }, /* R3879 - ANC Coefficient */ + { 0x00000F28, 0x0000 }, /* R3880 - ANC Coefficient */ + { 0x00000F29, 0x0000 }, /* R3881 - ANC Coefficient */ + { 0x00000F2A, 0x0000 }, /* R3882 - ANC Coefficient */ + { 0x00000F2B, 0x0000 }, /* R3883 - ANC Coefficient */ + { 0x00000F2C, 0x0000 }, /* R3884 - ANC Coefficient */ + { 0x00000F2D, 0x0000 }, /* R3885 - ANC Coefficient */ + { 0x00000F2E, 0x0000 }, /* R3886 - ANC Coefficient */ + { 0x00000F2F, 0x0000 }, /* R3887 - ANC Coefficient */ + { 0x00000F30, 0x0000 }, /* R3888 - ANC Coefficient */ + { 0x00000F31, 0x0000 }, /* R3889 - ANC Coefficient */ + { 0x00000F32, 0x0000 }, /* R3890 - ANC Coefficient */ + { 0x00000F33, 0x0000 }, /* R3891 - ANC Coefficient */ + { 0x00000F34, 0x0000 }, /* R3892 - ANC Coefficient */ + { 0x00000F35, 0x0000 }, /* R3893 - ANC Coefficient */ + { 0x00000F36, 0x0000 }, /* R3894 - ANC Coefficient */ + { 0x00000F37, 0x0000 }, /* R3895 - ANC Coefficient */ + { 0x00000F38, 0x0000 }, /* R3896 - ANC Coefficient */ + { 0x00000F39, 0x0000 }, /* R3897 - ANC Coefficient */ + { 0x00000F3A, 0x0000 }, /* R3898 - ANC Coefficient */ + { 0x00000F3B, 0x0000 }, /* R3899 - ANC Coefficient */ + { 0x00000F3C, 0x0000 }, /* R3900 - ANC Coefficient */ + { 0x00000F3D, 0x0000 }, /* R3901 - ANC Coefficient */ + { 0x00000F3E, 0x0000 }, /* R3902 - ANC Coefficient */ + { 0x00000F3F, 0x0000 }, /* R3903 - ANC Coefficient */ + { 0x00000F40, 0x0000 }, /* R3904 - ANC Coefficient */ + { 0x00000F41, 0x0000 }, /* R3905 - ANC Coefficient */ + { 0x00000F42, 0x0000 }, /* R3906 - ANC Coefficient */ + { 0x00000F43, 0x0000 }, /* R3907 - ANC Coefficient */ + { 0x00000F44, 0x0000 }, /* R3908 - ANC Coefficient */ + { 0x00000F45, 0x0000 }, /* R3909 - ANC Coefficient */ + { 0x00000F46, 0x0000 }, /* R3910 - ANC Coefficient */ + { 0x00000F47, 0x0000 }, /* R3911 - ANC Coefficient */ + { 0x00000F48, 0x0000 }, /* R3912 - ANC Coefficient */ + { 0x00000F49, 0x0000 }, /* R3913 - ANC Coefficient */ + { 0x00000F4A, 0x0000 }, /* R3914 - ANC Coefficient */ + { 0x00000F4B, 0x0000 }, /* R3915 - ANC Coefficient */ + { 0x00000F4C, 0x0000 }, /* R3916 - ANC Coefficient */ + { 0x00000F4D, 0x0000 }, /* R3917 - ANC Coefficient */ + { 0x00000F4E, 0x0000 }, /* R3918 - ANC Coefficient */ + { 0x00000F4F, 0x0000 }, /* R3919 - ANC Coefficient */ + { 0x00000F50, 0x0000 }, /* R3920 - ANC Coefficient */ + { 0x00000F51, 0x0000 }, /* R3921 - ANC Coefficient */ + { 0x00000F52, 0x0000 }, /* R3922 - ANC Coefficient */ + { 0x00000F53, 0x0000 }, /* R3923 - ANC Coefficient */ + { 0x00000F54, 0x0000 }, /* R3924 - ANC Coefficient */ + { 0x00000F55, 0x0000 }, /* R3925 - ANC Coefficient */ + { 0x00000F56, 0x0000 }, /* R3926 - ANC Coefficient */ + { 0x00000F57, 0x0000 }, /* R3927 - ANC Coefficient */ + { 0x00000F58, 0x0000 }, /* R3928 - ANC Coefficient */ + { 0x00000F59, 0x0000 }, /* R3929 - ANC Coefficient */ + { 0x00000F5A, 0x0000 }, /* R3930 - ANC Coefficient */ + { 0x00000F5B, 0x0000 }, /* R3931 - ANC Coefficient */ + { 0x00000F5C, 0x0000 }, /* R3932 - ANC Coefficient */ + { 0x00000F5D, 0x0000 }, /* R3933 - ANC Coefficient */ + { 0x00000F5E, 0x0000 }, /* R3934 - ANC Coefficient */ + { 0x00000F5F, 0x0000 }, /* R3935 - ANC Coefficient */ + { 0x00000F60, 0x0000 }, /* R3936 - ANC Coefficient */ + { 0x00000F61, 0x0000 }, /* R3937 - ANC Coefficient */ + { 0x00000F62, 0x0000 }, /* R3938 - ANC Coefficient */ + { 0x00000F63, 0x0000 }, /* R3939 - ANC Coefficient */ + { 0x00000F64, 0x0000 }, /* R3940 - ANC Coefficient */ + { 0x00000F65, 0x0000 }, /* R3941 - ANC Coefficient */ + { 0x00000F66, 0x0000 }, /* R3942 - ANC Coefficient */ + { 0x00000F67, 0x0000 }, /* R3943 - ANC Coefficient */ + { 0x00000F68, 0x0000 }, /* R3944 - ANC Coefficient */ + { 0x00000F69, 0x0000 }, /* R3945 - ANC Coefficient */ + { 0x00000F70, 0x0000 }, /* R3952 - FCR Filter Control */ + { 0x00000F72, 0x0004 }, /* R3954 - FCR ADC Reformatter Control */ + { 0x00000F73, 0x0004 }, /* R3955 - ANC Coefficient */ + { 0x00000F74, 0x0002 }, /* R3956 - ANC Coefficient */ + { 0x00000F75, 0x0000 }, /* R3957 - ANC Coefficient */ + { 0x00000F76, 0x0010 }, /* R3958 - ANC Coefficient */ + { 0x00000F77, 0x0000 }, /* R3959 - ANC Coefficient */ + { 0x00000F78, 0x0000 }, /* R3960 - ANC Coefficient */ + { 0x00000F79, 0x0000 }, /* R3961 - ANC Coefficient */ + { 0x00000F7A, 0x0000 }, /* R3962 - ANC Coefficient */ + { 0x00000F7B, 0x0000 }, /* R3963 - ANC Coefficient */ + { 0x00000F7C, 0x0000 }, /* R3964 - ANC Coefficient */ + { 0x00000F7D, 0x0000 }, /* R3965 - ANC Coefficient */ + { 0x00000F7E, 0x0000 }, /* R3966 - ANC Coefficient */ + { 0x00000F7F, 0x0000 }, /* R3967 - ANC Coefficient */ + { 0x00000F80, 0x0000 }, /* R3968 - ANC Coefficient */ + { 0x00000F81, 0x0000 }, /* R3969 - ANC Coefficient */ + { 0x00000F82, 0x0000 }, /* R3970 - ANC Coefficient */ + { 0x00000F83, 0x0000 }, /* R3971 - ANC Coefficient */ + { 0x00000F84, 0x0000 }, /* R3972 - ANC Coefficient */ + { 0x00000F85, 0x0000 }, /* R3973 - ANC Coefficient */ + { 0x00000F86, 0x0000 }, /* R3974 - ANC Coefficient */ + { 0x00000F87, 0x0000 }, /* R3975 - ANC Coefficient */ + { 0x00000F88, 0x0000 }, /* R3976 - ANC Coefficient */ + { 0x00000F89, 0x0000 }, /* R3977 - ANC Coefficient */ + { 0x00000F8A, 0x0000 }, /* R3978 - ANC Coefficient */ + { 0x00000F8B, 0x0000 }, /* R3979 - ANC Coefficient */ + { 0x00000F8C, 0x0000 }, /* R3980 - ANC Coefficient */ + { 0x00000F8D, 0x0000 }, /* R3981 - ANC Coefficient */ + { 0x00000F8E, 0x0000 }, /* R3982 - ANC Coefficient */ + { 0x00000F8F, 0x0000 }, /* R3983 - ANC Coefficient */ + { 0x00000F90, 0x0000 }, /* R3984 - ANC Coefficient */ + { 0x00000F91, 0x0000 }, /* R3985 - ANC Coefficient */ + { 0x00000F92, 0x0000 }, /* R3986 - ANC Coefficient */ + { 0x00000F93, 0x0000 }, /* R3987 - ANC Coefficient */ + { 0x00000F94, 0x0000 }, /* R3988 - ANC Coefficient */ + { 0x00000F95, 0x0000 }, /* R3989 - ANC Coefficient */ + { 0x00000F96, 0x0000 }, /* R3990 - ANC Coefficient */ + { 0x00000F97, 0x0000 }, /* R3991 - ANC Coefficient */ + { 0x00000F98, 0x0000 }, /* R3992 - ANC Coefficient */ + { 0x00000F99, 0x0000 }, /* R3993 - ANC Coefficient */ + { 0x00000F9A, 0x0000 }, /* R3994 - ANC Coefficient */ + { 0x00000F9B, 0x0000 }, /* R3995 - ANC Coefficient */ + { 0x00000F9C, 0x0000 }, /* R3996 - ANC Coefficient */ + { 0x00000F9D, 0x0000 }, /* R3997 - ANC Coefficient */ + { 0x00000F9E, 0x0000 }, /* R3998 - ANC Coefficient */ + { 0x00000F9F, 0x0000 }, /* R3999 - ANC Coefficient */ + { 0x00000FA0, 0x0000 }, /* R4000 - ANC Coefficient */ + { 0x00000FA1, 0x0000 }, /* R4001 - ANC Coefficient */ + { 0x00000FA2, 0x0000 }, /* R4002 - ANC Coefficient */ + { 0x00000FA3, 0x0000 }, /* R4003 - ANC Coefficient */ + { 0x00000FA4, 0x0000 }, /* R4004 - ANC Coefficient */ + { 0x00000FA5, 0x0000 }, /* R4005 - ANC Coefficient */ + { 0x00000FA6, 0x0000 }, /* R4006 - ANC Coefficient */ + { 0x00000FA7, 0x0000 }, /* R4007 - ANC Coefficient */ + { 0x00000FA8, 0x0000 }, /* R4008 - ANC Coefficient */ + { 0x00000FA9, 0x0000 }, /* R4009 - ANC Coefficient */ + { 0x00000FAA, 0x0000 }, /* R4010 - ANC Coefficient */ + { 0x00000FAB, 0x0000 }, /* R4011 - ANC Coefficient */ + { 0x00000FAC, 0x0000 }, /* R4012 - ANC Coefficient */ + { 0x00000FAD, 0x0000 }, /* R4013 - ANC Coefficient */ + { 0x00000FAE, 0x0000 }, /* R4014 - ANC Coefficient */ + { 0x00000FAF, 0x0000 }, /* R4015 - ANC Coefficient */ + { 0x00000FB0, 0x0000 }, /* R4016 - ANC Coefficient */ + { 0x00000FB1, 0x0000 }, /* R4017 - ANC Coefficient */ + { 0x00000FB2, 0x0000 }, /* R4018 - ANC Coefficient */ + { 0x00000FB3, 0x0000 }, /* R4019 - ANC Coefficient */ + { 0x00000FB4, 0x0000 }, /* R4020 - ANC Coefficient */ + { 0x00000FB5, 0x0000 }, /* R4021 - ANC Coefficient */ + { 0x00000FB6, 0x0000 }, /* R4022 - ANC Coefficient */ + { 0x00000FB7, 0x0000 }, /* R4023 - ANC Coefficient */ + { 0x00000FB8, 0x0000 }, /* R4024 - ANC Coefficient */ + { 0x00000FB9, 0x0000 }, /* R4025 - ANC Coefficient */ + { 0x00000FBA, 0x0000 }, /* R4026 - ANC Coefficient */ + { 0x00000FBB, 0x0000 }, /* R4027 - ANC Coefficient */ + { 0x00000FBC, 0x0000 }, /* R4028 - ANC Coefficient */ + { 0x00000FBD, 0x0000 }, /* R4029 - ANC Coefficient */ + { 0x00000FBE, 0x0000 }, /* R4030 - ANC Coefficient */ + { 0x00000FBF, 0x0000 }, /* R4031 - ANC Coefficient */ + { 0x00000FC0, 0x0000 }, /* R4032 - ANC Coefficient */ + { 0x00000FC1, 0x0000 }, /* R4033 - ANC Coefficient */ + { 0x00000FC2, 0x0000 }, /* R4034 - ANC Coefficient */ + { 0x00000FC3, 0x0000 }, /* R4035 - ANC Coefficient */ + { 0x00000FC4, 0x0000 }, /* R4036 - ANC Coefficient */ { 0x00001100, 0x0010 }, /* R4352 - DSP1 Control 1 */ { 0x00001101, 0x0000 }, /* R4353 - DSP1 Clocking 1 */ { 0x00001200, 0x0010 }, /* R4608 - DSP2 Control 1 */ @@ -2716,6 +2894,13 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_CLOCK_CONTROL: case ARIZONA_ANC_SRC: case ARIZONA_DSP_STATUS: + case ARIZONA_ANC_COEFF_START ... ARIZONA_ANC_COEFF_END: + case ARIZONA_FCL_FILTER_CONTROL: + case ARIZONA_FCL_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCL_COEFF_START ... ARIZONA_FCL_COEFF_END: + case ARIZONA_FCR_FILTER_CONTROL: + case ARIZONA_FCR_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCR_COEFF_START ... ARIZONA_FCR_COEFF_END: case ARIZONA_DSP1_CONTROL_1: case ARIZONA_DSP1_CLOCKING_1: case ARIZONA_DSP1_STATUS_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 11affb3..fdb12bf 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -1050,6 +1050,16 @@ #define ARIZONA_CLOCK_CONTROL 0xF00 #define ARIZONA_ANC_SRC 0xF01 #define ARIZONA_DSP_STATUS 0xF02 +#define ARIZONA_ANC_COEFF_START 0xF08 +#define ARIZONA_ANC_COEFF_END 0xF12 +#define ARIZONA_FCL_FILTER_CONTROL 0xF15 +#define ARIZONA_FCL_ADC_REFORMATTER_CONTROL 0xF17 +#define ARIZONA_FCL_COEFF_START 0xF18 +#define ARIZONA_FCL_COEFF_END 0xF69 +#define ARIZONA_FCR_FILTER_CONTROL 0xF70 +#define ARIZONA_FCR_ADC_REFORMATTER_CONTROL 0xF72 +#define ARIZONA_FCR_COEFF_START 0xF73 +#define ARIZONA_FCR_COEFF_END 0xFC4 #define ARIZONA_DSP1_CONTROL_1 0x1100 #define ARIZONA_DSP1_CLOCKING_1 0x1101 #define ARIZONA_DSP1_STATUS_1 0x1104 @@ -7823,6 +7833,66 @@ #define ARIZONA_ISRC3_NOTCH_ENA_WIDTH 1 /* ISRC3_NOTCH_ENA */ /* + * R3840 (0xF00) - Clock Control + */ +#define ARIZONA_EXT_NG_SEL_CLR 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_MASK 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_SHIFT 7 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_WIDTH 1 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_SET 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_MASK 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_SHIFT 6 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_WIDTH 1 /* EXT_NG_SEL_SET */ +#define ARIZONA_CLK_R_ENA_CLR 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_MASK 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_SHIFT 5 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_WIDTH 1 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_SET 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_MASK 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_SHIFT 4 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_WIDTH 1 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_CLR 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_MASK 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_SHIFT 3 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_WIDTH 1 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_SET 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_MASK 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_SHIFT 2 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_WIDTH 1 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_L_ENA_CLR 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_MASK 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_SHIFT 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_WIDTH 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_SET 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_MASK 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_SHIFT 0 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_WIDTH 1 /* CLK_L_ENA_SET */ + +/* + * R3841 (0xF01) - ANC SRC + */ +#define ARIZONA_IN_RXANCR_SEL_MASK 0x0070 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_SHIFT 4 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_WIDTH 3 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCL_SEL_MASK 0x0007 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_SHIFT 0 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_WIDTH 3 /* IN_RXANCL_SEL - [0:2] */ + +/* + * R3863 (0xF17) - FCL ADC Reformatter Control + */ +#define ARIZONA_FCL_MIC_MODE_SEL 0x000C /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_SHIFT 2 /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_WIDTH 2 /* FCL_MIC_MODE_SEL - [2:3] */ + +/* + * R3954 (0xF72) - FCR ADC Reformatter Control + */ +#define ARIZONA_FCR_MIC_MODE_SEL 0x000C /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_SHIFT 2 /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_WIDTH 2 /* FCR_MIC_MODE_SEL - [2:3] */ + +/* * R4352 (0x1100) - DSP1 Control 1 */ #define ARIZONA_DSP1_RATE_MASK 0x7800 /* DSP1_RATE - [14:11] */ -- 1.7.2.5

1 1

[alsa-devel] Out-of-tree HDMI audio driver for BayTrail-T
by David Henningsson 27 Aug '15

27 Aug '15

I don't usually deal with out-of-tree drivers, but when Canonical enabled the Intel Compute Stick [1] (which is using a BayTrail-T processor), we were using an out-of-tree driver to enable HDMI audio. The driver has earlier been made public by Intel [2], and I've put my modifications to it here [3]. Unfortunately, it looks like the effort and hardware knowledge required to make this driver upstreamable is beyond what I can currently manage. Also, I cannot answer for whether or not other parties (such as Intel) plan to upstream the driver. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] http://www.intel.com/content/www/us/en/compute-stick/intel-compute-stick.ht… [2] https://github.com/01org/baytrailaudio [3] http://kernel.ubuntu.com/git/diwic/baytrail-hdmi-audio.git/log/?h=baytrail-…

2 1

[alsa-devel] [PATCH v5 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 27 Aug '15

27 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..aabebcb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,7 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

5 18

[alsa-devel] Asus X553M audio issue SOLVED
by Rafael Gonzalez 27 Aug '15

27 Aug '15

Guys, I was having a nice chat with debianuser from #alsa on freenode, and he helped me fix my audio issues with my laptop that dual boots windows and linux. So, the easiest way to show you is post the log file of our conversation. ud> Hi there. You've been helping me with my card issues. <salviadud> I want to share what progress I've got. <salviadud> The machine is a dual boot laptop. <salviadud> I also had another issue with ACPI, but I solved that by turning D-Bus on <salviadud> In fact, after d-bus, i can see the d-bus message when I plug in my headphones to the jack. <salviadud> I don't think that windows has been tampering with the soundcard so that I need to do a full electrial reset on it. <salviadud> I just think the alsa driver is ok, but there's something missing. <salviadud> I need to develop it, or something. <salviadud> I want to know who should I talk to. <salviadud> The card vendor (Realtek) <salviadud> The laptop manufacturer (asus) <salviadud> ? <salviadud> If you got any ideas, I am all ears. so to speak <salviadud> And my logic is this: there's no default option to choose between headphones/headsets/speakers. And like I said before, I can only get audio on headphones with 3 ringed jacks. <salviadud> Windows on the other hand, has this nifty userspace program from realtek... I don't like crabs today. <debianuser> Have you managed to boot any other linux liveusb on that laptop? Ubuntu? Fedora? Mageia? Debian? <salviadud> Funny that you mention that <salviadud> I did try to load a kubuntu 14.10 <salviadud> but it hanged on me <salviadud> Same thing happened with mint. <salviadud> So, the other live cd I could load was a slax live usb stick. <salviadud> But that one in particular loaded a different driver <salviadud> and the sound that came out, was crackled. <salviadud> I can still get sound on this thing, but it has to be from external speakers, and hdmi out <salviadud> Although, I think I need to play with hda analyzer for that. <debianuser> I thought that internal speakers work fine, and you only can't hear the sound from headphones... <salviadud> Sorry <salviadud> I meant what you just said <salviadud> I called them external by mistake <salviadud> laptop speakers work, headphones don't <debianuser> Ah, ok :) <salviadud> So, I want to be proactive about this <salviadud> In what other way can I debug this <salviadud> so that I can finally reverse engineer it <salviadud> I think its an unusual thing to have a jack decide on what to output based on the number of rings. <salviadud> I guess I'm old. <salviadud> I'm used to left-right jacks. <salviadud> They are simple pieces of hardware <salviadud> And well, I just love linux, I can't stand windows as my main. <salviadud> Especially for music. <debianuser> It's just there doesn't seem to be anything special about your laptop or codec. It looks normal, but it behaves unusual, as when your mixer values were different from codec dump... <debianuser> Have you tried "full poweroff" by the way? <salviadud> I haven't done it, I need to take out some screws for that. <debianuser> What for? <salviadud> The battery <salviadud> Its not plug-n-play <salviadud> its inside where the hdd lies <debianuser> Ah, well, try just shut it down and unplug power supply, without removing the battery <debianuser> It's just has to be poweroff, not suspend, not hibernate... <salviadud> I'm pretty sure I've done that. <salviadud> start it up on just the battery. <debianuser> (actually sometimes suspend or hibernate may help too, for a different reason, but it's better to try poweroff first...) <salviadud> I'll do it when I get off work <salviadud> Still, getting ahead of it. Lets say it stays the same. <debianuser> The important part is not how you start it, but to power it off for a few minutes, then boot linux first. You can boot it with power supply attached, just boot linux first :) <salviadud> mmmmm <salviadud> Maybe its that <salviadud> the missing link <salviadud> I have never done that <salviadud> after a certain time <salviadud> windows doesn't allow me to access uefi <salviadud> and it boots directly into windows <salviadud> but <salviadud> its because its the first option <salviadud> I just need to change that <salviadud> Ok, let me rephrase <salviadud> If I shut down on linux <salviadud> and I dunno, 10 minutes pass <salviadud> I turn on my computer again, and I want to change my boot option by pressing ESC. <salviadud> I can't <salviadud> It goes into fast boot mode <salviadud> I can't control it <salviadud> So, if I change my boot option <salviadud> I should be able to do that. <salviadud> Ok, please excuse my long explanation. <debianuser> (it was reported to help for different laptops, including laptops having the same codec: launchpad.net/bugs/1087428 comment 4) <debianuser> How do you get to the linux then? <debianuser> I mean how can you boot linux at all? <salviadud> The first time was so hard. <salviadud> I had to go into repair mode in windows. <salviadud> So I could get into the bios. <salviadud> After that, I could choose the cdrom option for boot <salviadud> and that's how I installed it. <salviadud> I don't use a boot manager like grub or lilo <salviadud> just the Bios utility <salviadud> so <salviadud> I usually boot into windows first. Then I do restart, and then it allows me to select my different boot options <salviadud> Its so annoying <salviadud> I'm definetely changing the boot order <debianuser> What if you select restart in windows, get into that options list, and then just press the power button? <salviadud> That link you gave me <salviadud> same exact issue <salviadud> Its too much of a hazzle <salviadud> I want a permanent solution <debianuser> What if you select restart in windows, get into that options list, and then just press the power button - I mean does the laptop turns off? <salviadud> Yeah, it should <salviadud> but when I turn it on, its gonna know it was powered off for a while <salviadud> and it will go into the first boot option <salviadud> which is windows <debianuser> Hm... Is there some hotkey to stop it? Maybe you need to hold Ctrl or Alt? <salviadud> Nope, its NSA loving spyware windows <salviadud> what did you expect? hehehe <salviadud> I know, its terrible <debianuser> Is this the fastboot?: https://www.asus.com/support/FAQ/1005535/ <salviadud> My bios does not look as fancy as that one <salviadud> but yeah, that's the option. <salviadud> I think I had to disable it just to get access to other boot options <salviadud> I think that bios might be for a desktop <salviadud> I dunno, this laptop just feels really secure tight about working on windows <salviadud> and not on linux <salviadud> I'll give it a go in a couple of hourse, I'll tell ya what happened tomorrow <salviadud> U are a lot of help dude, thank you much. <debianuser> It's just its default setting. It may be useful for windows, I mean regular users often don't need boot menu, but they want OS to boot faster :) (I remember how people were impressed when I booted simple linux gui in 10 seconds). <salviadud> 10 seconds? that's still impressive, hehe <salviadud> u know what, I'm gonna try this now <salviadud> brb <salviadud> dude, it totally worked <salviadud> I'm listening to my tunes on audacious <debianuser> Great! :) <salviadud> as a token of my appreciation <salviadud> here's my ftp server <salviadud> anachronox.dyndns.za.org <salviadud> user:ftpuser <salviadud> pass:meridian2002 <salviadud> its full of music roms n pr0n <salviadud> hehe <debianuser> As for 10 seconds boot - it's just a generic kernel, with a custom init script: about 5 seconds to probe/load modules for available devices, 2 seconds for Xorg to start and switch to graphical mode, and 3 more seconds to boot firefox (I don't remember exact timings but something like that). <debianuser> There're better boot times on youtube: https://youtu.be/whkmChncjRk https://youtu.be/XcbKe3ct610 https://youtu.be/fMM-ZZijRc4 https://youtu.be/s7NxCM8ryF8 3-5 seconds - that's impressive. :) But I was too lazy to build a custom kernel for that. :) <salviadud> I could do something similar... <salviadud> I got a custom kernel <salviadud> whatever... I don't care about boot times. <salviadud> thanx again man. <debianuser> You're welcome! Feel free to ask whenever you want to know anything else :) <salviadud> that ftp server gets shutdown when I got to sleep, keep that in mind. <salviadud> I turn it on, every now and then. * debianuser saves a not that full poweroff makes sound headphones working on a dual-boot Asus X553MA (Codec: Realtek ALC270) <debianuser> *a note <debianuser> And thanks for reporting the problem, by the way. If you want to spend a few more minutes for it we can try to compare working and broken settings to check what exactly was different... Maybe we find a way to undo that... <debianuser> (basically I'll just ask you to dump alsa-info 2 more times with one special optio) <debianuser> (and if we're lucky we'll have something to report to alsa-devel mailing list, and devs would be able to include that into the kernel to make it working in next kernel version out of box) <salviadud> ok <salviadud> sounds good <salviadud> How do I run the script, headphones on or off? <debianuser> There're vendor-specific coefficients that control the chip but sometimes are not reset on reboot. To include those int alsa-info run: `echo 1 | sudo tee /sys/module/snd_hda_codec/parameters/dump_coef` then dump alsa-info as usual with headphones plugged in. That's the "working" dump. <debianuser> Then we'll break it: reboot into windows, reboot back into linux, make sure headphones don't work, run `echo 1 | sudo tee /sys/module/snd_hda_codec/parameters/dump_coef` and dump alsa-info with headphones plugged in again. That would be "broken" dump. <salviadud> ok, I don't sudo, so I'll just run that as root. <debianuser> Then we'll diff them and see if there's some difference that we can undo... <salviadud> ok, working dump coming <debianuser> yeah `echo 1 > /sys/module/snd_hda_codec/parameters/dump_coef` as root will work too <salviadud> http://www.alsa-project.org/db/?f=abda2cd2c9137fbf2319015707cf2d8d30aaf773 <salviadud> I ran alsa-info after the echo <salviadud> let me go back into windows..., and see if it breaks on reboot * Registro cargado desde Wed Aug 26 18:45:41 2015 <salviadud> ok, I got broken audio <debianuser> Now let's get broken alsa-info and diff them! <salviadud> http://www.alsa-project.org/db/?f=4d597f8a6f9eaa0f0823d195d5fa5093ea71d0eb <salviadud> Well, I'm glad it worked <salviadud> and hell, I even left windows as a special option <salviadud> no need to erase boot options <salviadud> I just disabled it so I can boot straight into linux <salviadud> feels great man. <debianuser> Hah! The difference is just in 2 coefficients: Coeff 0x04: 0x9827 (working) / 0x9027 (broken) and Coeff 0x0d: 0x4440 (working) / 0xc440 (broken) <debianuser> Let's try to change them back and see if that would fix headphones! <salviadud> so, How do I do that? <debianuser> To do that install "alsa-tools" package if you don't have it yet and run these 4 commands in the following order: <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x04 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x9827 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x0d <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x4440 <salviadud> i got alsatools, let me give it a go. <debianuser> (not sure what which one of them is doing, so let's set them both and see if that will change anything) <salviadud> I got sound <salviadud> You are pro <salviadud> although, I really wouldn't like to do that every time. * salviadud knows that sounded dumb <debianuser> Great! You don't have to do that every time (well, you can add those to /etc/rc.local, or whatever is used in gentoo instead). But the point was to find what was causing the issue. <salviadud> that's why we're doing this test... hehe <salviadud> Yeah, I think its a small, yet important contribution. * debianuser wonders which one of those coefs breaks the sound... <salviadud> I think its safe to just keep both like that. <debianuser> Can you check that by the way? For example revert one of them back to "broken" value and check if you still have the sound from headphones: <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x04 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x9027 <salviadud> I will <salviadud> for science <salviadud> that broke it <salviadud> I got no sound <salviadud> What else do I do? <salviadud> or is that it? <debianuser> How about the other way around? 0x04 to working, 0x0d to broken: <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x04 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x9827 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x0d <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0xc440 <debianuser> (that's the last test) <salviadud> it works <salviadud> I think the other coefficient might be the hdmi <salviadud> But I can't debug that one over here right now <salviadud> still, that's just a wild guess <salviadud> Well, you got all you need to report debugging info? <debianuser> Well, that's all we need for the report. Now if you still feel the power to make the world better (just a little bit) - you can report that to alsa-devel mailing list. Include both alsa-info links, tell which one of them works and which one has headphones broken after windows reboot, and which hda-verb commands pair makes headphones working again. :) <salviadud> I'm just gonna send the log of this conversation <debianuser> They may ask you to test a kernel patch (they often do that) before sending it upstream to Linus. That's why it's better for you to report it. <salviadud> if you don't minnd <debianuser> I don't :) <salviadud> alsa rocks! <debianuser> especially when it works :) <salviadud> \O/ <debianuser> Thank you for your patience, for reporting the problem and for doing all those checks!

1 0

[alsa-devel] different acpi_ipc_irq_index among baytrail boards issue
by Michele Curti 26 Aug '15

26 Aug '15

Hi, some baytrail machines, although belonging to the same platform, have the ADSP IRQ number at different ACPI index (.acpi_ipc_irq_index = 0 instead of 5). Not an issue for now, since there is no support for these machines (like the Asus X205TA) at the moment, but if the driver will work, a patch like the following would be acceptable (I ask so I can try to keep a sort of patchset)? * declare a new struct type sst_acpi_info where to place acpi infos lpe_res_index, ddr_index, ipc_irq_index * create a new 'const struct sst_acpi_info *acpi_info' field into the struct sst_platform_info (to be managed like the others ipc_info, res_info, ...) * move the acpi infos from the byt_rvp_res_info variable to a new byt_acpi_res_info one In this way one could duplicate byt_rvp_acpi_info instead of the whole byt_rvp_res_info (which seems "not good" to me, or it's ok?) to manage different acpi indexes. Hope I was clear enough.. Thanks, Michele --- arch/x86/include/asm/platform_sst_audio.h | 11 ++++++++--- sound/soc/intel/atom/sst/sst_acpi.c | 17 +++++++++++------ 2 files changed, 19 insertions(+), 9 deletions(-) diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h index 7249e6d..b10f6a7 100644 --- a/arch/x86/include/asm/platform_sst_audio.h +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -118,9 +118,6 @@ struct sst_res_info { unsigned int dram_size; unsigned int mbox_offset; unsigned int mbox_size; - unsigned int acpi_lpe_res_index; - unsigned int acpi_ddr_index; - unsigned int acpi_ipc_irq_index; }; struct sst_ipc_info { @@ -128,13 +125,21 @@ struct sst_ipc_info { unsigned int mbox_recv_off; }; +struct sst_acpi_info { + unsigned int lpe_res_index; + unsigned int ddr_index; + unsigned int ipc_irq_index; +}; + struct sst_platform_info { const struct sst_info *probe_data; const struct sst_ipc_info *ipc_info; + const struct sst_acpi_info *acpi_info; const struct sst_res_info *res_info; const struct sst_lib_dnld_info *lib_info; const char *platform; }; + int add_sst_platform_device(void); #endif diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index bb19b58..427e404 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -116,14 +116,18 @@ static const struct sst_res_info byt_rvp_res_info = { .dram_size = 0x28000, .mbox_offset = 0x144000, .mbox_size = 0x1000, - .acpi_lpe_res_index = 0, - .acpi_ddr_index = 2, - .acpi_ipc_irq_index = 5, +}; + +static const struct sst_acpi_info byt_rvp_acpi_info = { + .lpe_res_index = 0, + .ddr_index = 2, + .ipc_irq_index = 5, }; static struct sst_platform_info byt_rvp_platform_data = { .probe_data = &byt_fwparse_info, .ipc_info = &byt_ipc_info, + .acpi_info = &byt_rvp_acpi_info, .lib_info = &byt_lib_dnld_info, .res_info = &byt_rvp_res_info, .platform = "sst-mfld-platform", @@ -135,6 +139,7 @@ static struct sst_platform_info byt_rvp_platform_data = { static struct sst_platform_info chv_platform_data = { .probe_data = &byt_fwparse_info, .ipc_info = &byt_ipc_info, + .acpi_info = &byt_rvp_acpi_info, .lib_info = &byt_lib_dnld_info, .res_info = &byt_rvp_res_info, .platform = "sst-mfld-platform", @@ -148,7 +153,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) /* All ACPI resource request here */ /* Get Shim addr */ rsrc = platform_get_resource(pdev, IORESOURCE_MEM, - ctx->pdata->res_info->acpi_lpe_res_index); + ctx->pdata->acpi_info->lpe_res_index); if (!rsrc) { dev_err(ctx->dev, "Invalid SHIM base from IFWI"); return -EIO; @@ -202,7 +207,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) ctx->mailbox_add = ctx->info.mailbox_start; rsrc = platform_get_resource(pdev, IORESOURCE_MEM, - ctx->pdata->res_info->acpi_ddr_index); + ctx->pdata->acpi_info->ddr_index); if (!rsrc) { dev_err(ctx->dev, "Invalid DDR base from IFWI"); return -EIO; @@ -219,7 +224,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) /* Find the IRQ */ ctx->irq_num = platform_get_irq(pdev, - ctx->pdata->res_info->acpi_ipc_irq_index); + ctx->pdata->acpi_info->ipc_irq_index); return 0; } -- 2.5.0

1 0

[alsa-devel] [PATCH 0/3] USB-audio disconnection lockdep fixes
by Takashi Iwai 26 Aug '15

26 Aug '15

Hi, this is a series of patches for USB-audio to improve the lock handling at disconnection. Originally it was aimed to get rid of spurious lockdep warnings, but this ended up with the rewrite and improvement of locking code at disconnection and its checks in the mixer and PCM codes. Takashi === Takashi Iwai (3): ALSA: usb-audio: Avoid nested autoresume calls ALSA: usb-audio: Replace probing flag with active refcount ALSA: usb-audio: Handle normal and auto-suspend equally sound/usb/card.c | 108 ++++++++++++++++++++++------------------ sound/usb/endpoint.c | 10 ++-- sound/usb/mixer.c | 32 ++++-------- sound/usb/mixer_quirks.c | 126 ++++++++++++++++++++--------------------------- sound/usb/pcm.c | 32 ++++++------ sound/usb/proc.c | 4 +- sound/usb/usbaudio.h | 11 +++-- 7 files changed, 155 insertions(+), 168 deletions(-) -- 2.5.0

1 3

[alsa-devel] ALSA/usb-audio: snd_usb_autoresume possible recursive locking detected
by Alexnader Kuleshov 26 Aug '15

26 Aug '15

Hello all, Today I've update to 4.2.0-rc8+ and started to get following lines in the dmesg output: [ 13.884092] ============================================= [ 13.884092] [ INFO: possible recursive locking detected ] [ 13.884094] 4.2.0-rc8+ #61 Not tainted [ 13.884094] --------------------------------------------- [ 13.884095] pulseaudio/980 is trying to acquire lock: [ 13.884096] (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884103] but task is already holding lock: [ 13.884104] (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884107] other info that might help us debug this: [ 13.884108] Possible unsafe locking scenario: [ 13.884109] CPU0 [ 13.884110] ---- [ 13.884110] lock(&chip->shutdown_rwsem); [ 13.884111] lock(&chip->shutdown_rwsem); [ 13.884112] *** DEADLOCK *** [ 13.884113] May be due to missing lock nesting notation [ 13.884114] 2 locks held by pulseaudio/980: [ 13.884115] #0: (&pcm->open_mutex){+.+.+.}, at: [<ffffffffa02ac9bc>] snd_pcm_open+0xa9/0x1f8 [snd_pcm] [ 13.884120] #1: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884124] stack backtrace: [ 13.884125] CPU: 0 PID: 980 Comm: pulseaudio Not tainted 4.2.0-rc8+ #61 [ 13.884126] Hardware name: Gigabyte Technology Co., Ltd. Z97X-UD5H-BK/Z97X-UD5H-BK, BIOS F6 06/17/2014 [ 13.884127] 0000000000000000 000000003d4c66e6 ffff88040897b4d8 ffffffff815ba622 [ 13.884129] 0000000000000000 ffffffff82399320 ffff88040897b598 ffffffff810dd54e [ 13.884130] ffff88040ac486a8 0000000000000000 0000000000000001 ffffffff82399320 [ 13.884132] Call Trace: [ 13.884134] [<ffffffff815ba622>] dump_stack+0x4c/0x65 [ 13.884137] [<ffffffff810dd54e>] validate_chain.isra.9+0x75d/0xf59 [ 13.884139] [<ffffffff815c62b3>] ? _raw_spin_unlock_irq+0x28/0x34 [ 13.884140] [<ffffffff810e0964>] __lock_acquire+0x753/0xabf [ 13.884142] [<ffffffff81163819>] ? kfree+0xb8/0x10d [ 13.884143] [<ffffffff810e1131>] lock_acquire+0x7b/0x9c [ 13.884146] [<ffffffffa0355dac>] ? snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884147] [<ffffffff815c4896>] down_read+0x44/0x8d [ 13.884150] [<ffffffffa0355dac>] ? snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884152] [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884153] [<ffffffff815c37e5>] ? __mutex_unlock_slowpath+0x110/0x11b [ 13.884156] [<ffffffffa0359338>] snd_usb_mixer_set_ctl_value+0x9a/0x16d [snd_usb_audio] [ 13.884159] [<ffffffffa035957e>] snd_usb_set_cur_mix_value+0x4d/0x77 [snd_usb_audio] [ 13.884161] [<ffffffffa0359db9>] restore_mixer_value+0x74/0x87 [snd_usb_audio] [ 13.884164] [<ffffffffa035b581>] snd_usb_mixer_resume+0x64/0x70 [snd_usb_audio] [ 13.884166] [<ffffffffa035505c>] __usb_audio_resume+0x5c/0xd8 [snd_usb_audio] [ 13.884169] [<ffffffff81413ca4>] ? usb_runtime_suspend+0x63/0x63 [ 13.884171] [<ffffffffa03550e6>] usb_audio_reset_resume+0xe/0x10 [snd_usb_audio] [ 13.884172] [<ffffffff81412fde>] usb_resume_interface.isra.1+0x6c/0xbd [ 13.884174] [<ffffffff814130cf>] usb_resume_both+0xa0/0xc0 [ 13.884175] [<ffffffff81413cb9>] usb_runtime_resume+0x15/0x17 [ 13.884178] [<ffffffff813b92b0>] __rpm_callback+0x35/0x5d [ 13.884179] [<ffffffff81413ca4>] ? usb_runtime_suspend+0x63/0x63 [ 13.884180] [<ffffffff813b9330>] rpm_callback+0x58/0x77 [ 13.884182] [<ffffffff81413ca4>] ? usb_runtime_suspend+0x63/0x63 [ 13.884183] [<ffffffff813ba0db>] rpm_resume+0x370/0x412 [ 13.884184] [<ffffffff813ba06c>] rpm_resume+0x301/0x412 [ 13.884186] [<ffffffff813ba1e3>] __pm_runtime_resume+0x66/0x82 [ 13.884187] [<ffffffff81412e10>] usb_autopm_get_interface+0x1e/0x48 [ 13.884190] [<ffffffffa0355dcc>] snd_usb_autoresume+0x3d/0x52 [snd_usb_audio] [ 13.884191] [<ffffffff810d84b9>] ? __init_waitqueue_head+0x37/0x4b [ 13.884194] [<ffffffffa035e765>] snd_usb_pcm_open+0x181/0x3bb [snd_usb_audio] [ 13.884197] [<ffffffffa035e9ad>] snd_usb_capture_open+0xe/0x10 [snd_usb_audio] [ 13.884200] [<ffffffffa02ac8c6>] snd_pcm_open_substream+0x50/0x9d [snd_pcm] [ 13.884203] [<ffffffffa02ac9ce>] snd_pcm_open+0xbb/0x1f8 [snd_pcm] [ 13.884204] [<ffffffff810de87d>] ? trace_hardirqs_on+0xd/0xf [ 13.884206] [<ffffffff810c5550>] ? wake_up_q+0x53/0x53 [ 13.884209] [<ffffffffa02acb49>] snd_pcm_capture_open+0x3e/0x61 [snd_pcm] [ 13.884212] [<ffffffffa028232d>] snd_open+0x130/0x13f [snd] [ 13.884214] [<ffffffff811788c5>] chrdev_open+0x13b/0x175 [ 13.884215] [<ffffffff8117878a>] ? cdev_put+0x20/0x20 [ 13.884217] [<ffffffff81172f31>] do_dentry_open.isra.1+0x1d2/0x2c9 [ 13.884218] [<ffffffff8117d0e4>] ? __inode_permission+0x84/0x95 [ 13.884219] [<ffffffff81173bf6>] vfs_open+0x50/0x54 [ 13.884221] [<ffffffff811815e0>] path_openat+0xa90/0xd0f [ 13.884222] [<ffffffff810e0900>] ? __lock_acquire+0x6ef/0xabf [ 13.884223] [<ffffffff8118269e>] do_filp_open+0x48/0xac [ 13.884225] [<ffffffff815c6203>] ? _raw_spin_unlock+0x23/0x2e [ 13.884227] [<ffffffff8118f5b7>] ? __alloc_fd+0x144/0x156 [ 13.884228] [<ffffffff81173f3a>] do_sys_open+0x154/0x1f7 [ 13.884229] [<ffffffff81173ff6>] SyS_open+0x19/0x1b [ 13.884231] [<ffffffff815c6a57>] entry_SYSCALL_64_fastpath+0x12/0x6f Previous version of the kernel was 4.2.0-rc6+.

3 11

[alsa-devel] [PATCH] ALSA: hda - Remove superfluous pcm NULL check
by Takashi Iwai 26 Aug '15

26 Aug '15

The variable pcm can be never NULL since it was rewritten with list_for_each_entry(). Suggested-by: Markus Osterhoff <linux-kernel(a)k-raum.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c021911f88a..37f43a1b34ef 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3102,8 +3102,7 @@ static int add_std_chmaps(struct hda_codec *codec) struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (!pcm || !pcm->pcm || pcm->own_chmap || - !hinfo->substreams) + if (!pcm->pcm || pcm->own_chmap || !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, -- 2.5.0

1 0

[alsa-devel] [asoc:topic/rockchip 1/4] sound/soc/rockchip/rockchip_max98090.c:225:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 26 Aug '15

26 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rockchip head: f8ce20005d1694584448cd544be5fb32416a277c commit: 49bdb0440541ad1144ad08d5613f9a28bcd2a8dc [1/4] ASoC: rockchip: Add machine driver for max98090 codec coccinelle warnings: (new ones prefixed by >>) >> sound/soc/rockchip/rockchip_max98090.c:225:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [asoc:topic/rockchip 2/4] sound/soc/rockchip/rockchip_rt5645.c:214:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 26 Aug '15

26 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rockchip head: f8ce20005d1694584448cd544be5fb32416a277c commit: 86059653ea7ca7b30ed25d6bec5807ba59a4f2e6 [2/4] ASoC: rockchip: Add machine driver for rt5645/rt5650 codec coccinelle warnings: (new ones prefixed by >>) >> sound/soc/rockchip/rockchip_rt5645.c:214:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH v4] ASoC: ics43432: Add codec driver for InvenSense ICS-43432 (fwd)
by Ricard Wanderlof 26 Aug '15

26 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Tested on a setup with a single ICS-43432, although the device itself supports stereo operation using a hardware pin controlling left vs. right channel output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V4: Updated commit message that the driver has been tested with real hardware, after off-list comment on V3. V3: Fixed MODULE_LICENSE string. V2: Update after comments from Lars-Peter: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. I've left the checks in dai_set_fmt, as (IIUC) there currently doesn't seem to be a way for the ALSA framework to handle format constraints using the codec configuration struct. .../devicetree/bindings/sound/ics43432.txt | 17 ++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 101 ++++++++++++++++++++ 5 files changed, 125 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..fe0af4a --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,101 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPLv2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 2

[alsa-devel] [PATCH 1/5] ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning
by Oder Chiou 26 Aug '15

26 Aug '15

Add i2c shutdown function to prevent the pop sound of the headphone while the system is rebooting or shutdowning Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e5fef8f..4189d5a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3490,13 +3490,23 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) return 0; } +static void rt5645_i2c_shutdown(struct i2c_client *i2c) +{ + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, 0x0020, 0x0020); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, 0x1000, 0x1000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, 0x0000); +} + static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, - .remove = rt5645_i2c_remove, + .remove = rt5645_i2c_remove, + .shutdown = rt5645_i2c_shutdown, .id_table = rt5645_i2c_id, }; module_i2c_driver(rt5645_i2c_driver); -- 1.8.1.1.439.g50a6b54

2 11

[alsa-devel] [PATCH v2] ASoC: rockchip: fix a misjudgement by return
by Xing Zheng 26 Aug '15

26 Aug '15

Being careless, judge the return value of snd_soc_card_jack_new is opposite, so it should be fixed. --- Changes in v2: Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> Reviewed-by: Dylan Reid <dgreid(a)chromium.org> sound/soc/rockchip/rockchip_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 3c6bb1e..adfe98c 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -118,7 +118,7 @@ static int rk_init(struct snd_soc_pcm_runtime *runtime) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &headset_jack, NULL, 0); - if (!ret) { + if (ret) { dev_err(card->dev, "New Headset Jack failed! (%d)\n", ret); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v3 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 26 Aug '15

26 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..aabebcb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,7 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

5 11

[alsa-devel] [PATCH] ALSA: hda - Fix widget sysfs tree corruption after refresh
by Takashi Iwai 26 Aug '15

26 Aug '15

When snd_hdac_refresh_widget_sysfs() is called before the first hda_widget_sysfs_init(), the next call overrides and eventually fails. This results in unexpected Oops, something like: BUG: unable to handle kernel NULL pointer dereference at 00000000000000c8 IP: [<ffffffff8180e2a3>] hdmi_chmap_ctl_info+0x23/0x40 The fix is to add a check of the existing sysfs tree. Also, for more safety, this patch adds the checks of device_is_registered() in snd-hdac_refresh_wdiget_sysfs(), too. Fixes: fa4f18b4f402 ('ALSA: hda - Refresh widgets sysfs at probing Haswell+ HDMI codecs') Bugizlla: https://bugzilla.kernel.org/show_bug.cgi?id=103431 Reported-by: Andreas Reis <andreas.reis(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_device.c | 14 ++++++++------ sound/hda/hdac_sysfs.c | 3 +++ 2 files changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index aa6d6cec2380..db96042a497f 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -384,18 +384,20 @@ int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec) { int ret; - hda_widget_sysfs_exit(codec); + if (device_is_registered(&codec->dev)) + hda_widget_sysfs_exit(codec); ret = snd_hdac_refresh_widgets(codec); if (ret) { dev_err(&codec->dev, "failed to refresh widget: %d\n", ret); return ret; } - ret = hda_widget_sysfs_init(codec); - if (ret) { - dev_err(&codec->dev, "failed to init sysfs: %d\n", ret); - return ret; + if (device_is_registered(&codec->dev)) { + ret = hda_widget_sysfs_init(codec); + if (ret) { + dev_err(&codec->dev, "failed to init sysfs: %d\n", ret); + return ret; + } } - return ret; } EXPORT_SYMBOL_GPL(snd_hdac_refresh_widget_sysfs); diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 089b35f6f108..c71142dea98a 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -390,6 +390,9 @@ int hda_widget_sysfs_init(struct hdac_device *codec) { int err; + if (codec->widgets) + return 0; /* already created */ + err = widget_tree_create(codec); if (err < 0) { widget_tree_free(codec); -- 2.5.0

1 0

[alsa-devel] [asoc:topic/rt5677 6/6] sound/soc/codecs/rt5677-spi.c:130:3: warning: format '%zu' expects argument of type 'size_t', but argument 4 has type 'unsigned int'
by kbuild test robot 25 Aug '15

25 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5677 head: 7d4d443eb4386d6dbd420fa96303dd8fbc1eefc8 commit: 7d4d443eb4386d6dbd420fa96303dd8fbc1eefc8 [6/6] ASoC: rt5677: Allow arbitrary block read/write via SPI config: mn10300-allmodconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 7d4d443eb4386d6dbd420fa96303dd8fbc1eefc8 # save the attached .config to linux build tree make.cross ARCH=mn10300 All warnings (new ones prefixed by >>): sound/soc/codecs/rt5677-spi.c: In function 'rt5677_spi_read': >> sound/soc/codecs/rt5677-spi.c:130:3: warning: format '%zu' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat=] dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); ^ sound/soc/codecs/rt5677-spi.c: In function 'rt5677_spi_write': sound/soc/codecs/rt5677-spi.c:184:3: warning: format '%zu' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat=] dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); ^ vim +130 sound/soc/codecs/rt5677-spi.c 114 int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) 115 { 116 u32 offset; 117 int status = 0; 118 struct spi_transfer t[2]; 119 struct spi_message m; 120 /* +4 bytes is for the DummyPhase following the AddressPhase */ 121 u8 header[RT5677_SPI_HEADER + 4]; 122 u8 body[RT5677_SPI_BURST_LEN]; 123 u8 spi_cmd; 124 u8 *cb = rxbuf; 125 126 if (!g_spi) 127 return -ENODEV; 128 129 if ((addr & 1) || (len & 1)) { > 130 dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); 131 return -EACCES; 132 } 133 134 memset(t, 0, sizeof(t)); 135 t[0].tx_buf = header; 136 t[0].len = sizeof(header); 137 t[0].speed_hz = RT5677_SPI_FREQ; 138 t[1].rx_buf = body; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH V5 3/3] ASoC: AMD: add AMD ASoC ACP-I2S driver
by Alex Deucher 25 Aug '15

25 Aug '15

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> ACP IP block consists of dedicated DMA and I2S blocks. The PCM driver provides the DMA and CPU DAI components to ALSA core. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Reviewed-by: Murali Krishna Vemuri <murali-krishna.vemuri(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- v2: squash in Kconfig fix v3: squash additional commits, convert to mfd, drop rt286 changes v4: squash in naming fixes v5: move patch versioning out of commit message The module licensing follows the model used in the drm drivers. sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 5 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 676 ++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 686 insertions(+) create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 2ae9619..1b9ce31 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/adi/Kconfig" +source "sound/soc/amd/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index e189903..a6cbb99 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,7 @@ obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += adi/ +obj-$(CONFIG_SND_SOC) += amd/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += bcm/ diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig new file mode 100644 index 0000000..ce39979 --- /dev/null +++ b/sound/soc/amd/Kconfig @@ -0,0 +1,5 @@ +config SND_SOC_AMD_ACP + tristate "AMD Audio Coprocessor support" + depends on MFD_CORE + help + This option enables ACP support (DMA,I2S) on AMD platforms. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile new file mode 100644 index 0000000..1a66ec0 --- /dev/null +++ b/sound/soc/amd/Makefile @@ -0,0 +1,3 @@ +snd-soc-acp-pcm-objs := acp-pcm-dma.o + +obj-$(CONFIG_SND_SOC_AMD_ACP) += snd-soc-acp-pcm.o diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c new file mode 100644 index 0000000..18cc60c --- /dev/null +++ b/sound/soc/amd/acp-pcm-dma.c @@ -0,0 +1,676 @@ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2014-2015 Advanced Micro Devices, Inc. + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR + * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, + * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + * + */ + +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/pci.h> +#include <linux/pm_runtime.h> +#include <linux/mfd/amd_acp.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 2 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 1024 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 2 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 1024 + +#define NUM_DSCRS_PER_CHANNEL 2 + +#define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) +#define MIN_BUFFER MAX_BUFFER + +#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */ +#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ +#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ +#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ + + +static const struct snd_pcm_hardware acp_pcm_hardware_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + /* formats,rates,channels based on i2s doc. */ + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +static const struct snd_pcm_hardware acp_pcm_hardware_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + /* formats,rates,channels based on i2s doc. */ + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + +struct audio_drv_data { + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + struct amd_acp_device *acp_dev; + struct acp_irq_prv *iprv; +}; + +static const struct snd_soc_component_driver dw_i2s_component = { + .name = "dw-i2s", +}; + +static void acp_pcm_period_elapsed(struct device *dev, u16 play_intr, + u16 capture_intr) +{ + struct snd_pcm_substream *substream; + struct audio_drv_data *irq_data = dev_get_drvdata(dev); + + /* Inform ALSA about the period elapsed (one out of two periods) */ + if (play_intr) + substream = irq_data->play_stream; + else + substream = irq_data->capture_stream; + + if (substream->runtime && snd_pcm_running(substream)) + snd_pcm_period_elapsed(substream); +} + +static int acp_dma_open(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_drv_data *intr_data = dev_get_drvdata(prtd->platform->dev); + + struct audio_substream_data *adata = + kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); + if (adata == NULL) + return -ENOMEM; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp_pcm_hardware_playback; + else + runtime->hw = acp_pcm_hardware_capture; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(prtd->platform->dev, "set integer constraint failed\n"); + return ret; + } + + adata->acp_dev = intr_data->acp_dev; + runtime->private_data = adata; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + intr_data->play_stream = substream; + else + intr_data->capture_stream = substream; + + return 0; +} + +static int acp_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int status; + uint64_t size; + struct snd_dma_buffer *dma_buffer; + struct page *pg; + u16 num_of_pages; + struct snd_pcm_runtime *runtime; + struct audio_substream_data *rtd; + struct amd_acp_device *acp_dev; + + dma_buffer = &substream->dma_buffer; + + runtime = substream->runtime; + rtd = runtime->private_data; + + if (WARN_ON(!rtd)) + return -EINVAL; + acp_dev = rtd->acp_dev; + + size = params_buffer_bytes(params); + status = snd_pcm_lib_malloc_pages(substream, size); + if (status < 0) + return status; + + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + pg = virt_to_page(substream->dma_buffer.area); + + if (NULL != pg) { + /* Save for runtime private data */ + rtd->pg = pg; + rtd->order = get_order(size); + + /* Let ACP know the Allocated memory */ + num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; + + /* Fill the page table entries in ACP SRAM */ + rtd->pg = pg; + rtd->size = size; + rtd->num_of_pages = num_of_pages; + rtd->direction = substream->stream; + + acp_dev->config_dma(acp_dev, rtd); + status = 0; + } else { + status = -ENOMEM; + } + return status; +} + +static int acp_dma_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) +{ + u32 pos = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + pos = acp_dev->update_dma_pointer(acp_dev, substream->stream, + frames_to_bytes(runtime, runtime->period_size)); + return bytes_to_frames(runtime, pos); + +} + +static int acp_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return snd_pcm_lib_default_mmap(substream, vma); +} + +static int acp_dma_prepare(struct snd_pcm_substream *substream) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + acp_dev->config_dma_channel(acp_dev, SYSRAM_TO_ACP_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH12, + NUM_DSCRS_PER_CHANNEL, 0); + acp_dev->config_dma_channel(acp_dev, ACP_TO_I2S_DMA_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH13, + NUM_DSCRS_PER_CHANNEL, 0); + /* Fill ACP SRAM (2 periods) with zeros from System RAM + * which is zero-ed in hw_params */ + ret = acp_dev->dma_start(rtd->acp_dev, + SYSRAM_TO_ACP_CH_NUM, false); + if (ret < 0) + ret = -EFAULT; + + /* Now configure DMA to transfer only first half of System RAM + * buffer before playback is triggered. This will overwrite + * zero-ed second half of SRAM buffer */ + acp_dev->config_dma_channel(acp_dev, SYSRAM_TO_ACP_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH12, + 1, 0); + } else { + acp_dev->config_dma_channel(acp_dev, ACP_TO_SYSRAM_CH_NUM, + CAPTURE_START_DMA_DESCR_CH14, + NUM_DSCRS_PER_CHANNEL, 0); + acp_dev->config_dma_channel(acp_dev, I2S_TO_ACP_DMA_CH_NUM, + CAPTURE_START_DMA_DESCR_CH15, + NUM_DSCRS_PER_CHANNEL, 0); + } + return 0; +} + +static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + int ret; + + if (!rtd) + return -EINVAL; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dev->dma_start(rtd->acp_dev, + SYSRAM_TO_ACP_CH_NUM, false); + if (ret < 0) + return -EFAULT; + acp_dev->prebuffer_audio(rtd->acp_dev); + + ret = acp_dev->dma_start(acp_dev, + ACP_TO_I2S_DMA_CH_NUM, true); + } else { + ret = acp_dev->dma_start(acp_dev, + I2S_TO_ACP_DMA_CH_NUM, true); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dev->dma_stop(acp_dev, SYSRAM_TO_ACP_CH_NUM); + if (0 == ret) + ret = acp_dev->dma_stop(acp_dev, + ACP_TO_I2S_DMA_CH_NUM); + } else { + ret = acp_dev->dma_stop(acp_dev, I2S_TO_ACP_DMA_CH_NUM); + if (0 == ret) + ret = acp_dev->dma_stop(acp_dev, + ACP_TO_SYSRAM_CH_NUM); + } + break; + default: + ret = -EINVAL; + + } + return ret; +} + +static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_pcm *pcm; + + pcm = rtd->pcm; + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + NULL, MIN_BUFFER, MAX_BUFFER); + return ret; +} + +static int acp_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + + kfree(rtd); + + pm_runtime_mark_last_busy(prtd->platform->dev); + return 0; +} + +static int acp_dai_i2s_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + struct device *dev = prtd->platform->dev; + u32 chan_nr; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + rtd->xfer_resolution = 0x02; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + rtd->xfer_resolution = 0x04; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + rtd->xfer_resolution = 0x05; + break; + + default: + dev_err(dev, "unsuppted PCM fmt : %d\n", params_format(params)); + return -EINVAL; + } + + chan_nr = params_channels(params); + + switch (chan_nr) { + case EIGHT_CHANNEL_SUPPORT: + rtd->ch_reg = 3; + break; + case SIX_CHANNEL_SUPPORT: + rtd->ch_reg = 2; + break; + case FOUR_CHANNEL_SUPPORT: + rtd->ch_reg = 1; + break; + case TWO_CHANNEL_SUPPORT: + rtd->ch_reg = 0; + break; + default: + dev_err(dev, "channel not supported : %d\n", chan_nr); + return -EINVAL; + } + + rtd->direction = substream->stream; + + acp_dev->config_i2s(acp_dev, rtd); + + return 0; +} + +static int acp_dai_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + acp_dev->i2s_start(acp_dev, substream->stream); + ret = 0; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + acp_dev->i2s_stop(acp_dev, substream->stream); + ret = 0; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int acp_dai_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + acp_dev->i2s_reset(acp_dev, substream->stream); + return 0; +} + +static struct snd_pcm_ops acp_dma_ops = { + .open = acp_dma_open, + .close = acp_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = acp_dma_hw_params, + .hw_free = acp_dma_hw_free, + .trigger = acp_dma_trigger, + .pointer = acp_dma_pointer, + .mmap = acp_dma_mmap, + .prepare = acp_dma_prepare, +}; + +static struct snd_soc_dai_ops acp_dai_i2s_ops = { + .prepare = acp_dai_i2s_prepare, + .hw_params = acp_dai_i2s_hwparams, + .trigger = acp_dai_i2s_trigger, +}; + +static struct snd_soc_dai_driver i2s_dai_driver = { + .playback = { + .stream_name = "I2S Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .stream_name = "I2S Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp_dai_i2s_ops, +}; + +static struct snd_soc_platform_driver acp_asoc_platform = { + .ops = &acp_dma_ops, + .pcm_new = acp_dma_new, +}; + +static int acp_alsa_register(struct device *dev, struct amd_acp_device *acp_dev, + struct platform_device *pdev) +{ + int status; + + status = snd_soc_register_platform(dev, &acp_asoc_platform); + if (0 != status) { + dev_err(dev, "Unable to register ALSA platform device\n"); + goto exit_platform; + } else { + status = snd_soc_register_component(dev, + &dw_i2s_component, + &i2s_dai_driver, 1); + + if (0 != status) { + dev_err(dev, "Unable to register i2s dai\n"); + goto exit_dai; + } else { + dev_info(dev, "ACP device registered with ALSA\n"); + return status; + } + } + +exit_dai: + snd_soc_unregister_platform(dev); +exit_platform: + acp_dev->fini(acp_dev); + return status; +} + +static int acp_audio_probe(struct platform_device *pdev) +{ + int status; + struct audio_drv_data *audio_drv_data; + struct amd_acp_device *acp_dev = dev_get_platdata(&pdev->dev); + + audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), + GFP_KERNEL); + if (audio_drv_data == NULL) + return -ENOMEM; + + audio_drv_data->iprv = devm_kzalloc(&pdev->dev, + sizeof(struct acp_irq_prv), + GFP_KERNEL); + if (audio_drv_data->iprv == NULL) + return -ENOMEM; + + /* The following members gets populated in device 'open' + * function. Till then interrupts are disabled in 'acp_hw_init' + * and device doesn't generate any interrupts. + */ + + audio_drv_data->play_stream = NULL; + audio_drv_data->capture_stream = NULL; + audio_drv_data->acp_dev = acp_dev; + + audio_drv_data->iprv->dev = &pdev->dev; + audio_drv_data->iprv->acp_dev = acp_dev; + audio_drv_data->iprv->set_elapsed = acp_pcm_period_elapsed; + + dev_set_drvdata(&pdev->dev, audio_drv_data); + + /* Initialize the ACP */ + status = acp_dev->init(acp_dev, audio_drv_data->iprv); + + if (0 == status) + status = acp_alsa_register(&pdev->dev, acp_dev, pdev); + else + dev_err(&pdev->dev, "ACP initialization Failed\n"); + + pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); + + return status; +} + +static int acp_audio_remove(struct platform_device *pdev) +{ + struct amd_acp_device *acp_dev = dev_get_platdata(&pdev->dev); + + snd_soc_unregister_component(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); + + acp_dev->fini(acp_dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static int acp_pcm_suspend(struct device *dev) +{ + bool pm_rts; + struct audio_drv_data *adata = dev_get_drvdata(dev); + + pm_rts = pm_runtime_status_suspended(dev); + if (pm_rts == false) + adata->acp_dev->fini(adata->acp_dev); + + return 0; +} + +static int acp_pcm_resume(struct device *dev) +{ + bool pm_rts; + struct snd_pcm_substream *pstream, *cstream; + struct snd_pcm_runtime *prtd, *crtd; + struct audio_substream_data *rtd; + struct audio_drv_data *adata = dev_get_drvdata(dev); + + pm_rts = pm_runtime_status_suspended(dev); + if (pm_rts == true) { + /* Resumed from system wide suspend and there is + * no pending audio activity to resume. */ + pm_runtime_disable(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + + goto out; + } + + pstream = adata->play_stream; + prtd = pstream ? pstream->runtime : NULL; + if (prtd != NULL) { + /* Resume playback stream from a suspended state */ + rtd = prtd->private_data; + + adata->acp_dev->config_dma(adata->acp_dev, rtd); + adata->acp_dev->config_i2s(adata->acp_dev, rtd); + } + + cstream = adata->capture_stream; + crtd = cstream ? cstream->runtime : NULL; + if (crtd != NULL) { + /* Resume capture stream from a suspended state */ + rtd = crtd->private_data; + + adata->acp_dev->config_dma(adata->acp_dev, rtd); + adata->acp_dev->config_i2s(adata->acp_dev, rtd); + } +out: + return 0; +} + +static int acp_pcm_runtime_suspend(struct device *dev) +{ + struct audio_drv_data *adata = dev_get_drvdata(dev); + + adata->acp_dev->acp_suspend(adata->acp_dev); + return 0; +} + +static int acp_pcm_runtime_resume(struct device *dev) +{ + struct audio_drv_data *adata = dev_get_drvdata(dev); + + adata->acp_dev->acp_resume(adata->acp_dev); + return 0; +} + +static const struct dev_pm_ops acp_pm_ops = { + .suspend = acp_pcm_suspend, + .resume = acp_pcm_resume, + .runtime_suspend = acp_pcm_runtime_suspend, + .runtime_resume = acp_pcm_runtime_resume, +}; + +static struct platform_driver acp_dma_driver = { + .probe = acp_audio_probe, + .remove = acp_audio_remove, + .driver = { + .name = "acp-i2s-audio", + .pm = &acp_pm_ops, + }, +}; + +module_platform_driver(acp_dma_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP PCM Driver"); +MODULE_LICENSE("GPL and additional rights"); +MODULE_ALIAS("platform:acp-i2s-audio"); -- 1.8.3.1

3 8

[alsa-devel] [PATCH] ASoC: rt5645: Add struct dmi_system_id "Google Celes" for chrome platform
by Oder Chiou 25 Aug '15

25 Aug '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4917057..bae1c5f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3198,6 +3198,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Strago"), }, }, + { + .ident = "Google Celes", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: arizona: Poll for FLL clock OK rather than use interrupts
by Charles Keepax 25 Aug '15

25 Aug '15

The extcon driver takes the DAPM mutex from within the interrupt thread in several places, which makes it possible to get into a situation where the interrupt thread is blocked waiting on the DAPM mutex whilst a DAPM sequence is running which is attempting to configure the FLL. In this case the FLL completion can't be completed as as the IRQ handler is ONE_SHOT, which cause the FLL lock to use the full time out (250mS) and report that the process timed out. It is not really practical to make the extcon driver not take the DAPM mutex from within the interrupt thread, at least not without extensive modification. So this patch fixes the issue by switching the wait for the FLL lock to polling. A few fast polls are done first as the FLL should lock quickly for a good quality reference clock, (indeed it hits on the first poll on my system) and it will poll every 20mS after that until it times out. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 47 +++++++++++++++++-------------------------- sound/soc/codecs/arizona.h | 1 - 2 files changed, 19 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 42b2021..8a2221a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1756,17 +1756,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id) } EXPORT_SYMBOL_GPL(arizona_init_dai); -static irqreturn_t arizona_fll_clock_ok(int irq, void *data) -{ - struct arizona_fll *fll = data; - - arizona_fll_dbg(fll, "clock OK\n"); - - complete(&fll->ok); - - return IRQ_HANDLED; -} - static struct { unsigned int min; unsigned int max; @@ -2048,10 +2037,11 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll) static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; - unsigned long time_left; bool use_sync = false; int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; + int i; + unsigned int val; if (already_enabled < 0) return already_enabled; @@ -2110,9 +2100,6 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (!already_enabled) pm_runtime_get(arizona->dev); - /* Clear any pending completions */ - try_wait_for_completion(&fll->ok); - regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) @@ -2124,10 +2111,24 @@ static int arizona_enable_fll(struct arizona_fll *fll) regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); - time_left = wait_for_completion_timeout(&fll->ok, - msecs_to_jiffies(250)); - if (time_left == 0) + arizona_fll_dbg(fll, "Waiting for FLL lock...\n"); + val = 0; + for (i = 0; i < 15; i++) { + if (i < 5) + usleep_range(200, 400); + else + msleep(20); + + regmap_read(arizona->regmap, + ARIZONA_INTERRUPT_RAW_STATUS_5, + &val); + if (val & (ARIZONA_FLL1_CLOCK_OK_STS << (fll->id - 1))) + break; + } + if (i == 15) arizona_fll_warn(fll, "Timed out waiting for lock\n"); + else + arizona_fll_dbg(fll, "FLL locked (%d polls)\n", i); return 0; } @@ -2212,11 +2213,8 @@ EXPORT_SYMBOL_GPL(arizona_set_fll); int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll) { - int ret; unsigned int val; - init_completion(&fll->ok); - fll->id = id; fll->base = base; fll->arizona = arizona; @@ -2238,13 +2236,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); - ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, - arizona_fll_clock_ok, fll); - if (ret != 0) { - dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", - id, ret); - } - regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 13598b0..ada0a41 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -261,7 +261,6 @@ struct arizona_fll { int id; unsigned int base; unsigned int vco_mult; - struct completion ok; unsigned int fout; int sync_src; -- 1.7.2.5

2 1

[alsa-devel] [PATCH v1 0/1] A misjudgement should be fixed
by Xing Zheng 25 Aug '15

25 Aug '15

Hi, Being careless, judge the return value of snd_soc_card_jack_new is opposite, so it should be fixed. Thanks. Changes in v1: Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> Reviewed-by: Dylan Reid <dgreid(a)chromium.org> Xing Zheng (1): ASoC: rockchip: fix a misjudgement by return sound/soc/rockchip/rockchip_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: rt5677: Allow arbitrary block read/write via SPI
by Ben Zhang 25 Aug '15

25 Aug '15

Added rt5677_spi_read() and refactored rt5677_spi_write() so that an arbitrary block in the DSP address space can be read/written via SPI. For example, this allows us to load an ELF DSP firmware with sparse sections, and stream audio samples from DSP ring buffer. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 233 +++++++++++++++++++++++++++++++----------- sound/soc/codecs/rt5677-spi.h | 8 +- sound/soc/codecs/rt5677.c | 4 +- 3 files changed, 178 insertions(+), 67 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index ef6348c..3505aaf 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -31,84 +31,197 @@ #include "rt5677-spi.h" +#define RT5677_SPI_BURST_LEN 240 +#define RT5677_SPI_HEADER 5 +#define RT5677_SPI_FREQ 6000000 + +/* The AddressPhase and DataPhase of SPI commands are MSB first on the wire. + * DataPhase word size of 16-bit commands is 2 bytes. + * DataPhase word size of 32-bit commands is 4 bytes. + * DataPhase word size of burst commands is 8 bytes. + * The DSP CPU is little-endian. + */ +#define RT5677_SPI_WRITE_BURST 0x5 +#define RT5677_SPI_READ_BURST 0x4 +#define RT5677_SPI_WRITE_32 0x3 +#define RT5677_SPI_READ_32 0x2 +#define RT5677_SPI_WRITE_16 0x1 +#define RT5677_SPI_READ_16 0x0 + static struct spi_device *g_spi; +static DEFINE_MUTEX(spi_mutex); -/** - * rt5677_spi_write - Write data to SPI. - * @txbuf: Data Buffer for writing. - * @len: Data length. +/* Select a suitable transfer command for the next transfer to ensure + * the transfer address is always naturally aligned while minimizing + * the total number of transfers required. + * + * 3 transfer commands are available: + * RT5677_SPI_READ/WRITE_16: Transfer 2 bytes + * RT5677_SPI_READ/WRITE_32: Transfer 4 bytes + * RT5677_SPI_READ/WRITE_BURST: Transfer any multiples of 8 bytes + * + * For example, reading 260 bytes at 0x60030002 uses the following commands: + * 0x60030002 RT5677_SPI_READ_16 2 bytes + * 0x60030004 RT5677_SPI_READ_32 4 bytes + * 0x60030008 RT5677_SPI_READ_BURST 240 bytes + * 0x600300F8 RT5677_SPI_READ_BURST 8 bytes + * 0x60030100 RT5677_SPI_READ_32 4 bytes + * 0x60030104 RT5677_SPI_READ_16 2 bytes * + * Input: + * @read: true for read commands; false for write commands + * @align: alignment of the next transfer address + * @remain: number of bytes remaining to transfer * - * Returns true for success. + * Output: + * @len: number of bytes to transfer with the selected command + * Returns the selected command */ -int rt5677_spi_write(u8 *txbuf, size_t len) +static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len) { - int status; - - status = spi_write(g_spi, txbuf, len); - - if (status) - dev_err(&g_spi->dev, "rt5677_spi_write error %d\n", status); - - return status; + u8 cmd; + + if (align == 2 || align == 6 || remain == 2) { + cmd = RT5677_SPI_READ_16; + *len = 2; + } else if (align == 4 || remain <= 6) { + cmd = RT5677_SPI_READ_32; + *len = 4; + } else { + cmd = RT5677_SPI_READ_BURST; + *len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN); + } + return read ? cmd : cmd + 1; } -EXPORT_SYMBOL_GPL(rt5677_spi_write); -/** - * rt5677_spi_burst_write - Write data to SPI by rt5677 dsp memory address. - * @addr: Start address. - * @txbuf: Data Buffer for writng. - * @len: Data length, it must be a multiple of 8. - * - * - * Returns true for success. +/* Copy dstlen bytes from src to dst, while reversing byte order for each word. + * If srclen < dstlen, zeros are padded. */ -int rt5677_spi_burst_write(u32 addr, const struct firmware *fw) +static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) { - u8 spi_cmd = RT5677_SPI_CMD_BURST_WRITE; - u8 *write_buf; - unsigned int i, end, offset = 0; - - write_buf = kmalloc(RT5677_SPI_BUF_LEN + 6, GFP_KERNEL); - - if (write_buf == NULL) - return -ENOMEM; - - while (offset < fw->size) { - if (offset + RT5677_SPI_BUF_LEN <= fw->size) - end = RT5677_SPI_BUF_LEN; - else - end = fw->size % RT5677_SPI_BUF_LEN; - - write_buf[0] = spi_cmd; - write_buf[1] = ((addr + offset) & 0xff000000) >> 24; - write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; - write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; - write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; - - for (i = 0; i < end; i += 8) { - write_buf[i + 12] = fw->data[offset + i + 0]; - write_buf[i + 11] = fw->data[offset + i + 1]; - write_buf[i + 10] = fw->data[offset + i + 2]; - write_buf[i + 9] = fw->data[offset + i + 3]; - write_buf[i + 8] = fw->data[offset + i + 4]; - write_buf[i + 7] = fw->data[offset + i + 5]; - write_buf[i + 6] = fw->data[offset + i + 6]; - write_buf[i + 5] = fw->data[offset + i + 7]; + u32 w, i, si; + u32 word_size = min_t(u32, dstlen, 8); + + for (w = 0; w < dstlen; w += word_size) { + for (i = 0; i < word_size; i++) { + si = w + word_size - i - 1; + dst[w + i] = si < srclen ? src[si] : 0; } + } +} - write_buf[end + 5] = spi_cmd; +/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */ +int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) +{ + u32 offset; + int status = 0; + struct spi_transfer t[2]; + struct spi_message m; + /* +4 bytes is for the DummyPhase following the AddressPhase */ + u8 header[RT5677_SPI_HEADER + 4]; + u8 body[RT5677_SPI_BURST_LEN]; + u8 spi_cmd; + u8 *cb = rxbuf; + + if (!g_spi) + return -ENODEV; + + if ((addr & 1) || (len & 1)) { + dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); + return -EACCES; + } - rt5677_spi_write(write_buf, end + 6); + memset(t, 0, sizeof(t)); + t[0].tx_buf = header; + t[0].len = sizeof(header); + t[0].speed_hz = RT5677_SPI_FREQ; + t[1].rx_buf = body; + t[1].speed_hz = RT5677_SPI_FREQ; + spi_message_init_with_transfers(&m, t, ARRAY_SIZE(t)); + + for (offset = 0; offset < len; offset += t[1].len) { + spi_cmd = rt5677_spi_select_cmd(true, (addr + offset) & 7, + len - offset, &t[1].len); + + /* Construct SPI message header */ + header[0] = spi_cmd; + header[1] = ((addr + offset) & 0xff000000) >> 24; + header[2] = ((addr + offset) & 0x00ff0000) >> 16; + header[3] = ((addr + offset) & 0x0000ff00) >> 8; + header[4] = ((addr + offset) & 0x000000ff) >> 0; + + mutex_lock(&spi_mutex); + status |= spi_sync(g_spi, &m); + mutex_unlock(&spi_mutex); + + /* Copy data back to caller buffer */ + rt5677_spi_reverse(cb + offset, t[1].len, body, t[1].len); + } + return status; +} +EXPORT_SYMBOL_GPL(rt5677_spi_read); - offset += RT5677_SPI_BUF_LEN; +/* Write DSP address space using SPI. addr has to be 2-byte aligned. + * If len is not 2-byte aligned, an extra byte of zero is written at the end + * as padding. + */ +int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) +{ + u32 offset, len_with_pad = len; + int status = 0; + struct spi_transfer t; + struct spi_message m; + /* +1 byte is for the DummyPhase following the DataPhase */ + u8 buf[RT5677_SPI_HEADER + RT5677_SPI_BURST_LEN + 1]; + u8 *body = buf + RT5677_SPI_HEADER; + u8 spi_cmd; + const u8 *cb = txbuf; + + if (!g_spi) + return -ENODEV; + + if (addr & 1) { + dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); + return -EACCES; } - kfree(write_buf); + if (len & 1) + len_with_pad = len + 1; + + memset(&t, 0, sizeof(t)); + t.tx_buf = buf; + t.speed_hz = RT5677_SPI_FREQ; + spi_message_init_with_transfers(&m, &t, 1); + + for (offset = 0; offset < len_with_pad;) { + spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7, + len_with_pad - offset, &t.len); + + /* Construct SPI message header */ + buf[0] = spi_cmd; + buf[1] = ((addr + offset) & 0xff000000) >> 24; + buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + /* Fetch data from caller buffer */ + rt5677_spi_reverse(body, t.len, cb + offset, len - offset); + offset += t.len; + t.len += RT5677_SPI_HEADER + 1; + + mutex_lock(&spi_mutex); + status |= spi_sync(g_spi, &m); + mutex_unlock(&spi_mutex); + } + return status; +} +EXPORT_SYMBOL_GPL(rt5677_spi_write); - return 0; +int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw) +{ + return rt5677_spi_write(addr, fw->data, fw->size); } -EXPORT_SYMBOL_GPL(rt5677_spi_burst_write); +EXPORT_SYMBOL_GPL(rt5677_spi_write_firmware); static int rt5677_spi_probe(struct spi_device *spi) { diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h index ec41b2b..662db16 100644 --- a/sound/soc/codecs/rt5677-spi.h +++ b/sound/soc/codecs/rt5677-spi.h @@ -12,10 +12,8 @@ #ifndef __RT5677_SPI_H__ #define __RT5677_SPI_H__ -#define RT5677_SPI_BUF_LEN 240 -#define RT5677_SPI_CMD_BURST_WRITE 0x05 - -int rt5677_spi_write(u8 *txbuf, size_t len); -int rt5677_spi_burst_write(u32 addr, const struct firmware *fw); +int rt5677_spi_read(u32 addr, void *rxbuf, size_t len); +int rt5677_spi_write(u32 addr, const void *txbuf, size_t len); +int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw); #endif /* __RT5677_SPI_H__ */ diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 3f890a6..d916d0c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -745,14 +745,14 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, codec->dev); if (ret == 0) { - rt5677_spi_burst_write(0x50000000, rt5677->fw1); + rt5677_spi_write_firmware(0x50000000, rt5677->fw1); release_firmware(rt5677->fw1); } ret = request_firmware(&rt5677->fw2, RT5677_FIRMWARE2, codec->dev); if (ret == 0) { - rt5677_spi_burst_write(0x60000000, rt5677->fw2); + rt5677_spi_write_firmware(0x60000000, rt5677->fw2); release_firmware(rt5677->fw2); } -- 2.5.0.457.gab17608

3 2

Re: [alsa-devel] [RESEND PATCH v1 0/1] A misjudgement should be fixed
by Mark Brown 25 Aug '15

25 Aug '15

On Tue, Aug 25, 2015 at 03:52:41PM +0800, Xing Zheng wrote: > > Hi, > Being careless, judge the return value of snd_soc_card_jack_new > is opposite, so it should be fixed. Please don't send cover letters for single patches, if there is anything useful it should either be in the changelog or after the --- in the patch.

1 0

[alsa-devel] [sound:for-next 174/174] sound/ppc/keywest.c:104:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE'
by kbuild test robot 25 Aug '15

25 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: a2bc2af66a5b656577fc2d71b6c277b73537777e commit: a2bc2af66a5b656577fc2d71b6c277b73537777e [174/174] ALSA: ppc: keywest: Export I2C module alias information config: powerpc-defconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout a2bc2af66a5b656577fc2d71b6c277b73537777e # save the attached .config to linux build tree make.cross ARCH=powerpc All error/warnings (new ones prefixed by >>): >> sound/ppc/keywest.c:104:1: warning: data definition has no type or storage class MODULE_DEVICE_TABLE(i2c, keywest_i2c_id); ^ >> sound/ppc/keywest.c:104:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int] >> sound/ppc/keywest.c:104:1: warning: parameter names (without types) in function declaration cc1: some warnings being treated as errors vim +104 sound/ppc/keywest.c 98 99 static const struct i2c_device_id keywest_i2c_id[] = { 100 { "MAC,tas3004", 0 }, /* instantiated by i2c-powermac */ 101 { "keywest", 0 }, /* instantiated by us if needed */ 102 { } 103 }; > 104 MODULE_DEVICE_TABLE(i2c, keywest_i2c_id); 105 106 static struct i2c_driver keywest_driver = { 107 .driver = { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 3

[alsa-devel] [PATCH] ALSA: ppc: Add missing inclusion of linux/module.h
by Takashi Iwai 25 Aug '15

25 Aug '15

Otherwise it triggers a compile warning like: sound/ppc/keywest.c:104:1: warning: data definition has no type or storage class sound/ppc/keywest.c:104:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int] Fixes: a2bc2af66a5b ('ALSA: ppc: keywest: Export I2C module alias information') Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/ppc/keywest.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index f644a8c57e0a..4373615f13e2 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -22,6 +22,7 @@ #include <linux/init.h> #include <linux/i2c.h> #include <linux/delay.h> +#include <linux/module.h> #include <sound/core.h> #include "pmac.h" -- 2.5.0

2 1

[alsa-devel] [RESEND PATCH 0/7] Export I2C and OF module aliases in missing drivers
by Javier Martinez Canillas 25 Aug '15

25 Aug '15

Hello, This is a resend of the patches that were not picked from the series "[PATCH 00/27] Export I2C and OF module aliases in missing drivers" [0] posted about a month ago. The patches have no dependencies and can be picked individually by the relevant maintainer. I preferred to resend instead of sending a naked ping for each patch that I got no answer. Best regards, Javier Javier Martinez Canillas (7): i2c: core: Export I2C module alias information in dummy driver backlight: tosa: Export I2C module alias information usb: phy: isp1301: Export I2C module alias information ALSA: ppc: keywest: Export I2C module alias information extcon: Export OF module alias information in missing drivers leds: Export OF module alias information in missing drivers regulator: isl9305: Export OF module alias information drivers/extcon/extcon-rt8973a.c | 1 + drivers/extcon/extcon-sm5502.c | 1 + drivers/i2c/i2c-core.c | 1 + drivers/leds/leds-pca963x.c | 1 + drivers/leds/leds-tca6507.c | 1 + drivers/regulator/isl9305.c | 1 + drivers/usb/phy/phy-isp1301.c | 1 + drivers/video/backlight/tosa_bl.c | 1 + sound/ppc/keywest.c | 1 + 9 files changed, 9 insertions(+) -- 2.4.3

2 2

[alsa-devel] [PATCH 1/3] ASoC: davinci: Convert to use devm_ioremap_resource
by Axel Lin 25 Aug '15

25 Aug '15

Use devm_ioremap_resource() instead of open code. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/davinci/davinci-i2s.c | 25 ++++++------------------- sound/soc/davinci/davinci-mcasp.c | 18 ++++-------------- 2 files changed, 10 insertions(+), 33 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 56cb4d9..ec98548 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -651,23 +651,15 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea, *res; + struct resource *mem, *res; + void __iomem *io_base; int *dma; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "no mem resource?\n"); - return -ENODEV; - } - - ioarea = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), - pdev->name); - if (!ioarea) { - dev_err(&pdev->dev, "McBSP region already claimed\n"); - return -EBUSY; - } + io_base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(io_base)) + return PTR_ERR(io_base); dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcbsp_dev), GFP_KERNEL); @@ -679,12 +671,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) return -ENODEV; clk_enable(dev->clk); - dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!dev->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_release_clk; - } + dev->base = io_base; dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b960e62..add6bb9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1613,7 +1613,7 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; - struct resource *mem, *ioarea, *res, *dat; + struct resource *mem, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; char *irq_name; @@ -1648,22 +1648,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } } - ioarea = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), pdev->name); - if (!ioarea) { - dev_err(&pdev->dev, "Audio region already claimed\n"); - return -EBUSY; - } + mcasp->base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(mcasp->base)) + return PTR_ERR(mcasp->base); pm_runtime_enable(&pdev->dev); - mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!mcasp->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err; - } - mcasp->op_mode = pdata->op_mode; /* sanity check for tdm slots parameter */ if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) { -- 2.1.0

3 7

[alsa-devel] Applied "ASoC: cs4349: fix platform_no_drv_owner.cocci warnings" to the asoc tree
by Mark Brown 25 Aug '15

25 Aug '15

The patch ASoC: cs4349: fix platform_no_drv_owner.cocci warnings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f755d114cba98708932cba89d8c98c4ab89a83b0 Mon Sep 17 00:00:00 2001 From: kbuild test robot <fengguang.wu(a)intel.com> Date: Sun, 23 Aug 2015 00:30:19 +0800 Subject: [PATCH] ASoC: cs4349: fix platform_no_drv_owner.cocci warnings sound/soc/codecs/cs4349.c:389:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs4349.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 0d010c2b6021..0ac8fc5ed4ae 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -378,7 +378,6 @@ MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", - .owner = THIS_MODULE, .of_match_table = cs4349_of_match, }, .id_table = cs4349_i2c_id, -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: tegra: Fix unused variable 'spdif' warning
by Axel Lin 25 Aug '15

25 Aug '15

Fix below build warning: CC [M] sound/soc/tegra/tegra20_spdif.o sound/soc/tegra/tegra20_spdif.c: In function 'tegra20_spdif_platform_remove': sound/soc/tegra/tegra20_spdif.c:361:24: warning: unused variable 'spdif' [-Wunused-variable] Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/tegra/tegra20_spdif.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 66237bc..0809b1e 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -358,8 +358,6 @@ err_pm_disable: static int tegra20_spdif_platform_remove(struct platform_device *pdev) { - struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_spdif_runtime_suspend(&pdev->dev); -- 2.1.0

3 2

[alsa-devel] [PATCH] ALSA: hda - Refresh sysfs at snd_hda_codec_update_widgets()
by Takashi Iwai 25 Aug '15

25 Aug '15

In the commit [fa4f18b4f402: ALSA: hda - Refresh widgets sysfs at probing Haswell+ HDMI codecs], snd_hdac_refresh_widget_sysfs() is explicitly called in the codec driver. But this results in refreshing twice, as snd_hdac_refresh_widget_sysfs() itself calls snd_hdac_refresh_widgets() function. Instead, we can replace the call in snd_hda_codec_update_widgets() with snd_hdac_refresh_widget_sysfs(). This also fixes the missing sysfs update for ca0132, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/patch_hdmi.c | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 792948418cf0..2c021911f88a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -955,7 +955,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) hda_nid_t fg; int err; - err = snd_hdac_refresh_widgets(&codec->core); + err = snd_hdac_refresh_widget_sysfs(&codec->core); if (err < 0) return err; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index df982fc6a872..a97db5fc8a15 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2331,7 +2331,6 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (is_haswell_plus(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); - snd_hdac_refresh_widget_sysfs(&codec->core); } /* For Valleyview/Cherryview, only the display codec is in the display -- 2.5.0

1 0

[alsa-devel] [PATCH v5 0/6] Add dw_hdmi i2s audio support
by Yakir Yang 25 Aug '15

25 Aug '15

As Russell have send an series patches for dw_hdmi audio, and most important it works well on rockchip platform. Cause some patches have been achieve in Russell's series, so I abondon some patches in this verison. Here are the abondon patches that already be achieved: * drm: bridge/dw_hdmi: adjust n/cts setting order * drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual * drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() * drm: bridge/dw_hdmi: add enable/disable to dw_hdmi_audio callbacks And I also abondon some interrupt patches that is not a common bug in dw_hdmi driver, but is a rockchip platform bug which HDMI registers would reset by CRU module after system suspend/resume (in this case HPD interrupt mask would reset, then driver can't respond hot plug event after system resume). Here are the abondon patches that used to "fix" dw_hdmi interrupt bug: * drm: bridge/dw_hdmi: add irq control to suspend/resume * drm: bridge/dw_hdmi: wrap irq control in fucntions * drm: rockchip/dw_hdmi_rockchip: add resume/suspend support Here are the leave list that I have make in my series: - add n/cts combinations for more display modes - enable audio support when sink device is HDMI monitor and has audio - rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h - add dw_hdmi i2s audio driver - add rockchip_hdmi_audio sound card driver - add dt-bindings for documentation for rockchip_hdmi_audio driver Changes in v5: - make more words in commit message. - Take Russell suggest that create "sink_has_audio" and "sink_is_hdmi" in struct hdmi, and keep vmode's mdvi changeless. - Config fc_invidconf to HDMI mode when sink device is HDMI monitor. - Take Mark Brown suggest that remove jack_status recorded, report jack status directly when hdmi plug happend, and remove devm_kfree and devm_free_irq. - Correct the MODULE_LICENSE to "GPL v2" - Move source from sound/soc/codecs to drivers/gpu/drm/bridge/ - Rename dai driver name to "dw-hdmi-i2s-hifi" - Take Mark Brown suggest that remove the no useful code in probe func, and remove the snd_soc_dai_set_fmt() snd_soc_unregister_card(), remove those noisy dev_info() - Rename codec_dai_name to "dw-hdmi-i2s-hifi" Changes in v4: - Take Dainel suggest that introduce drm_detect_monitor_audio() to judge whether source should config audio. - Replace delaywork with irq thread, and add suspend/resume interfaces, replace "dw-hdmi-audio" with consecutive strings. - Add ".pm = &snd_soc_pm_ops," Changes in v3: - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Yakir Yang (6): drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio when sink device is HDMI and has audio drm: bridge/dw_hdmi: rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h drm: bridge/dw_hdmi-i2s-audio: add audio driver ASoC: rockchip/rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for rockchip_hdmi_audio driver .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/Kconfig | 9 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 2 +- .../{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} | 0 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c | 398 +++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi.c | 100 +++++- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 138 +++++++ 11 files changed, 658 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt rename drivers/gpu/drm/bridge/{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} (100%) create mode 100644 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

4 10

[alsa-devel] [PATCH 1/2] ALSA: hda - Check all inputs for is_active_nid_for_any()
by Takashi Iwai 25 Aug '15

25 Aug '15

The is_active_nid_for_any() function in the generic parser is supposed to check all connections from/to the given widget, but the current code checks only the first input connection (index = 0). This patch corrects the code to check all inputs by passing -1 to index argument. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102521 Cc: <stable(a)vger.kernel.org> [v4.1+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b077bb644434..368181d8b0ce 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -671,7 +671,8 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, } for (i = 0; i < path->depth; i++) { if (path->path[i] == nid) { - if (dir == HDA_OUTPUT || path->idx[i] == idx) + if (dir == HDA_OUTPUT || idx == -1 || + path->idx[i] == idx) return true; break; } @@ -682,7 +683,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, /* check whether the NID is referred by any active paths */ #define is_active_nid_for_any(codec, nid) \ - is_active_nid(codec, nid, HDA_OUTPUT, 0) + is_active_nid(codec, nid, HDA_OUTPUT, -1) /* get the default amp value for the target state */ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, -- 2.5.0

1 1

[alsa-devel] [PATCH (alsa-lib)] pcm: Fix doxygen for two enums
by David Henningsson 24 Aug '15

24 Aug '15

The doxygen comments were wrong, making doxygen output weird. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/pcm.h | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index a1d14a9..0be1a32 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -324,9 +324,9 @@ typedef enum _snd_pcm_tstamp { } snd_pcm_tstamp_t; typedef enum _snd_pcm_tstamp_type { - SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /** gettimeofday equivalent */ - SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ - SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /** monotonic_raw (no NTP) */ + SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /**< gettimeofday equivalent */ + SND_PCM_TSTAMP_TYPE_MONOTONIC, /**< posix_clock_monotonic equivalent */ + SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /**< monotonic_raw (no NTP) */ SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; @@ -527,11 +527,11 @@ int snd_pcm_unlink(snd_pcm_t *pcm); /** channel map list type */ enum snd_pcm_chmap_type { - SND_CHMAP_TYPE_NONE = 0,/** unspecified channel position */ - SND_CHMAP_TYPE_FIXED, /** fixed channel position */ - SND_CHMAP_TYPE_VAR, /** freely swappable channel position */ - SND_CHMAP_TYPE_PAIRED, /** pair-wise swappable channel position */ - SND_CHMAP_TYPE_LAST = SND_CHMAP_TYPE_PAIRED, /** last entry */ + SND_CHMAP_TYPE_NONE = 0,/**< unspecified channel position */ + SND_CHMAP_TYPE_FIXED, /**< fixed channel position */ + SND_CHMAP_TYPE_VAR, /**< freely swappable channel position */ + SND_CHMAP_TYPE_PAIRED, /**< pair-wise swappable channel position */ + SND_CHMAP_TYPE_LAST = SND_CHMAP_TYPE_PAIRED, /**< last entry */ }; /** channel positions */ -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] [resend #2] snd-waveartist: Introduce Rockwell WaveArtist RWA010 driver
by Ondrej Zary 24 Aug '15

24 Aug '15

This is a new ALSA driver driver for Rockwell WaveArtist RWA010 chips found on some (rare) ISA sound cards, such as DCS Multimedia S717. I wasn't able to get the old OSS WaveArtist driver to work with my card but it was a great source of information about the chip (as the full datasheet is not available - only a brief one and a design guide). However, the OSS driver only supports few of the mixer registers so I had to install the card in Windows and dump mixer registers while changing the mixer settings. Then tried what the rest of the registers do and the result is a fully working mixer which even supports more controls than the Windows driver. Someone with a NetWinder can add support for it to this driver and then remove the old OSS one. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- include/sound/mpu401.h | 1 + sound/isa/Kconfig | 11 + sound/isa/Makefile | 2 + sound/isa/waveartist.c | 1399 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/waveartist.h | 74 +++ 5 files changed, 1487 insertions(+) create mode 100644 sound/isa/waveartist.c create mode 100644 sound/isa/waveartist.h diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h index e942096..8fceeba 100644 --- a/include/sound/mpu401.h +++ b/include/sound/mpu401.h @@ -44,6 +44,7 @@ #define MPU401_HW_INTEL8X0 17 /* Intel8x0 driver */ #define MPU401_HW_PC98II 18 /* Roland PC98II */ #define MPU401_HW_AUREAL 19 /* Aureal Vortex */ +#define MPU401_HW_WAVEARTIST 20 /* Rockwell WaveArtist */ #define MPU401_INFO_INPUT (1 << 0) /* input stream */ #define MPU401_INFO_OUTPUT (1 << 1) /* output stream */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 0216475..7a3b4a2 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -454,5 +454,16 @@ config SND_MSND_CLASSIC To compile this driver as a module, choose M here: the module will be called snd-msnd-classic. +config SND_WAVEARTIST + tristate "Rockwell WaveArtist RWA010" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + help + Say Y here to include support for Rockwell WaveArtist RWA010 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-waveartist. + endif # SND_ISA diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..23e11e2 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -12,6 +12,7 @@ snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o snd-sscape-objs := sscape.o +snd-waveartist-objs := waveartist.o # Toplevel Module Dependency obj-$(CONFIG_SND_ADLIB) += snd-adlib.o @@ -23,6 +24,7 @@ obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o +obj-$(CONFIG_SND_WAVEARTIST) += snd-waveartist.o obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ galaxy/ gus/ msnd/ opti9xx/ \ sb/ wavefront/ wss/ diff --git a/sound/isa/waveartist.c b/sound/isa/waveartist.c new file mode 100644 index 0000000..e5bda5c --- /dev/null +++ b/sound/isa/waveartist.c @@ -0,0 +1,1399 @@ +/* + * Driver for Rockwell WaveArtist RWA010 soundcards + * + * Copyright (c) 2015 Ondrej Zary + * + * HW-related parts based on OSS WaveArtist driver by Hannu Savolainen + * + * ALSA code based on ES18xx driver by Christian Fischbach & Abramo Bagnara + * and also by OPL3-SA2 driver by Jaroslav Kysela + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/isa.h> +#include <linux/pnp.h> +#include <linux/isapnp.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> +#include "waveartist.h" + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("Driver for Rockwell WaveArtist RWA010 sound cards"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; +#ifdef CONFIG_PNP +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +#endif +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x250-0x3f0 */ +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x388-0x3f0 */ +static long midi_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;/* 0x300-0x3f0 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,10,11 */ +static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 5,6,7 */ +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for WaveArtist soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for WaveArtist soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable WaveArtist soundcard."); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); +#endif +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for WaveArtist driver."); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for WaveArtist driver."); +module_param_array(midi_port, long, NULL, 0444); +MODULE_PARM_DESC(midi_port, "MIDI port # for WaveArtist driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for WaveArtist driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "DMA1 # for WaveArtist driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 # for WaveArtist driver."); + +#ifdef CONFIG_PNP +static int isa_registered; +static int pnp_registered; +#endif + +#define PFX "waveartist: " + +#ifdef CONFIG_PNP +#define WA_DEVICE(pnpid) { \ + .id = pnpid, \ + .devs = { {"RSS5000"}, {"RSS5001"}, {"RSS5002"} } \ +} +/* + * RSS5000 = WaveArtist, RSS5001 = SB, RSS5002 = MPU-401, RSS5003 = IDE, + * RSS5004 = gameport, RSS5005 = modem, RSS5006 = 3D + */ + +static struct pnp_card_device_id snd_waveartist_pnpids[] = { + WA_DEVICE("RSS5000"), /* 16-bit decode */ + WA_DEVICE("RSS5100"), /* 16-bit decode + modem */ + WA_DEVICE("RSS5200"), /* 10-bit decode + modem */ + WA_DEVICE("RSS5300"), /* 10-bit decode */ + WA_DEVICE("RSS5400"), /* 16-bit decode + IDE */ + WA_DEVICE("RSS5500"), /* 16-bit decode + modem + IDE */ + WA_DEVICE("RSS5600"), /* 10-bit decode + IDE */ + WA_DEVICE("RSS5700"), /* 10-bit decode + modem + IDE */ + WA_DEVICE("RSS5800"), /* 10-bit decode + modem + 3D */ + WA_DEVICE("RSS5900"), /* 10-bit decode + 3D */ + WA_DEVICE("RSS5A00"), /* 16-bit decode + modem + 3D */ + WA_DEVICE("RSS5B00"), /* 16-bit decode + 3D */ + { .id = "" } /* end */ +}; +MODULE_DEVICE_TABLE(pnp_card, snd_waveartist_pnpids); +#endif /* CONFIG_PNP */ + +struct snd_waveartist { +#ifdef CONFIG_PNP + struct pnp_dev *wa; /* WaveArtist device */ + struct pnp_dev *sb; /* SB emulation device */ + struct pnp_dev *mpu; /* MPU-401 device */ +#endif + unsigned long port; /* base port */ + struct resource *res_port; /* base port resource */ + int irq; + int dma_playback; + int dma_capture; + + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + + spinlock_t reg_lock; + struct snd_hwdep *synth; + struct snd_rawmidi *rmidi; + + u16 image[20]; /* mixer registers image */ +}; + +static inline void wa_outb(struct snd_waveartist *chip, u8 reg, u8 val) +{ + outb(val, chip->port + reg); +} + +static inline u8 wa_inb(struct snd_waveartist *chip, u8 reg) +{ + return inb(chip->port + reg); +} + +static inline void wa_outw(struct snd_waveartist *chip, u8 reg, u16 val) +{ + outw(val, chip->port + reg); +} + +static inline u16 wa_inw(struct snd_waveartist *chip, u8 reg) +{ + return inw(chip->port + reg); +} + +static inline void waveartist_set_ctlr(struct snd_waveartist *chip, u8 clear, + u8 set) +{ + clear = ~clear & wa_inb(chip, CTLR); + wa_outb(chip, CTLR, clear | set); +} + +/* acknowledge IRQ */ +static inline void waveartist_iack(struct snd_waveartist *chip) +{ + u8 old_ctlr = wa_inb(chip, CTLR) & ~IRQ_ACK; + + wa_outb(chip, CTLR, old_ctlr | IRQ_ACK); + wa_outb(chip, CTLR, old_ctlr); +} + +static int waveartist_reset(struct snd_waveartist *chip) +{ + unsigned int timeout, res = -1; + + waveartist_set_ctlr(chip, -1, RESET); + msleep(200); + waveartist_set_ctlr(chip, RESET, 0); + + timeout = 500; + do { + mdelay(2); + + if (wa_inb(chip, STATR) & CMD_RF) { + res = wa_inw(chip, CMDR); + if (res == 0x55aa) + break; + } + } while (--timeout); + + if (timeout == 0) { + dev_warn(chip->card->dev, "WaveArtist: reset timeout (res=0x%x)\n", + res); + return 1; + } + + return 0; +} + +/* Helper function to send and receive words + * from WaveArtist. It handles all the handshaking + * and can send or receive multiple words. + */ +static int waveartist_cmd(struct snd_waveartist *chip, + int nr_cmd, u16 *cmd, + int nr_resp, u16 *resp) +{ + unsigned long flags; + unsigned int timed_out = 0, i; + + spin_lock_irqsave(&chip->reg_lock, flags); + /* + * The chip can hang if we access the STATR register too quickly + * after a write. Do a dummy read to slow down. + */ + wa_inb(chip, CTLR); + + if (wa_inb(chip, STATR) & CMD_RF) { + /* flush the port */ + wa_inw(chip, CMDR); + udelay(10); + } + + for (i = 0; !timed_out && i < nr_cmd; i++) { + int count; + + for (count = 5000; count; count--) + if (wa_inb(chip, STATR) & CMD_WE) + break; + + if (!count) + timed_out = 1; + else + wa_outw(chip, CMDR, cmd[i]); + /* Another dummy read */ + wa_inb(chip, CTLR); + } + + for (i = 0; !timed_out && i < nr_resp; i++) { + int count; + + for (count = 5000; count; count--) + if (wa_inb(chip, STATR) & CMD_RF) + break; + + if (!count) + timed_out = 1; + else + resp[i] = wa_inw(chip, CMDR); + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + + return timed_out ? 1 : 0; +} + +/* Send one command word */ +static inline int waveartist_cmd1(struct snd_waveartist *chip, u16 cmd) +{ + return waveartist_cmd(chip, 1, &cmd, 0, NULL); +} + +/* Send one command, receive one word */ +static inline u16 waveartist_cmd1_r(struct snd_waveartist *chip, u16 cmd) +{ + u16 ret; + + waveartist_cmd(chip, 1, &cmd, 1, &ret); + + return ret; +} + +/* Send a double command, receive one word (and throw it away) */ +static inline int waveartist_cmd2(struct snd_waveartist *chip, u16 cmd, u16 arg) +{ + u16 vals[2] = { cmd, arg }; + + return waveartist_cmd(chip, 2, vals, 1, vals); +} + +/* Send a triple command */ +static inline int waveartist_cmd3(struct snd_waveartist *chip, u16 cmd, + u16 arg1, u16 arg2) +{ + u16 vals[3] = { cmd, arg1, arg2 }; + + return waveartist_cmd(chip, 3, vals, 0, NULL); +} + +static u16 waveartist_getrev(struct snd_waveartist *chip) +{ + u16 temp[2]; + u16 cmd = WACMD_GETREV; + + waveartist_cmd(chip, 1, &cmd, 2, temp); + + return temp[0]; +} + +static irqreturn_t snd_waveartist_interrupt(int irq, void *dev_id) +{ + u8 status, irqstatus; + struct snd_card *card = dev_id; + struct snd_waveartist *chip; + + if (card == NULL) + return IRQ_NONE; + + chip = card->private_data; + + irqstatus = wa_inb(chip, IRQSTAT); + status = wa_inb(chip, STATR); + + if (status & IRQ_REQ) /* clear interrupt */ + waveartist_iack(chip); + + if (irqstatus & IRQ_PCM) { /* PCM buffer done */ + if ((status & DMA1) && chip->playback_substream) + snd_pcm_period_elapsed(chip->playback_substream); + if ((status & DMA0) && chip->capture_substream) + snd_pcm_period_elapsed(chip->capture_substream); + if (!(status & (DMA0 | DMA1))) + dev_warn(chip->card->dev, "Unknown PCM interrupt\n"); + } + + if (irqstatus & IRQ_SB) /* we do not use SB mode */ + dev_warn(chip->card->dev, "Unexpected SB interrupt\n"); + + if ((irqstatus & IRQ_MPU) && chip->rmidi) + snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); + + return IRQ_HANDLED; +} + +#ifdef CONFIG_PM +static int snd_waveartist_suspend(struct snd_card *card, pm_message_t state) +{ + struct snd_waveartist *chip = card->private_data; + int i; + + if (!card) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + + /* save mixer registers */ + for (i = 0; i < ARRAY_SIZE(chip->image); i++) + chip->image[i] = waveartist_cmd1_r(chip, + WACMD_GET_LEVEL | i << 8); + + return 0; +} + +static int snd_waveartist_resume(struct snd_card *card) +{ + struct snd_waveartist *chip; + int i; + + if (!card) + return 0; + + chip = card->private_data; + + /* restore mixer registers */ + for (i = 0; i < 10; i += 2) + waveartist_cmd3(chip, WACMD_SET_MIXER, + chip->image[i], chip->image[i + 1]); + for (i = 10; i < ARRAY_SIZE(chip->image); i += 2) + waveartist_cmd3(chip, WACMD_SET_LEVEL | + ((i - 10) / 2) << 8, + chip->image[i], chip->image[i + 1]); + + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif /* CONFIG_PM */ + +#ifdef CONFIG_PNP +static void snd_waveartist_set_irq(struct pnp_dev *pdev, int irq) +{ + if (!pdev->active) + return; + isapnp_cfg_begin(isapnp_card_number(pdev), isapnp_csn_number(pdev)); + isapnp_write_byte(0x70, irq); /* ISAPNP_CFG_IRQ */ + isapnp_cfg_end(); +} + +static int snd_waveartist_pnp(int dev, struct snd_waveartist *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *id) +{ + chip->wa = pnp_request_card_device(card, id->devs[0].id, NULL); + if (!chip->wa) + return -EBUSY; + + chip->sb = pnp_request_card_device(card, id->devs[1].id, NULL); + if (!chip->sb) + return -EBUSY; + + chip->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); + if (!chip->mpu) + return -EBUSY; + + if (pnp_activate_dev(chip->wa) < 0) { + dev_err(chip->card->dev, "WA PnP configure failure\n"); + return -EBUSY; + } + if (pnp_activate_dev(chip->sb) < 0) { + dev_err(chip->card->dev, "SB PnP configure failure\n"); + return -EBUSY; + } + port[dev] = pnp_port_start(chip->wa, 0); + dma2[dev] = pnp_dma(chip->wa, 0); + fm_port[dev] = pnp_port_start(chip->sb, 1); + dma1[dev] = pnp_dma(chip->sb, 0); + irq[dev] = pnp_irq(chip->sb, 0); + + /* + * The card uses only one IRQ (listed in the resources of SB device) + * which needs to be shared by WaveArtist and MPU-401 devices. They + * don't have an IRQ resource so it must be forced. + */ + snd_waveartist_set_irq(chip->wa, irq[dev]); + + /* allocate MPU-401 resources */ + if (pnp_activate_dev(chip->mpu) < 0) + dev_err(chip->card->dev, "MPU-401 PnP configure failure: will be disabled\n"); + else { + midi_port[dev] = pnp_port_start(chip->mpu, 0); + snd_waveartist_set_irq(chip->mpu, irq[dev]); + } + + dev_dbg(chip->card->dev, "PnP WaveArtist: port=0x%lx, fm port=0x%lx, midi port=0x%lx\n", + port[dev], fm_port[dev], midi_port[dev]); + dev_dbg(chip->card->dev, "PnP WaveArtist: dma1=%i, dma2=%i, irq=%i\n", + dma1[dev], dma2[dev], irq[dev]); + + return 0; +} +#endif /* CONFIG_PNP */ + +static int snd_waveartist_playback_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (err < 0) + return err; + + return 0; +} + +static int snd_waveartist_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static enum wa_format waveartist_format(snd_pcm_format_t format) +{ + if (snd_pcm_format_width(format) == 16) + return WA_FMT_S16; + if (snd_pcm_format_unsigned(format)) + return WA_FMT_U8; + else + return WA_FMT_S8; +} + +static u16 waveartist_rate(struct snd_pcm_runtime *runtime) +{ + return (runtime->rate << 16) / 44100; +} + +static int snd_waveartist_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + /* Set rate */ + if (waveartist_cmd2(chip, WACMD_OUTPUTSPEED, waveartist_rate(runtime))) + dev_warn(chip->card->dev, "error setting playback rate %dHz\n", + runtime->rate); + /* Set channel count */ + if (waveartist_cmd2(chip, WACMD_OUTPUTCHANNELS, runtime->channels)) + dev_warn(chip->card->dev, "error setting playback %d channels\n", + runtime->channels); + /* Set DMA channel */ + if (waveartist_cmd2(chip, WACMD_OUTPUTDMA, + chip->dma_playback > 3 ? WA_DMA_16BIT : WA_DMA_8BIT)) + dev_warn(chip->card->dev, "error setting playback data path\n"); + /* Set format */ + if (waveartist_cmd2(chip, WACMD_OUTPUTFORMAT, + waveartist_format(runtime->format))) + dev_warn(chip->card->dev, "error setting playback format %d\n", + runtime->format); + /* Set sample count */ + if (waveartist_cmd2(chip, WACMD_OUTPUTSIZE, count - 1)) + dev_warn(chip->card->dev, "error setting playback count %d\n", + count); + /* Configure DMA controller */ + snd_dma_program(chip->dma_playback, runtime->dma_addr, size, + DMA_MODE_WRITE | DMA_AUTOINIT); + + return 0; +} + +static int snd_waveartist_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + waveartist_cmd1(chip, WACMD_OUTPUTSTART); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + waveartist_cmd1(chip, WACMD_OUTPUTSTOP); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + waveartist_cmd1(chip, WACMD_OUTPUTPAUSE); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + waveartist_cmd1(chip, WACMD_OUTPUTRESUME); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int snd_waveartist_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (err < 0) + return err; + + return 0; +} + +static int snd_waveartist_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + /* Set rate */ + if (waveartist_cmd2(chip, WACMD_INPUTSPEED, waveartist_rate(runtime))) + dev_warn(chip->card->dev, "error setting capture rate %dHz\n", + runtime->rate); + /* Set channel count */ + if (waveartist_cmd2(chip, WACMD_INPUTCHANNELS, runtime->channels)) + dev_warn(chip->card->dev, "error setting capture %d channels\n", + runtime->channels); + /* Set DMA channel */ + if (waveartist_cmd2(chip, WACMD_INPUTDMA, + chip->dma_capture > 3 ? WA_DMA_16BIT : WA_DMA_8BIT)) + dev_warn(chip->card->dev, "error setting capture data path\n"); + /* Set format */ + if (waveartist_cmd2(chip, WACMD_INPUTFORMAT, + waveartist_format(runtime->format))) + dev_warn(chip->card->dev, "error setting capture format %d\n", + runtime->format); + /* Set sample count */ + if (waveartist_cmd2(chip, WACMD_INPUTSIZE, count - 1)) + dev_warn(chip->card->dev, "error setting capture count %d\n", + count); + /* Configure DMA controller */ + snd_dma_program(chip->dma_capture, runtime->dma_addr, size, + DMA_MODE_READ | DMA_AUTOINIT); + + return 0; +} + +static int snd_waveartist_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + waveartist_cmd1(chip, WACMD_INPUTSTART); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + waveartist_cmd1(chip, WACMD_INPUTSTOP); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + waveartist_cmd1(chip, WACMD_INPUTPAUSE); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + waveartist_cmd1(chip, WACMD_INPUTRESUME); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_waveartist_playback_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + size_t size = snd_pcm_lib_buffer_bytes(substream); + size_t ptr = snd_dma_pointer(chip->dma_playback, size); + + return bytes_to_frames(substream->runtime, ptr); +} + +static snd_pcm_uframes_t snd_waveartist_capture_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + size_t size = snd_pcm_lib_buffer_bytes(substream); + size_t ptr = snd_dma_pointer(chip->dma_capture, size); + + return bytes_to_frames(substream->runtime, ptr); +} + +static struct snd_pcm_hardware snd_waveartist_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_44100, + .rate_min = 4000, + .rate_max = 44100, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static struct snd_pcm_hardware snd_waveartist_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_44100, + .rate_min = 4000, + .rate_max = 44100, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static int snd_waveartist_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = substream; + substream->runtime->hw = snd_waveartist_playback; + + snd_pcm_limit_isa_dma_size(chip->dma_playback, + &substream->runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma_playback, + &substream->runtime->hw.period_bytes_max); + + return 0; +} + +static int snd_waveartist_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = substream; + substream->runtime->hw = snd_waveartist_capture; + + snd_pcm_limit_isa_dma_size(chip->dma_capture, + &substream->runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma_capture, + &substream->runtime->hw.period_bytes_max); + + return 0; +} + +static int snd_waveartist_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = NULL; + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int snd_waveartist_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = NULL; + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static struct snd_pcm_ops snd_waveartist_playback_ops = { + .open = snd_waveartist_playback_open, + .close = snd_waveartist_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_waveartist_playback_hw_params, + .hw_free = snd_waveartist_pcm_hw_free, + .prepare = snd_waveartist_playback_prepare, + .trigger = snd_waveartist_playback_trigger, + .pointer = snd_waveartist_playback_pointer, +}; + +static struct snd_pcm_ops snd_waveartist_capture_ops = { + .open = snd_waveartist_capture_open, + .close = snd_waveartist_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_waveartist_capture_hw_params, + .hw_free = snd_waveartist_pcm_hw_free, + .prepare = snd_waveartist_capture_prepare, + .trigger = snd_waveartist_capture_trigger, + .pointer = snd_waveartist_capture_pointer, +}; + +static int snd_waveartist_pcm(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "WaveArtist", 0, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_waveartist_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_waveartist_capture_ops); + + /* global setup */ + pcm->private_data = chip; + pcm->info_flags = 0; + if (chip->dma_playback == chip->dma_capture) + pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX; + strcpy(pcm->name, card->shortname); + chip->pcm = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), 64 * 1024, + chip->dma_playback > 3 || chip->dma_capture > 3 ? + 128 * 1024 : 64 * 1024); + + return 0; +} + +static void snd_waveartist_free(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + + release_and_free_resource(chip->res_port); + if (chip->irq >= 0) + free_irq(chip->irq, card); + if (chip->dma_playback >= 0) { + disable_dma(chip->dma_playback); + free_dma(chip->dma_playback); + } + if (chip->dma_capture >= 0 && chip->dma_capture != chip->dma_playback) { + disable_dma(chip->dma_capture); + free_dma(chip->dma_capture); + } +} + +static int snd_waveartist_dev_free(struct snd_device *device) +{ + snd_waveartist_free(device->card); + + return 0; +} + +static int snd_waveartist_init(struct snd_waveartist *chip) +{ + if (waveartist_reset(chip)) + return -ENODEV; + dev_dbg(chip->card->dev, "chip rev. 0x%x\n", waveartist_getrev(chip)); + + waveartist_cmd1(chip, WACMD_RST_MIXER); /* reset mixer */ + waveartist_iack(chip); /* clear any pending interrupt */ + waveartist_set_ctlr(chip, 0, DMA1_IE | DMA0_IE); /* enable DMA IRQs */ + waveartist_iack(chip); /* clear any pending interrupt */ + + return 0; +} + +static int snd_waveartist_new_device(struct snd_card *card, + unsigned long port, + unsigned long mpu_port, + unsigned long fm_port, + int irq, int dma1, int dma2) +{ + struct snd_waveartist *chip = card->private_data; + static struct snd_device_ops ops = { + .dev_free = snd_waveartist_dev_free, + }; + int err; + + spin_lock_init(&chip->reg_lock); + chip->card = card; + chip->port = port; + chip->irq = -1; + chip->dma_playback = -1; + chip->dma_capture = -1; + + chip->res_port = request_region(port, 16, "WaveArtist"); + if (!chip->res_port) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab ports 0x%lx-0x%lx\n", + port, port + 16 - 1); + return -EBUSY; + } + + if (snd_waveartist_init(chip) < 0) { + snd_waveartist_free(card); + return -ENODEV; + } + + if (request_irq(irq, snd_waveartist_interrupt, 0, "WaveArtist", card)) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab IRQ %d\n", irq); + return -EBUSY; + } + chip->irq = irq; + + if (dma1 == dma2 || dma1 == SNDRV_AUTO_DMA || dma2 == SNDRV_AUTO_DMA) { + /* we have only one DMA channel */ + if (dma1 == SNDRV_AUTO_DMA) + dma1 = dma2; + if (request_dma(dma1, "WaveArtist")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab DMA %d\n", + dma1); + return -EBUSY; + } + chip->dma_playback = chip->dma_capture = dma1; + } else { + /* + * 2 channels: use 16-bit for playback and 8-bit for capture as + * full-duplex works better this way. However, the chip seems to + * have some band-width limit so full-duplex at 44kHz/16-bit/2ch + * is not possible - some frames are dropped during capture, + * resulting in too-fast recording. If capture is done using + * lower rate or 8-bit or mono, everything is fine. + */ + if (dma2 > 3) + swap(dma1, dma2); + + if (request_dma(dma1, "WaveArtist playback")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab playback DMA %d\n", + dma1); + return -EBUSY; + } + chip->dma_playback = dma1; + + if (dma2 != dma1 && request_dma(dma2, "WaveArtist capture")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab capture DMA %d\n", + dma2); + return -EBUSY; + } + chip->dma_capture = dma2; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_waveartist_free(card); + return err; + } + + return 0; +} + +static int snd_waveartist_card_new(struct device *pdev, int dev, + struct snd_card **cardp) +{ + return snd_card_new(pdev, index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_waveartist), cardp); +} + +static int snd_waveartist_info_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + + return 0; +} + +static int snd_waveartist_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u16 val; + + val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | reg << 8); + ucontrol->value.integer.value[0] = (val >> shift) & mask; + + return 0; +} + +static int snd_waveartist_put_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u16 val, old_val, new_val; + + val = (ucontrol->value.integer.value[0] & mask); + mask <<= shift; + val <<= shift; + + old_val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | reg << 8); + new_val = (old_val & ~mask) | (val & mask); + + if (new_val == old_val) + return 0; + + /* new_val already contains register number (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_val, new_val); + + return 1; +} + +static int snd_waveartist_info_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask ? mask : 0x7fff; + + return 0; +} + +static int snd_waveartist_get_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x0f; + int shift_right = (kcontrol->private_value >> 20) & 0x0f; + int mask = (kcontrol->private_value >> 24) & 0xff; + u16 left, right; + + if (mask == 0) + mask = 0x7fff; + + left = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | left_reg << 8); + if (left_reg != right_reg) + right = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | + right_reg << 8); + else + right = left; + + ucontrol->value.integer.value[0] = (left >> shift_left) & mask; + ucontrol->value.integer.value[1] = (right >> shift_right) & mask; + + return 0; +} + +static int snd_waveartist_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x0f; + int shift_right = (kcontrol->private_value >> 20) & 0x0f; + int mask = (kcontrol->private_value >> 24) & 0xff; + u16 val1, val2, mask1, mask2; + u16 old_left, old_right, new_left, new_right; + + if (mask == 0) + mask = 0x7fff; + + val1 = ucontrol->value.integer.value[0] & mask; + val2 = ucontrol->value.integer.value[1] & mask; + val1 <<= shift_left; + val2 <<= shift_right; + mask1 = mask << shift_left; + mask2 = mask << shift_right; + + old_left = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | left_reg << 8); + if (left_reg != right_reg) + old_right = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | + right_reg << 8); + else + old_right = old_left; + + new_left = (old_left & ~mask1) | (val1 & mask1); + new_right = (old_right & ~mask2) | (val2 & mask2); + + if (new_left == old_left && new_right == old_right) + return 0; + + if (left_reg < 10) + /* new_* already contain reg. num. (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_left, new_right); + else + waveartist_cmd3(chip, WACMD_SET_LEVEL | + ((left_reg - 10) / 2) << 8, + new_left, new_right); + + return 1; +} + +static int snd_waveartist_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const mux_texts[] = { + "None", "Mix", "Line", "Phone", "CD", "Mic" + }; + + return snd_ctl_enum_info(uinfo, 2, ARRAY_SIZE(mux_texts), mux_texts); +} + +static int snd_waveartist_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int mux = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | 0x08 << 8); + + ucontrol->value.enumerated.item[0] = mux & 0x07; + ucontrol->value.enumerated.item[1] = (mux >> 3) & 0x07; + + return 0; +} + +static int snd_waveartist_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + u16 old_val, new_val; + u16 val1 = ucontrol->value.enumerated.item[0]; + u16 val2 = ucontrol->value.enumerated.item[1]; + + old_val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | 0x08 << 8); + new_val = (old_val & ~0x3f) | (val1 & 0x07) | ((val2 & 0x07) << 3); + if (new_val == old_val) + return 0; + + /* new_val already contains register number (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_val, new_val); + + return 1; +} + +#define WAVEARTIST_SINGLE(xname, reg, shift, mask) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_waveartist_info_single, \ + .get = snd_waveartist_get_single, .put = snd_waveartist_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) \ +} + +#define WAVEARTIST_DOUBLE(xname, left_reg, right_reg, shift_left, shift_right, \ + mask) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_waveartist_info_double, \ + .get = snd_waveartist_get_double, .put = snd_waveartist_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 20) | (mask << 24) \ +} + +static struct snd_kcontrol_new snd_waveartist_controls[] = { +WAVEARTIST_DOUBLE("Master Playback Volume", 2, 6, 1, 1, 0x07), +WAVEARTIST_DOUBLE("Master Playback Switch", 0, 4, 0, 0, 1), +WAVEARTIST_DOUBLE("PCM Playback Volume", 10, 11, 0, 0, 0x00),/* 0x00 = 0x7fff */ +WAVEARTIST_DOUBLE("FM Playback Volume", 12, 13, 0, 0, 0x00),/* 0x00 = 0x7fff */ +/* WAVEARTIST_DOUBLE("Wavetable? Playback Volume", 14, 15, 0, 0, 0x00), */ +/* WAVEARTIST_DOUBLE("? Playback Volume", 16, 17, 0, 0, 0x00), */ +/* WAVEARTIST_DOUBLE("? Playback Volume", 18, 19, 0, 0, 0x00), */ +WAVEARTIST_DOUBLE("Digital Playback Switch", 3, 7, 10, 10, 1), /* PCM + FM */ +WAVEARTIST_DOUBLE("CD Playback Volume", 0, 4, 1, 1, 0x1f), +WAVEARTIST_DOUBLE("CD Playback Switch", 3, 7, 6, 6, 1), +WAVEARTIST_DOUBLE("Line Playback Volume", 0, 4, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Line Playback Switch", 3, 7, 4, 4, 1), +WAVEARTIST_DOUBLE("Phone Playback Volume", 1, 5, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Phone Playback Switch", 3, 7, 5, 5, 1), +WAVEARTIST_SINGLE("Mono Playback Volume", 8, 5, 0x1f), +WAVEARTIST_DOUBLE("Mono Playback Switch", 3, 7, 9, 9, 1), +WAVEARTIST_DOUBLE("Mic Playback Volume", 2, 6, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Mic 2 Playback Volume", 1, 5, 1, 1, 0x1f), +WAVEARTIST_DOUBLE("Mic Playback Switch", 3, 7, 7, 7, 1), +WAVEARTIST_DOUBLE("Mic 2 Playback Switch", 3, 7, 8, 8, 1), +WAVEARTIST_DOUBLE("Mic Gain", 2, 6, 4, 4, 0x03), +WAVEARTIST_DOUBLE("Capture Volume", 3, 7, 0, 0, 0x0f), +WAVEARTIST_SINGLE("Mono Output Playback Switch", 1, 0, 1), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_waveartist_info_mux, + .get = snd_waveartist_get_mux, + .put = snd_waveartist_put_mux, +} +}; + +static int snd_waveartist_mixer(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + int err; + unsigned int idx; + + strcpy(card->mixername, chip->pcm->name); + + for (idx = 0; idx < ARRAY_SIZE(snd_waveartist_controls); idx++) { + struct snd_kcontrol *kctl; + + kctl = snd_ctl_new1(&snd_waveartist_controls[idx], chip); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + + return 0; +} + +static int snd_waveartist_probe(struct snd_card *card, int dev) +{ + struct snd_waveartist *chip = card->private_data; + struct snd_opl3 *opl3; + int err; + + err = snd_waveartist_new_device(card, + port[dev], midi_port[dev], fm_port[dev], + irq[dev], dma1[dev], dma2[dev]); + if (err < 0) + return err; + + strcpy(card->driver, "WaveArtist"); + strcpy(card->shortname, "Rockwell WaveArtist RWA010"); + sprintf(card->longname, "%s at 0x%lx, irq %d, dma1 %d, dma2 %d", + card->shortname, chip->port, irq[dev], dma1[dev], dma2[dev]); + + err = snd_waveartist_pcm(card); + if (err < 0) + return err; + err = snd_waveartist_mixer(card); + if (err < 0) + return err; + + if (fm_port[dev] >= 0x388 && fm_port[dev] < 0x3f0) { + err = snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3, 0, &opl3); + if (err < 0) + return err; + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + return err; + err = snd_opl3_hwdep_new(opl3, 0, 1, &chip->synth); + if (err < 0) + return err; + } + if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x3f0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_WAVEARTIST, + midi_port[dev], MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); + if (err < 0) + return err; + } + + return snd_card_register(card); +} + +static int snd_waveartist_isa_match(struct device *pdev, + unsigned int dev) +{ + if (!enable[dev]) + return 0; +#ifdef CONFIG_PNP + if (isapnp[dev]) + return 0; +#endif + if (port[dev] == SNDRV_AUTO_PORT) { + dev_err(pdev, "specify port\n"); + return 0; + } + if (irq[dev] == SNDRV_AUTO_IRQ) { + dev_err(pdev, "specify irq\n"); + return 0; + } + if (dma1[dev] == SNDRV_AUTO_DMA) { + dev_err(pdev, "specify dma1\n"); + return 0; + } + + return 1; +} + +static int snd_waveartist_isa_probe(struct device *pdev, + unsigned int dev) +{ + struct snd_card *card; + int err; + + err = snd_waveartist_card_new(pdev, dev, &card); + if (err < 0) + return err; + err = snd_waveartist_probe(card, dev); + if (err < 0) { + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; +} + +static int snd_waveartist_isa_remove(struct device *devptr, + unsigned int dev) +{ + snd_card_free(dev_get_drvdata(devptr)); + + return 0; +} + +#ifdef CONFIG_PM +static int snd_waveartist_isa_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_waveartist_suspend(dev_get_drvdata(dev), state); +} + +static int snd_waveartist_isa_resume(struct device *dev, unsigned int n) +{ + return snd_waveartist_resume(dev_get_drvdata(dev)); +} +#endif + +static struct isa_driver waveartist_isa_driver = { + .match = snd_waveartist_isa_match, + .probe = snd_waveartist_isa_probe, + .remove = snd_waveartist_isa_remove, +#ifdef CONFIG_PM + .suspend = snd_waveartist_isa_suspend, + .resume = snd_waveartist_isa_resume, +#endif + .driver = { .name = "waveartist" }, +}; + +#ifdef CONFIG_PNP +static int snd_waveartist_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int dev; + int err; + struct snd_card *card; + + for (; dev < SNDRV_CARDS; dev++) { + if (enable[dev] && isapnp[dev]) + break; + } + if (dev >= SNDRV_CARDS) + return -ENODEV; + + err = snd_waveartist_card_new(&pcard->card->dev, dev, &card); + if (err < 0) + return err; + err = snd_waveartist_pnp(dev, card->private_data, pcard, pid); + if (err < 0) { + dev_err(&pcard->card->dev, "PnP detection failed\n"); + snd_card_free(card); + return err; + } + err = snd_waveartist_probe(card, dev); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + dev++; + + return 0; +} + +static void snd_waveartist_pnp_remove(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + /* disable forced IRQs */ + snd_waveartist_set_irq(chip->wa, 0); + snd_waveartist_set_irq(chip->mpu, 0); + + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +#ifdef CONFIG_PM +static int snd_waveartist_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + snd_waveartist_suspend(card, state); + + /* disable forced IRQs to prevent opps */ + snd_waveartist_set_irq(chip->wa, 0); + snd_waveartist_set_irq(chip->mpu, 0); + + return 0; +} +static int snd_waveartist_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + /* re-enable forced IRQs */ + snd_waveartist_set_irq(chip->wa, chip->irq); + snd_waveartist_set_irq(chip->mpu, chip->irq); + + return snd_waveartist_resume(pnp_get_card_drvdata(pcard)); +} +#endif + +static struct pnp_card_driver waveartist_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "waveartist", + .id_table = snd_waveartist_pnpids, + .probe = snd_waveartist_pnp_detect, + .remove = snd_waveartist_pnp_remove, +#ifdef CONFIG_PM + .suspend = snd_waveartist_pnp_suspend, + .resume = snd_waveartist_pnp_resume, +#endif +}; +#endif /* CONFIG_PNP */ + +static int __init alsa_card_waveartist_init(void) +{ + int err; + + err = isa_register_driver(&waveartist_isa_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&waveartist_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit alsa_card_waveartist_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&waveartist_pnpc_driver); + + if (isa_registered) +#endif + isa_unregister_driver(&waveartist_isa_driver); +} + +module_init(alsa_card_waveartist_init) +module_exit(alsa_card_waveartist_exit) diff --git a/sound/isa/waveartist.h b/sound/isa/waveartist.h new file mode 100644 index 0000000..72254cf --- /dev/null +++ b/sound/isa/waveartist.h @@ -0,0 +1,74 @@ +/* + * Rockwell WaveArtist RWA010 chip register definitions + * + * Based on OSS WaveArtist driver by Hannu Savolainen + */ + +/* registers */ +#define CMDR 0 /* command (16-bit) */ +#define DATR 2 /* data (for PIO?) (16-bit) */ +#define CTLR 4 /* control */ +#define STATR 5 /* status */ +#define EXPCR1 6 /* expansion control 1 */ +#define EXPCR2 7 /* expansion control 2 */ +#define EXPDAT1 8 /* expansion data 1 (16-bit) */ +#define EXPDAT2 10 /* expansion data 2 (16-bit) */ +#define IRQSTAT 12 /* IRQ status */ + +/* STATR register bit definitions */ +#define CMD_WE BIT(7) /* CMDR write empty (ready for write) */ +#define CMD_RF BIT(6) /* CMDR read full (ready for read) */ +#define DAT_WE BIT(5) /* DATR write empty (ready for write) */ +#define DAT_RF BIT(4) /* DATR read full (ready for read) */ +#define IRQ_REQ BIT(3) /* IRQ was requested */ +#define DMA1 BIT(2) /* DMA1 IRQ was requested */ +#define DMA0 BIT(1) /* DMA0 IRQ was requested */ + +/* CTLR register bit definitions */ +#define CMD_WEIE BIT(7) /* CMDR write empty interrupt enable */ +#define CMD_RFIE BIT(6) /* CMDR read full interrupt enable */ +#define DAT_WEIE BIT(5) /* DATR write empty interrupt enable */ +#define DAT_RFIE BIT(4) /* DATR read full interrupt enable */ +#define RESET BIT(3) /* chip reset */ +#define DMA1_IE BIT(2) /* DMA1 interrupt enable */ +#define DMA0_IE BIT(1) /* DMA0 interrupt enable */ +#define IRQ_ACK BIT(0) /* IRQ acknowlege */ + +/* IRQSTAT register bit definitions */ +#define IRQ_MPU BIT(2) /* MPU-401 */ +#define IRQ_SB BIT(1) /* SB emulation */ +#define IRQ_PCM BIT(0) /* native PCM */ + +/* commands */ +#define WACMD_GETREV 0x00 + +#define WACMD_INPUTFORMAT 0x10 /* 0=S8, 1=S16, 2=U8 */ +#define WACMD_INPUTCHANNELS 0x11 /* 1=mono, 2=Stereo */ +#define WACMD_INPUTSPEED 0x12 /* sampling rate */ +#define WACMD_INPUTDMA 0x13 /* 0=8bit, 1=16bit, 2=PIO */ +#define WACMD_INPUTSIZE 0x14 /* samples to interrupt */ +#define WACMD_INPUTSTART 0x15 /* start ADC */ +#define WACMD_INPUTPAUSE 0x16 /* pause ADC */ +#define WACMD_INPUTSTOP 0x17 /* stop ADC */ +#define WACMD_INPUTRESUME 0x18 /* resume ADC */ +#define WACMD_INPUTPIO 0x19 /* PIO ADC */ + +#define WACMD_OUTPUTFORMAT 0x20 /* 0=S8, 1=S16, 2=U8 */ +#define WACMD_OUTPUTCHANNELS 0x21 /* 1=mono, 2=stereo */ +#define WACMD_OUTPUTSPEED 0x22 /* sampling rate */ +#define WACMD_OUTPUTDMA 0x23 /* 0=8bit, 1=16bit, 2=PIO */ +#define WACMD_OUTPUTSIZE 0x24 /* samples to interrupt */ +#define WACMD_OUTPUTSTART 0x25 /* start DAC */ +#define WACMD_OUTPUTPAUSE 0x26 /* pause DAC */ +#define WACMD_OUTPUTSTOP 0x27 /* stop DAC */ +#define WACMD_OUTPUTRESUME 0x28 /* resume DAC */ +#define WACMD_OUTPUTPIO 0x29 /* PIO DAC */ + +#define WACMD_GET_LEVEL 0x30 /* read mixer reg */ +#define WACMD_SET_LEVEL 0x31 /* set mixer regs (10..19) */ +#define WACMD_SET_MIXER 0x32 /* set mixer regs (0..9) */ +#define WACMD_RST_MIXER 0x33 /* mixer reset */ +#define WACMD_SET_MONO 0x34 /* set mono mode (|0x000=L, |0x100=R) */ + +enum wa_format { WA_FMT_S8 = 0, WA_FMT_S16, WA_FMT_U8 }; +enum wa_dma { WA_DMA_8BIT = 0, WA_DMA_16BIT, WA_DMA_PIO }; -- Ondrej Zary

1 1

[alsa-devel] PATCH: ALSA: hda: fix possible null dereference
by Markus Osterhoff 24 Aug '15

24 Aug '15

Upgrading on Gentoo hardened from 4.0.8 to 4.1.6 (with intermediate 4.1.4), I noticed a NULL pointer derefence during booting, and tackled it down to static int add_std_chmaps(struct hda_codec *codec) in file sound/pci/hda/hda_codec.c With commits bbbc7e85 and 751e2216, a for-loop was restated using list_for_each_entry(...); originally, a local pcm* was extracted from an array and checked against NULL, if not-NULL then passed along to snd_pcm_add_chmap_ctls(...). Now, pcm->pcm is passed to snd_pcm_add_chmap_ctls, but the NULL pointer check still uses the “upper level” pcm, not the “in-loop” pcm->pcm. Please have a look at the attached (rather small) suggestion to fix this problem. For a history of the function in question, have a look at lines 2430ff for git diff v4.0 v4.1 sound/pci/hda/hda_codec.c The two commits that touched this function are Commit bbbc7e8502c9 ("ALSA: hda - Allocate hda_pcm objects dynamically") Commit 751e2216899c ("ALSA: hda: fix possible null dereference") where the latter only fixes the order of two fixes, but not the pcm / pcm->pcm confusion. Using v4.1-descendant sources, boot got OOPS like this (two bug-hunting printk's added up front): > pcm: ffff88040cf83400 > pcm->pcm: (nil) > BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 > IP: [<ffffffffa07feab9>] snd_pcm_add_chmap_ctls+0x119/0x1f0 [snd_pcm] > PGD 0 > Oops: 0000 [#1] PREEMPT SMP NULL checking pcm->pcm instead of pcm fixes this problem. Greetings and thanks for your time, Markus Signed-off-by: Markus Osterhoff <linux-kernel(a)k-raum.org> --- --- sound/pci/hda/hda_codec.c 2015-08-23 20:02:57.281782648 +0200 +++ sound/pci/hda/hda_codec.c.orig 2015-08-23 20:02:49.723783063 +0200 @@ -3172,7 +3172,7 @@ static int add_std_chmaps(struct hda_cod struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (!pcm->pcm || pcm->own_chmap || + if (!pcm || pcm->own_chmap || !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps;

2 5

[alsa-devel] [PATCH v3] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 24 Aug '15

24 Aug '15

Add the capability to use multiple codecs on the same DAI linke where one codec is used for playback and another one is used for capture. Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS microphone for capture. No verification is made that the playback and capture codec setups are compatible in terms of number of TDM slots (where applicable). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V3: Updated commit message regarding the fact that the patch does not consider the slot setup in a TDM system with separate capture and playback codecs, after a comment from Pierre-Louis Bossart. Such a setup was not supported anyway before this patch, so let's handle that situation when and if it arises. V2: Minor code change, otherwise the differences compared to V1 are solely related to comments, in particular considerations given to the potential consequences of the patch. After consideration, it seems to me that the patch as it stands augments the current framework with the functionality indicated in the commit message above, without restricting other current uses. Further functionality could be added, but IMHO that should be done as the need arises, when it can be properly tested in a real-world setup. The patch was created after a discussion on the alsa-devel mailing list as to how best to implement this functionality. (http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html). sound/soc/soc-pcm.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4..08407ba 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -34,6 +34,24 @@ #define DPCM_MAX_BE_USERS 8 +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return codec_stream->rates; +} + /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* + * Skip CODECs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + * At least one CODEC should match, otherwise we should have + * bailed out on a higher level, since there would be no + * CODEC to support the transfer direction in that case. + */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* + * Skip CODECs which don't support the current stream type, + * the idea being that if a CODEC is not used for the currently + * set up transfer direction, it should not need to be + * configured, especially since the configuration used might + * not even be supported by that CODEC. There may be cases + * however where a CODEC needs to be set up although it is + * actually not being used for the transfer, e.g. if a + * capture-only CODEC is acting as an LRCLK and/or BCLK master + * for the DAI link including a playback-only CODEC. + * If this becomes necessary, we will have to augment the + * machine driver setup with information on how to act, so + * we can do the right thing here. + */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [PATCH v2 1/3] ALSA: hdac: Fix size allocation for ext device allocation
by Vinod Koul 23 Aug '15

23 Aug '15

While creating hdac_ext_device, we used hdev for sizeof insteadof edev, which resulted in eventual crash of the system Fix the size here Fixes: a512f5611646 ('ALSA: hdac: add hdac extended device') Reported-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 8544e4fb1cd3..4449d1a99089 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -139,7 +139,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) char name[15]; int ret; - edev = kzalloc(sizeof(*hdev), GFP_KERNEL); + edev = kzalloc(sizeof(*edev), GFP_KERNEL); if (!edev) return -ENOMEM; hdev = &edev->hdac; -- 1.9.1

2 3

[alsa-devel] [PATCH 0/5] ALSA: hdac: More hdac core fixes and updates
by Vinod Koul 23 Aug '15

23 Aug '15

Hi Takashi, We had few more hdac core fixes & enhancements so senidng those. First one is fix for register pointers for spcap and link registers. This is required for SKL driver when we use link registers. rest add new APIs for powering down all links, getting stream from stream tag and direction, enabling SPIB. Last one remove keys as we dont need them for host streams Jeeja KP (5): ALSA: hdac: Fix to read the correct offset of spcap/link register ALSA: hdac: Add snd_hdac_ext_bus_link_power_down_all() ALSA: hdac: Add snd_hdac_get_hdac_stream() ALSA: hdac: Add support to enable SPIB for hdac ext stream ALSA: hdac: Remove the usage of key for host stream include/sound/hda_register.h | 4 +++ include/sound/hdaudio.h | 2 ++ include/sound/hdaudio_ext.h | 11 +++++++ sound/hda/ext/hdac_ext_controller.c | 30 +++++++++++++++--- sound/hda/ext/hdac_ext_stream.c | 62 ++++++++++++++++++++++++++++++++----- sound/hda/hdac_stream.c | 19 ++++++++++++ 6 files changed, 116 insertions(+), 12 deletions(-) Thanks -- ~Vinod

3 10

[alsa-devel] [PATCH v2 0/3] ASoC: topology: update kcontrol ops binding & support TLV bytes controls
by mengdong.lin＠intel.com 22 Aug '15

22 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> History: v2: - let vendor specific kcontrol handlers override standard ones. - Merge handler binding for TLV bytes controls into soc_tplg_kcontrol_bind_io() Mengdong Lin (3): ASoC: topology: Bind vendor specific kcontrol handlers before standard ones ASoC: topology: Reduce arguments of soc_tplg_kcontrol_bind_io() ASoC: topology: Add support for TLV bytes controls include/sound/soc-topology.h | 13 +++++- sound/soc/soc-topology.c | 101 ++++++++++++++++++++++++++++--------------- 2 files changed, 79 insertions(+), 35 deletions(-) -- 1.9.1

2 6

[alsa-devel] [patch] ALSA: hdsp: silence a sprinft() overflow warning
by Dan Carpenter 22 Aug '15

22 Aug '15

card->shortname is a 32 char string so the sprintf() can theoretically overflow. snd_rawmidi_new() can accept strings up to 64 bytes long. I have made the temporay buf[] array 40 bytes long and changed the sprintf() to snprintf(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index c19e021..c1fe1d3 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1526,7 +1526,7 @@ static struct snd_rawmidi_ops snd_hdsp_midi_input = static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int id) { - char buf[32]; + char buf[40]; hdsp->midi[id].id = id; hdsp->midi[id].rmidi = NULL; @@ -1537,7 +1537,7 @@ static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int i hdsp->midi[id].pending = 0; spin_lock_init (&hdsp->midi[id].lock); - sprintf (buf, "%s MIDI %d", card->shortname, id+1); + snprintf(buf, sizeof(buf), "%s MIDI %d", card->shortname, id + 1); if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0) return -1;

2 4

[alsa-devel] [PATCH v2] ALSA: ac97: Switch to dev_pm_ops
by Lars-Peter Clausen 21 Aug '15

21 Aug '15

Convert the ac97_bus from legacy suspend/resume callbacks to dev_pm_ops. Since there isn't anything special to do at the bus level the bus driver does not have to implement any callbacks. The device driver core will automatically pick up and execute the device's PM ops. As there is only a single AC'97 driver implementing suspend and resume, update both the core and driver at the same time to avoid unnecessary code churn. While we are at it also drop the ifdefs around the suspend/resume functions to increase compile test coverage. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This patch touches code in both ALSA and the input subsystem. Ideally this patch will be merged via the ALSA tree. Given that the wm97xx touchscreen driver is not seeing too many changes these days the risk of conflicts should be low. Changes since v1: * Drop ifdefs around suspend/resum functions --- drivers/input/touchscreen/wm97xx-core.c | 13 ++++--------- sound/ac97_bus.c | 26 -------------------------- 2 files changed, 4 insertions(+), 35 deletions(-) diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c index b1ae779..1534e9b 100644 --- a/drivers/input/touchscreen/wm97xx-core.c +++ b/drivers/input/touchscreen/wm97xx-core.c @@ -732,8 +732,7 @@ static int wm97xx_remove(struct device *dev) return 0; } -#ifdef CONFIG_PM -static int wm97xx_suspend(struct device *dev, pm_message_t state) +static int __maybe_unused wm97xx_suspend(struct device *dev) { struct wm97xx *wm = dev_get_drvdata(dev); u16 reg; @@ -765,7 +764,7 @@ static int wm97xx_suspend(struct device *dev, pm_message_t state) return 0; } -static int wm97xx_resume(struct device *dev) +static int __maybe_unused wm97xx_resume(struct device *dev) { struct wm97xx *wm = dev_get_drvdata(dev); @@ -799,10 +798,7 @@ static int wm97xx_resume(struct device *dev) return 0; } -#else -#define wm97xx_suspend NULL -#define wm97xx_resume NULL -#endif +static SIMPLE_DEV_PM_OPS(wm97xx_pm_ops, wm97xx_suspend, wm97xx_resume); /* * Machine specific operations @@ -836,8 +832,7 @@ static struct device_driver wm97xx_driver = { .owner = THIS_MODULE, .probe = wm97xx_probe, .remove = wm97xx_remove, - .suspend = wm97xx_suspend, - .resume = wm97xx_resume, + .pm = &wm97xx_pm_ops, }; static int __init wm97xx_init(void) diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 2b50cbe..57a6dfc 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -27,35 +27,9 @@ static int ac97_bus_match(struct device *dev, struct device_driver *drv) return 1; } -#ifdef CONFIG_PM -static int ac97_bus_suspend(struct device *dev, pm_message_t state) -{ - int ret = 0; - - if (dev->driver && dev->driver->suspend) - ret = dev->driver->suspend(dev, state); - - return ret; -} - -static int ac97_bus_resume(struct device *dev) -{ - int ret = 0; - - if (dev->driver && dev->driver->resume) - ret = dev->driver->resume(dev); - - return ret; -} -#endif /* CONFIG_PM */ - struct bus_type ac97_bus_type = { .name = "ac97", .match = ac97_bus_match, -#ifdef CONFIG_PM - .suspend = ac97_bus_suspend, - .resume = ac97_bus_resume, -#endif /* CONFIG_PM */ }; static int __init ac97_bus_init(void) -- 2.1.4

4 6

[alsa-devel] [PATCH] ASoC: Set missing card owner field
by Axel Lin 21 Aug '15

21 Aug '15

Set the card owner field to prevent the module from being removed from underneath its users. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/blackfin/bfin-eval-adau1x61.c | 1 + sound/soc/intel/boards/byt-max98090.c | 1 + sound/soc/intel/boards/byt-rt5640.c | 1 + sound/soc/intel/boards/bytcr_rt5640.c | 1 + sound/soc/intel/boards/cht_bsw_max98090_ti.c | 1 + sound/soc/intel/boards/cht_bsw_rt5645.c | 2 ++ sound/soc/intel/boards/cht_bsw_rt5672.c | 1 + sound/soc/mediatek/mt8173-max98090.c | 1 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 1 + sound/soc/rockchip/rockchip_max98090.c | 1 + sound/soc/rockchip/rockchip_rt5645.c | 1 + sound/soc/samsung/arndale_rt5631.c | 1 + sound/soc/samsung/snow.c | 1 + 13 files changed, 14 insertions(+) diff --git a/sound/soc/blackfin/bfin-eval-adau1x61.c b/sound/soc/blackfin/bfin-eval-adau1x61.c index 4229f76..fddfe00c 100644 --- a/sound/soc/blackfin/bfin-eval-adau1x61.c +++ b/sound/soc/blackfin/bfin-eval-adau1x61.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link bfin_eval_adau1x61_dai = { static struct snd_soc_card bfin_eval_adau1x61 = { .name = "bfin-eval-adau1x61", + .owner = THIS_MODULE, .driver_name = "eval-adau1x61", .dai_link = &bfin_eval_adau1x61_dai, .num_links = 1, diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 7ab8cc9..d9f81b8 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -126,6 +126,7 @@ static struct snd_soc_dai_link byt_max98090_dais[] = { static struct snd_soc_card byt_max98090_card = { .name = "byt-max98090", + .owner = THIS_MODULE, .dai_link = byt_max98090_dais, .num_links = ARRAY_SIZE(byt_max98090_dais), .dapm_widgets = byt_max98090_widgets, diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index ae89b9b9..de9788a 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -197,6 +197,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { static struct snd_soc_card byt_rt5640_card = { .name = "byt-rt5640", + .owner = THIS_MODULE, .dai_link = byt_rt5640_dais, .num_links = ARRAY_SIZE(byt_rt5640_dais), .dapm_widgets = byt_rt5640_widgets, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7f55d59..c445312 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -185,6 +185,7 @@ static struct snd_soc_dai_link byt_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_byt = { .name = "baytrailcraudio", + .owner = THIS_MODULE, .dai_link = byt_dailink, .num_links = ARRAY_SIZE(byt_dailink), .dapm_widgets = byt_dapm_widgets, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 7810fc0..49f4869 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -275,6 +275,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "chtmax98090", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .aux_dev = &cht_max98090_headset_dev, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf46..7be8461 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -305,6 +305,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_chtrt5645 = { .name = "chtrt5645", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, @@ -317,6 +318,7 @@ static struct snd_soc_card snd_soc_card_chtrt5645 = { static struct snd_soc_card snd_soc_card_chtrt5650 = { .name = "chtrt5650", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5b..23fe040 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -323,6 +323,7 @@ static int cht_resume_post(struct snd_soc_card *card) /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "cherrytrailcraudio", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 39e5660..684e8a7 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -136,6 +136,7 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { static struct snd_soc_card mt8173_max98090_card = { .name = "mt8173-max98090", + .owner = THIS_MODULE, .dai_link = mt8173_max98090_dais, .num_links = ARRAY_SIZE(mt8173_max98090_dais), .controls = mt8173_max98090_controls, diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index b722145..86cf975 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -191,6 +191,7 @@ static struct snd_soc_codec_conf mt8173_rt5650_rt5676_codec_conf[] = { static struct snd_soc_card mt8173_rt5650_rt5676_card = { .name = "mtk-rt5650-rt5676", + .owner = THIS_MODULE, .dai_link = mt8173_rt5650_rt5676_dais, .num_links = ARRAY_SIZE(mt8173_rt5650_rt5676_dais), .codec_conf = mt8173_rt5650_rt5676_codec_conf, diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index acace20..cc26f81 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -147,6 +147,7 @@ static struct snd_soc_dai_link rk_dailink = { static struct snd_soc_card snd_soc_card_rk = { .name = "ROCKCHIP-I2S", + .owner = THIS_MODULE, .dai_link = &rk_dailink, .num_links = 1, .aux_dev = &rk_98090_headset_dev, diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 3c6bb1e..0940279 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -146,6 +146,7 @@ static struct snd_soc_dai_link rk_dailink = { static struct snd_soc_card snd_soc_card_rk = { .name = "I2S-RT5650", + .owner = THIS_MODULE, .dai_link = &rk_dailink, .num_links = 1, .dapm_widgets = rk_dapm_widgets, diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index 9e371eb..ee1fda9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -71,6 +71,7 @@ static struct snd_soc_dai_link arndale_rt5631_dai[] = { static struct snd_soc_card arndale_rt5631 = { .name = "Arndale RT5631", + .owner = THIS_MODULE, .dai_link = arndale_rt5631_dai, .num_links = ARRAY_SIZE(arndale_rt5631_dai), }; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 7651dc9..07ce2cf 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -56,6 +56,7 @@ static int snow_late_probe(struct snd_soc_card *card) static struct snd_soc_card snow_snd = { .name = "Snow-I2S", + .owner = THIS_MODULE, .dai_link = snow_dai, .num_links = ARRAY_SIZE(snow_dai), -- 2.1.0

2 1

Re: [alsa-devel] [PATCH v2] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 21 Aug '15

21 Aug '15

On Fri, 21 Aug 2015, Pierre-Louis Bossart wrote: > On 8/21/15 2:11 AM, Ricard Wanderlof wrote: > > > > On Thu, 20 Aug 2015, Pierre-Louis Bossart wrote: > > > >>> /** > >>> * snd_soc_runtime_activate() - Increment active count for PCM runtime components > >>> * @rtd: ASoC PCM runtime that is activated > >>> @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) > >>> > >>> /* first calculate min/max only for CODECs in the DAI link */ > >>> for (i = 0; i < rtd->num_codecs; i++) { > >>> + > >>> + /* > >>> + * Skip CODECs which don't support the current stream type. > >>> + * Otherwise, since the rate, channel, and format values will > >>> + * zero in that case, we would have no usable settings left, > >>> + * causing the resulting setup to fail. > >>> + * At least one CODEC should match, otherwise we should have > >>> + * bailed out on a higher level, since there would be no > >>> + * CODEC to support the transfer direction in that case. > >>> + */ > >>> + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], > >>> + substream->stream)) > >> > >> Maybe I misunderstood but shouldn't there be some sort of verification > >> that the codecs can use the same number of slots between playback and > >> capture if they share the same LRCLK/FS? e.g. it's not uncommon to have > >> 4 mic capture and 2 ch playback. If the capture and playback is handled > >> by different chips you'd still need to maintain some level of consistency. > > > > You're probably right, although it may be that that sort of thing is > > handled at a higher level, e.g. the machine driver in such a setup would > > configure both codecs to use TDM before we even get to this function, I > > don't know. > > > > I think we'd have to take care of that situation if and when it arises. > > The setup I have here doesn't allow me to test it. > > ok, sounds fine. maybe you could add this to the comments or commit > message then so that this assumption is known. Ok, good point. Will do. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [PATCH] ALSA: hda - Refresh widgets sysfs at probing Haswell+ HDMI codecs
by Takashi Iwai 21 Aug '15

21 Aug '15

Intel Haswell (and later) codec refreshes the widgets tree to expose the whole widget nodes at probing. However, this refresh was missing for sysfs tree. This patch adds the missing piece, as we have now a proper API. Reported-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a97db5fc8a15..df982fc6a872 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2331,6 +2331,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (is_haswell_plus(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); + snd_hdac_refresh_widget_sysfs(&codec->core); } /* For Valleyview/Cherryview, only the display codec is in the display -- 2.5.0

1 0

[alsa-devel] [patch] ALSA: hdsp: silence and underflow warning
by Dan Carpenter 21 Aug '15

21 Aug '15

I believe this probably cannot happen, as the code suggests. There would have to be an kcontrol->index.id which was zero, otherwise this would be prevented in snd_ctl_find_id(). But snd_BUG_ON() is just a WARN() or a no-op so static checkers complain that we keep on going with a negative offset. Let's just handle the error as well as printing a warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index c19e021..468a95c 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2806,7 +2806,8 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); offset = ucontrol->id.index - 1; - snd_BUG_ON(offset < 0); + if (snd_BUG_ON(offset < 0)) + return -EINVAL; switch (hdsp->io_type) { case Digiface:

2 1

[alsa-devel] [PATCH 0/4] ALSA: hdac: add codec driver support
by Vinod Koul 21 Aug '15

21 Aug '15

This patch adds the core framework support for adding hdac extended devices and drivers. We introduce a minimal hdac extended device and driver and APIs for registration This series also ads extended device cleanup API and refresh sysfs tree API Vinod Koul (4): ALSA: hdac: Add API for removing hdac extended device ALSA: hdac: add hdac extended device ALSA: hdac: add extended device driver registration ALSA: hdac: add snd_hdac_refresh_widget_sysfs() include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 60 +++++++++++++++++++++++++++ sound/hda/ext/hdac_ext_bus.c | 98 ++++++++++++++++++++++++++++++++++++++++++-- sound/hda/hdac_device.c | 28 +++++++++++++ 4 files changed, 183 insertions(+), 4 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH] usb: Add native DSD support for Gustard DAC-X20U
by Jurgen Kramer 21 Aug '15

21 Aug '15

This patch adds native DSD support for the Gustard DAC-X20U. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 754e689..00ebc0c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1268,6 +1268,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */ case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ if (fp->altsetting == 3) -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: hda - Shutdown CX20722 on reboot/free to avoid spurious noises
by David Henningsson 21 Aug '15

21 Aug '15

On shutdown/reboot of CX20722, first shut down all EAPDs, then shut down the afg node to D3. Failure to do so can lead to spurious noises from the internal speaker directly after reboot (and before the codec is reinitialized again, i e in BIOS setup or GRUB menus). BugLink: https://bugs.launchpad.net/bugs/1487345 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f788a91..ca03c40 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -200,12 +200,33 @@ static int cx_auto_init(struct hda_codec *codec) return 0; } -#define cx_auto_free snd_hda_gen_free +static void cx_auto_reboot_notify(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + + if (codec->core.vendor_id != 0x14f150f2) + return; + + /* Turn the CX20722 codec into D3 to avoid spurious noises + from the internal speaker during (and after) reboot */ + cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); + + snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} + +static void cx_auto_free(struct hda_codec *codec) +{ + cx_auto_reboot_notify(codec); + snd_hda_gen_free(codec); +} static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = cx_auto_init, + .reboot_notify = cx_auto_reboot_notify, .free = cx_auto_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM -- 1.9.1

2 3

[alsa-devel] [PATCH 0/3] Add ASoC support for AMD APUs [v2]
by Alex Deucher 21 Aug '15

21 Aug '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create an mfd cell which we can hang the i2s codec on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review There was a question about the code and the module licensing. IANAL, but this follows the same pattern that most if not all drm drivers use. There was also a comment about the patch changelog. I think this is more of a subsystem preference kind of thing. Generally in the drm we include the patch changelog and versions in the commit messages so we have a history of the patch if we need to reference it in the future. If you have a strong preference for the audio patch, we can drop that. Patch 1 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream2 Thanks, Alex Chunming Zhou (1): drm/amd: add ACP 2.x register headers Maruthi Bayyavarapu (1): drm/amd: add ACP driver support [v6] Maruthi Srinivas Bayyavarapu (1): ASoC: AMD: add AMD ASoC ACP-I2S driver [v4] drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 1133 ++++++++++ drivers/gpu/drm/amd/acp/acp_hw.h | 91 + drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 5 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 208 ++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 40 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + .../gpu/drm/amd/include/asic_reg/acp/acp_2_1_d.h | 437 ++++ .../drm/amd/include/asic_reg/acp/acp_2_1_enum.h | 1198 ++++++++++ .../drm/amd/include/asic_reg/acp/acp_2_1_sh_mask.h | 1568 +++++++++++++ .../gpu/drm/amd/include/asic_reg/acp/acp_2_2_d.h | 609 ++++++ .../drm/amd/include/asic_reg/acp/acp_2_2_enum.h | 1068 +++++++++ .../drm/amd/include/asic_reg/acp/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++ .../gpu/drm/amd/include/asic_reg/acp/acp_2_3_d.h | 582 +++++ .../drm/amd/include/asic_reg/acp/acp_2_3_enum.h | 1079 +++++++++ .../drm/amd/include/asic_reg/acp/acp_2_3_sh_mask.h | 2192 +++++++++++++++++++ include/linux/mfd/amd_acp.h | 211 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 5 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 676 ++++++ 27 files changed, 13494 insertions(+), 1 deletion(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.h create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_sh_mask.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c -- 1.8.3.1

2 10

[alsa-devel] [PATCH] ASoC: Samsung: Remove redundant arndale_audio_remove
by Vaishali Thakkar 21 Aug '15

21 Aug '15

There is no use of snd_soc_unregister_card in remove function as devm_snd_soc_register_card in probe function automatically handles it. So, remove use of snd_soc_unregister_card and with this change remove arndale_audio_remove as it is now redundant. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/samsung/arndale_rt5631.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index 8bf2e2c..9e371eb 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -116,15 +116,6 @@ static int arndale_audio_probe(struct platform_device *pdev) return ret; } -static int arndale_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { { .compatible = "samsung,arndale-rt5631", }, { .compatible = "samsung,arndale-alc5631", }, @@ -139,7 +130,6 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), }, .probe = arndale_audio_probe, - .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); -- 1.9.1

3 2

[alsa-devel] [PATCH V6] ASoC: AMD: add AMD ASoC ACP-I2S driver
by Alex Deucher 21 Aug '15

21 Aug '15

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> ACP IP block consists of dedicated DMA and I2S blocks. The PCM driver provides the DMA and CPU DAI components to ALSA core. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Reviewed-by: Murali Krishna Vemuri <murali-krishna.vemuri(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- v2: - squash in Kconfig fix v3: - squash in the following patches: ASoC: AMD: add suspend/resume for ACP PCM driver ASoC: AMD: add ACP PCM driver runtime PM - Keep the Makefile sorted - Fix the Kconfig whitespace - drop RT286 changes to rework it using ACPI as a separate patch. The gpio stuff we need to handle this with ACPI was not added to the kernel until 4.2. This patch was originally developed much earlier. - put headers in include - remove static structure initialization - fix component name - remove casts from void - fix handling of combinations of play_intr and capture_intr - fix coding style with {} - switch to dev_ prints - remove substream checks handled by subsystem - fix whitespace in comments - pass return code directly on error from acp_dev->dma_start - drop references to designware - fix codeing style (STATUS_SUCCESS) - drop left over debugging messages - drop stuff that is already set in the dai_link - convert from using the amd_gnb_bus to an mfd cell v4: - change MODULE_ALIAS to platform:acp-i2s-audio v5: - failed attempt at a patch revision changelog v6: - respin the patch with a proper patch change log The module licensing follows the model used in the drm drivers. AMD prefers to contribute code under an MIT/X11 license. IANAL, but "GPL and additional rights" as the MODULE_LICENSE seems to be compatible with that and is exactly what most of the drm drivers do. sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 5 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 676 ++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 686 insertions(+) create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 2ae9619..1b9ce31 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/adi/Kconfig" +source "sound/soc/amd/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index e189903..a6cbb99 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,7 @@ obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += adi/ +obj-$(CONFIG_SND_SOC) += amd/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += bcm/ diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig new file mode 100644 index 0000000..ce39979 --- /dev/null +++ b/sound/soc/amd/Kconfig @@ -0,0 +1,5 @@ +config SND_SOC_AMD_ACP + tristate "AMD Audio Coprocessor support" + depends on MFD_CORE + help + This option enables ACP support (DMA,I2S) on AMD platforms. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile new file mode 100644 index 0000000..1a66ec0 --- /dev/null +++ b/sound/soc/amd/Makefile @@ -0,0 +1,3 @@ +snd-soc-acp-pcm-objs := acp-pcm-dma.o + +obj-$(CONFIG_SND_SOC_AMD_ACP) += snd-soc-acp-pcm.o diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c new file mode 100644 index 0000000..18cc60c --- /dev/null +++ b/sound/soc/amd/acp-pcm-dma.c @@ -0,0 +1,676 @@ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2014-2015 Advanced Micro Devices, Inc. + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR + * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, + * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + * + */ + +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/pci.h> +#include <linux/pm_runtime.h> +#include <linux/mfd/amd_acp.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 2 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 1024 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 2 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 1024 + +#define NUM_DSCRS_PER_CHANNEL 2 + +#define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) +#define MIN_BUFFER MAX_BUFFER + +#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */ +#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ +#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ +#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ + + +static const struct snd_pcm_hardware acp_pcm_hardware_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + /* formats,rates,channels based on i2s doc. */ + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +static const struct snd_pcm_hardware acp_pcm_hardware_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + /* formats,rates,channels based on i2s doc. */ + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + +struct audio_drv_data { + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + struct amd_acp_device *acp_dev; + struct acp_irq_prv *iprv; +}; + +static const struct snd_soc_component_driver dw_i2s_component = { + .name = "dw-i2s", +}; + +static void acp_pcm_period_elapsed(struct device *dev, u16 play_intr, + u16 capture_intr) +{ + struct snd_pcm_substream *substream; + struct audio_drv_data *irq_data = dev_get_drvdata(dev); + + /* Inform ALSA about the period elapsed (one out of two periods) */ + if (play_intr) + substream = irq_data->play_stream; + else + substream = irq_data->capture_stream; + + if (substream->runtime && snd_pcm_running(substream)) + snd_pcm_period_elapsed(substream); +} + +static int acp_dma_open(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_drv_data *intr_data = dev_get_drvdata(prtd->platform->dev); + + struct audio_substream_data *adata = + kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); + if (adata == NULL) + return -ENOMEM; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp_pcm_hardware_playback; + else + runtime->hw = acp_pcm_hardware_capture; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(prtd->platform->dev, "set integer constraint failed\n"); + return ret; + } + + adata->acp_dev = intr_data->acp_dev; + runtime->private_data = adata; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + intr_data->play_stream = substream; + else + intr_data->capture_stream = substream; + + return 0; +} + +static int acp_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int status; + uint64_t size; + struct snd_dma_buffer *dma_buffer; + struct page *pg; + u16 num_of_pages; + struct snd_pcm_runtime *runtime; + struct audio_substream_data *rtd; + struct amd_acp_device *acp_dev; + + dma_buffer = &substream->dma_buffer; + + runtime = substream->runtime; + rtd = runtime->private_data; + + if (WARN_ON(!rtd)) + return -EINVAL; + acp_dev = rtd->acp_dev; + + size = params_buffer_bytes(params); + status = snd_pcm_lib_malloc_pages(substream, size); + if (status < 0) + return status; + + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + pg = virt_to_page(substream->dma_buffer.area); + + if (NULL != pg) { + /* Save for runtime private data */ + rtd->pg = pg; + rtd->order = get_order(size); + + /* Let ACP know the Allocated memory */ + num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; + + /* Fill the page table entries in ACP SRAM */ + rtd->pg = pg; + rtd->size = size; + rtd->num_of_pages = num_of_pages; + rtd->direction = substream->stream; + + acp_dev->config_dma(acp_dev, rtd); + status = 0; + } else { + status = -ENOMEM; + } + return status; +} + +static int acp_dma_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) +{ + u32 pos = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + pos = acp_dev->update_dma_pointer(acp_dev, substream->stream, + frames_to_bytes(runtime, runtime->period_size)); + return bytes_to_frames(runtime, pos); + +} + +static int acp_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return snd_pcm_lib_default_mmap(substream, vma); +} + +static int acp_dma_prepare(struct snd_pcm_substream *substream) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + acp_dev->config_dma_channel(acp_dev, SYSRAM_TO_ACP_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH12, + NUM_DSCRS_PER_CHANNEL, 0); + acp_dev->config_dma_channel(acp_dev, ACP_TO_I2S_DMA_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH13, + NUM_DSCRS_PER_CHANNEL, 0); + /* Fill ACP SRAM (2 periods) with zeros from System RAM + * which is zero-ed in hw_params */ + ret = acp_dev->dma_start(rtd->acp_dev, + SYSRAM_TO_ACP_CH_NUM, false); + if (ret < 0) + ret = -EFAULT; + + /* Now configure DMA to transfer only first half of System RAM + * buffer before playback is triggered. This will overwrite + * zero-ed second half of SRAM buffer */ + acp_dev->config_dma_channel(acp_dev, SYSRAM_TO_ACP_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH12, + 1, 0); + } else { + acp_dev->config_dma_channel(acp_dev, ACP_TO_SYSRAM_CH_NUM, + CAPTURE_START_DMA_DESCR_CH14, + NUM_DSCRS_PER_CHANNEL, 0); + acp_dev->config_dma_channel(acp_dev, I2S_TO_ACP_DMA_CH_NUM, + CAPTURE_START_DMA_DESCR_CH15, + NUM_DSCRS_PER_CHANNEL, 0); + } + return 0; +} + +static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + int ret; + + if (!rtd) + return -EINVAL; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dev->dma_start(rtd->acp_dev, + SYSRAM_TO_ACP_CH_NUM, false); + if (ret < 0) + return -EFAULT; + acp_dev->prebuffer_audio(rtd->acp_dev); + + ret = acp_dev->dma_start(acp_dev, + ACP_TO_I2S_DMA_CH_NUM, true); + } else { + ret = acp_dev->dma_start(acp_dev, + I2S_TO_ACP_DMA_CH_NUM, true); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dev->dma_stop(acp_dev, SYSRAM_TO_ACP_CH_NUM); + if (0 == ret) + ret = acp_dev->dma_stop(acp_dev, + ACP_TO_I2S_DMA_CH_NUM); + } else { + ret = acp_dev->dma_stop(acp_dev, I2S_TO_ACP_DMA_CH_NUM); + if (0 == ret) + ret = acp_dev->dma_stop(acp_dev, + ACP_TO_SYSRAM_CH_NUM); + } + break; + default: + ret = -EINVAL; + + } + return ret; +} + +static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_pcm *pcm; + + pcm = rtd->pcm; + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + NULL, MIN_BUFFER, MAX_BUFFER); + return ret; +} + +static int acp_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + + kfree(rtd); + + pm_runtime_mark_last_busy(prtd->platform->dev); + return 0; +} + +static int acp_dai_i2s_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + struct device *dev = prtd->platform->dev; + u32 chan_nr; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + rtd->xfer_resolution = 0x02; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + rtd->xfer_resolution = 0x04; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + rtd->xfer_resolution = 0x05; + break; + + default: + dev_err(dev, "unsuppted PCM fmt : %d\n", params_format(params)); + return -EINVAL; + } + + chan_nr = params_channels(params); + + switch (chan_nr) { + case EIGHT_CHANNEL_SUPPORT: + rtd->ch_reg = 3; + break; + case SIX_CHANNEL_SUPPORT: + rtd->ch_reg = 2; + break; + case FOUR_CHANNEL_SUPPORT: + rtd->ch_reg = 1; + break; + case TWO_CHANNEL_SUPPORT: + rtd->ch_reg = 0; + break; + default: + dev_err(dev, "channel not supported : %d\n", chan_nr); + return -EINVAL; + } + + rtd->direction = substream->stream; + + acp_dev->config_i2s(acp_dev, rtd); + + return 0; +} + +static int acp_dai_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + acp_dev->i2s_start(acp_dev, substream->stream); + ret = 0; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + acp_dev->i2s_stop(acp_dev, substream->stream); + ret = 0; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int acp_dai_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + acp_dev->i2s_reset(acp_dev, substream->stream); + return 0; +} + +static struct snd_pcm_ops acp_dma_ops = { + .open = acp_dma_open, + .close = acp_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = acp_dma_hw_params, + .hw_free = acp_dma_hw_free, + .trigger = acp_dma_trigger, + .pointer = acp_dma_pointer, + .mmap = acp_dma_mmap, + .prepare = acp_dma_prepare, +}; + +static struct snd_soc_dai_ops acp_dai_i2s_ops = { + .prepare = acp_dai_i2s_prepare, + .hw_params = acp_dai_i2s_hwparams, + .trigger = acp_dai_i2s_trigger, +}; + +static struct snd_soc_dai_driver i2s_dai_driver = { + .playback = { + .stream_name = "I2S Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .stream_name = "I2S Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp_dai_i2s_ops, +}; + +static struct snd_soc_platform_driver acp_asoc_platform = { + .ops = &acp_dma_ops, + .pcm_new = acp_dma_new, +}; + +static int acp_alsa_register(struct device *dev, struct amd_acp_device *acp_dev, + struct platform_device *pdev) +{ + int status; + + status = snd_soc_register_platform(dev, &acp_asoc_platform); + if (0 != status) { + dev_err(dev, "Unable to register ALSA platform device\n"); + goto exit_platform; + } else { + status = snd_soc_register_component(dev, + &dw_i2s_component, + &i2s_dai_driver, 1); + + if (0 != status) { + dev_err(dev, "Unable to register i2s dai\n"); + goto exit_dai; + } else { + dev_info(dev, "ACP device registered with ALSA\n"); + return status; + } + } + +exit_dai: + snd_soc_unregister_platform(dev); +exit_platform: + acp_dev->fini(acp_dev); + return status; +} + +static int acp_audio_probe(struct platform_device *pdev) +{ + int status; + struct audio_drv_data *audio_drv_data; + struct amd_acp_device *acp_dev = dev_get_platdata(&pdev->dev); + + audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), + GFP_KERNEL); + if (audio_drv_data == NULL) + return -ENOMEM; + + audio_drv_data->iprv = devm_kzalloc(&pdev->dev, + sizeof(struct acp_irq_prv), + GFP_KERNEL); + if (audio_drv_data->iprv == NULL) + return -ENOMEM; + + /* The following members gets populated in device 'open' + * function. Till then interrupts are disabled in 'acp_hw_init' + * and device doesn't generate any interrupts. + */ + + audio_drv_data->play_stream = NULL; + audio_drv_data->capture_stream = NULL; + audio_drv_data->acp_dev = acp_dev; + + audio_drv_data->iprv->dev = &pdev->dev; + audio_drv_data->iprv->acp_dev = acp_dev; + audio_drv_data->iprv->set_elapsed = acp_pcm_period_elapsed; + + dev_set_drvdata(&pdev->dev, audio_drv_data); + + /* Initialize the ACP */ + status = acp_dev->init(acp_dev, audio_drv_data->iprv); + + if (0 == status) + status = acp_alsa_register(&pdev->dev, acp_dev, pdev); + else + dev_err(&pdev->dev, "ACP initialization Failed\n"); + + pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); + + return status; +} + +static int acp_audio_remove(struct platform_device *pdev) +{ + struct amd_acp_device *acp_dev = dev_get_platdata(&pdev->dev); + + snd_soc_unregister_component(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); + + acp_dev->fini(acp_dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static int acp_pcm_suspend(struct device *dev) +{ + bool pm_rts; + struct audio_drv_data *adata = dev_get_drvdata(dev); + + pm_rts = pm_runtime_status_suspended(dev); + if (pm_rts == false) + adata->acp_dev->fini(adata->acp_dev); + + return 0; +} + +static int acp_pcm_resume(struct device *dev) +{ + bool pm_rts; + struct snd_pcm_substream *pstream, *cstream; + struct snd_pcm_runtime *prtd, *crtd; + struct audio_substream_data *rtd; + struct audio_drv_data *adata = dev_get_drvdata(dev); + + pm_rts = pm_runtime_status_suspended(dev); + if (pm_rts == true) { + /* Resumed from system wide suspend and there is + * no pending audio activity to resume. */ + pm_runtime_disable(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + + goto out; + } + + pstream = adata->play_stream; + prtd = pstream ? pstream->runtime : NULL; + if (prtd != NULL) { + /* Resume playback stream from a suspended state */ + rtd = prtd->private_data; + + adata->acp_dev->config_dma(adata->acp_dev, rtd); + adata->acp_dev->config_i2s(adata->acp_dev, rtd); + } + + cstream = adata->capture_stream; + crtd = cstream ? cstream->runtime : NULL; + if (crtd != NULL) { + /* Resume capture stream from a suspended state */ + rtd = crtd->private_data; + + adata->acp_dev->config_dma(adata->acp_dev, rtd); + adata->acp_dev->config_i2s(adata->acp_dev, rtd); + } +out: + return 0; +} + +static int acp_pcm_runtime_suspend(struct device *dev) +{ + struct audio_drv_data *adata = dev_get_drvdata(dev); + + adata->acp_dev->acp_suspend(adata->acp_dev); + return 0; +} + +static int acp_pcm_runtime_resume(struct device *dev) +{ + struct audio_drv_data *adata = dev_get_drvdata(dev); + + adata->acp_dev->acp_resume(adata->acp_dev); + return 0; +} + +static const struct dev_pm_ops acp_pm_ops = { + .suspend = acp_pcm_suspend, + .resume = acp_pcm_resume, + .runtime_suspend = acp_pcm_runtime_suspend, + .runtime_resume = acp_pcm_runtime_resume, +}; + +static struct platform_driver acp_dma_driver = { + .probe = acp_audio_probe, + .remove = acp_audio_remove, + .driver = { + .name = "acp-i2s-audio", + .pm = &acp_pm_ops, + }, +}; + +module_platform_driver(acp_dma_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP PCM Driver"); +MODULE_LICENSE("GPL and additional rights"); +MODULE_ALIAS("platform:acp-i2s-audio"); -- 1.8.3.1

1 0

[alsa-devel] Boss Micro BR-80 not workin
by Karsten Heymann 21 Aug '15

21 Aug '15

Hi, I'm trying to get my Boss Micro BR-80 (id 0582:0130) usb device to work. The device is detected, but with my limited knowledge of the alsa basics i could not get any audio in or out of it, pulse audio and jack don't seem to work too. I'm not a kernel hacker, but hope to be helpful with providing output, testing patches and if necessary providing ssh access to a system with this device attached to a more knowledgeable developer. There used to be dedicated quirk code in sound/usb/quirk-table.h which was removed in http://thread.gmane.org/gmane.linux.alsa.devel/109660/focus=111768 during a roland module update. Older postings on the net indicate that the br-80 seem to have worked in the past. My questions: * which command outputs do you need to debug this? * should compiling an older version of the snd-usb module for a current kernel work? Regards, Karsten

1 0

[alsa-devel] [PATCH] ASoC: wm8994: fix add dynamic path error
by Zidan Wang 20 Aug '15

20 Aug '15

If there don't exist dynamic sink or source widget, it will failed to add dynamic path. "AIF3ADCDAT" is snd_soc_dapm_aif_out, can't be dynamic sink widget. So change the audio route to fix this issue. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8994.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 962e1d3..2ccbb32 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1942,14 +1942,16 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1L" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1R" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2L" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2R" }, - { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCL" }, - { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCR" }, - { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, - { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC1L" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC1R" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC2L" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC2R" }, + { "AIF3ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, + { "AIF3ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, + { "AIF3ADC Mux", "AIF2DACDAT", "AIF2DACL" }, + { "AIF3ADC Mux", "AIF2DACDAT", "AIF2DACR" }, + + { "AIF3ADCDAT", NULL, "AIF3ADC Mux" }, /* Loopback */ { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: ac97: Switch to dev_pm_ops
by Lars-Peter Clausen 20 Aug '15

20 Aug '15

Convert the ac97_bus from legacy suspend/resume callbacks to dev_pm_ops. Since there isn't anything special to do at the bus level the bus driver does not have to implement any callbacks. The device driver core will automatically pick up and execute the device's PM ops. As there is only a single AC'97 driver implementing suspend and resume, update both the core and driver at the same time to avoid unnecessary code churn. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This patch touches code in both ALSA and the input subsystem. Ideally this patch will be merged via the ALSA tree. Given that the wm97xx touchscreen driver is not seeing too many changes these days the risk of conflicts should be low. --- drivers/input/touchscreen/wm97xx-core.c | 13 +++++++------ sound/ac97_bus.c | 26 -------------------------- 2 files changed, 7 insertions(+), 32 deletions(-) diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c index b1ae779..1d11fd8 100644 --- a/drivers/input/touchscreen/wm97xx-core.c +++ b/drivers/input/touchscreen/wm97xx-core.c @@ -732,8 +732,8 @@ static int wm97xx_remove(struct device *dev) return 0; } -#ifdef CONFIG_PM -static int wm97xx_suspend(struct device *dev, pm_message_t state) +#ifdef CONFIG_PM_SLEEP +static int wm97xx_suspend(struct device *dev) { struct wm97xx *wm = dev_get_drvdata(dev); u16 reg; @@ -799,9 +799,11 @@ static int wm97xx_resume(struct device *dev) return 0; } +static SIMPLE_DEV_PM_OPS(wm97xx_pm_ops, wm97xx_suspend, wm97xx_resume); +#define WM97XX_PM_OPS (&wm97xx_pm_ops) + #else -#define wm97xx_suspend NULL -#define wm97xx_resume NULL +#define WM97XX_PM_OPS NULL #endif /* @@ -836,8 +838,7 @@ static struct device_driver wm97xx_driver = { .owner = THIS_MODULE, .probe = wm97xx_probe, .remove = wm97xx_remove, - .suspend = wm97xx_suspend, - .resume = wm97xx_resume, + .pm = WM97XX_PM_OPS, }; static int __init wm97xx_init(void) diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 2b50cbe..57a6dfc 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -27,35 +27,9 @@ static int ac97_bus_match(struct device *dev, struct device_driver *drv) return 1; } -#ifdef CONFIG_PM -static int ac97_bus_suspend(struct device *dev, pm_message_t state) -{ - int ret = 0; - - if (dev->driver && dev->driver->suspend) - ret = dev->driver->suspend(dev, state); - - return ret; -} - -static int ac97_bus_resume(struct device *dev) -{ - int ret = 0; - - if (dev->driver && dev->driver->resume) - ret = dev->driver->resume(dev); - - return ret; -} -#endif /* CONFIG_PM */ - struct bus_type ac97_bus_type = { .name = "ac97", .match = ac97_bus_match, -#ifdef CONFIG_PM - .suspend = ac97_bus_suspend, - .resume = ac97_bus_resume, -#endif /* CONFIG_PM */ }; static int __init ac97_bus_init(void) -- 2.1.4

3 10

[alsa-devel] [GIT PULL] ASoC update for v4.2
by Mark Brown 19 Aug '15

19 Aug '15

The following changes since commit bc0195aad0daa2ad5b0d76cce22b167bc3435590: Linux 4.2-rc2 (2015-07-12 15:10:30 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.2-disable-topology for you to fetch changes up to 8c90503bf246bebb48caa5590d41df755ba08550: ASoC: topology: Disable use from userspace (2015-08-18 09:33:21 -0700) ---------------------------------------------------------------- ASoC: Disable topology support for v4.2 The topology code merged in the v4.2 merge window introduced a new ABI which was believed to be suitable for use but subsequently additional work by the developers of this feature have revealed some problems that need to be addressed. In order to allow this to be done without having to support the initial ABI add Kconfig to disable the build and also add some #error statements to the UAPI header so users can't use them. ---------------------------------------------------------------- Mark Brown (2): ASoC: topology: Add Kconfig option for topology ASoC: topology: Disable use from userspace include/sound/soc-topology.h | 12 ++++++++++++ include/uapi/sound/asoc.h | 6 ++++++ sound/soc/Kconfig | 3 +++ sound/soc/Makefile | 3 +++ 4 files changed, 24 insertions(+)

2 1

[alsa-devel] [PATCH v3] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 19 Aug '15

19 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V3: Fixed MODULE_LICENSE string. V2: Update after comments from Lars-Peter: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. I've left the checks in dai_set_fmt, as (IIUC) there currently doesn't seem to be a way for the ALSA framework to handle format constraints using the codec configuration struct. .../devicetree/bindings/sound/ics43432.txt | 17 ++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 101 ++++++++++++++++++++ 5 files changed, 125 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..fe0af4a --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,101 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPLv2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] ASoC: rockchip: i2s: Adjust devm usage
by Vaishali Thakkar 19 Aug '15

19 Aug '15

Remove use of snd_soc_unregister_component in remove function as devm_snd_soc_register_component in probe function automatically handles it. Also, convert call of snd_dmaengine_pcm_register to managed resource function devm_snd_dmaengine_pcm_register and remove usage of snd_dmaengine_pcm_unregister in probe and remove functions. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index acb5be5..b936102 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -483,16 +483,14 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - goto err_pcm_register; + return ret; } return 0; -err_pcm_register: - snd_dmaengine_pcm_unregister(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) i2s_runtime_suspend(&pdev->dev); @@ -512,8 +510,6 @@ static int rockchip_i2s_remove(struct platform_device *pdev) clk_disable_unprepare(i2s->mclk); clk_disable_unprepare(i2s->hclk); - snd_dmaengine_pcm_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: arizona: Fix gain settings of FLL in free-run mode
by Charles Keepax 19 Aug '15

19 Aug '15

From: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> The wrong register was used to set the gain of ref loop, when changing the FLL output on an active FLL. This patch corrects the offset of the gain register. Signed-off-by: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e759d39..42b2021 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2058,7 +2058,7 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (already_enabled) { /* Facilitate smooth refclk across the transition */ - regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7, + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, ARIZONA_FLL1_GAIN_MASK, 0); regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, -- 1.7.2.5

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: Recurse before saving terminal properties
by Julian Scheel 19 Aug '15

19 Aug '15

The input terminal parser recurses into the referenced clock entity to verify it is existant and thus the terminal descriptor is valid. The actual property values of the term instance which is initially parsed must not be overriden by the recursion. For this to work the term properties have to be assigned after recursing into the referenced clock entity descriptors. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- Changes in v2: - Store term->id after recursion as well, as it gets overriden in the recursion - Add comments explaining that the recursion is only for validation of the descriptor validity --- sound/usb/mixer.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81055d3..c50790c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -731,15 +731,21 @@ static int check_input_term(struct mixer_build *state, int id, term->name = d->iTerminal; } else { /* UAC_VERSION_2 */ struct uac2_input_terminal_descriptor *d = p1; - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le32_to_cpu(d->bmChannelConfig); - term->name = d->iTerminal; - /* call recursively to get the clock selectors */ + /* call recursively to verify that the + * referenced clock entity is valid */ err = check_input_term(state, d->bCSourceID, term); if (err < 0) return err; + + /* save input term properties after recursion, + * to ensure they are not overriden by the + * recursion calls */ + term->id = id; + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; } return 0; case UAC_FEATURE_UNIT: { -- 2.5.0

3 2

[alsa-devel] [PATCH] ASoc: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 19 Aug '15

19 Aug '15

(Referring to this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html) This patch adds the capability to use multiple codecs on the same DAI link where one codec is used for playback and another one is used for capture. As noted in the thread, there might be more places where the new snd_soc_dai_stream_valid() function is needed, however, these two were sufficient for our use case (SSM2518 playback and ICS-43432 capture). Signed-off-by: Ricard Wanderlof <ricardw(a)xis.com> --- sound/soc/soc-pcm.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e..df1f261 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -35,6 +35,32 @@ #define DPCM_MAX_BE_USERS 8 /** + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * @dai: The DAI to check for stream support + * @stream: The stream type (playback / capture) to check for + * + * Checks if the DAI in question supports the indicated stream type. + * Needed to skip over codecs when iterating over + * snd_soc_pcm_runtime->codec_dais in order to skip codecs that are playback- + * or capture-only, which would otherwise inhibit capture or playback, + * respectively, to the other codec in a multiple codec / common I2S bus + * environment. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return !!codec_stream->rates; +} + +/** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated * @stream: Direction of the PCM stream @@ -371,6 +397,12 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* Skip codecs which don't support the current stream type. */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* Skip codecs which don't support the current stream type. */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 4

[alsa-devel] [PATCH v3 0/4] Add SPDIF support for rockchip
by Sjoerd Simons 19 Aug '15

19 Aug '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board. Changes in v3: - Fix typos in commit message - Resent patches to a more complete list of maintainers. Changes in v2: - Remove platform: module alias as it was unused - Call MODULE_DEVICE_TABLE(of, ) right after the of match table - use rk_spdif as a prefix consistenly throughout the driver - Check return code of regmap_update and bubble it up - Sort the spdif node properties - Drop the 0x prefix from the node name - Rename the spdif@ node to sound@ Sjoerd Simons (4): ASoC: dt-bindings: add rockchip tranceiver bindings ASoC: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock .../devicetree/bindings/sound/rockchip-spdif.txt | 41 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 + arch/arm/boot/dts/rk3188.dtsi | 22 ++ sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdif.c | 381 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 7 files changed, 536 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.5.0

3 9

[alsa-devel] [PATCH] ALSA: hda - Add dock support for Thinkpad W541 (17aa:2211)
by Rick Sherman 19 Aug '15

19 Aug '15

Tested with W541 and Ultra Dock 170w Signed-off-by: Rick Sherman <rick(a)shermdog.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b9847a..39a0342 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5291,6 +5291,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.4.3

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix runtime PM unbalance
by Takashi Iwai 19 Aug '15

19 Aug '15

The fix for deadlock in PM in commit [1ee23fe07ee8: ALSA: usb-audio: Fix deadlocks at resuming] introduced a new check of in_pm flag. However, the brainless patch author evaluated it in a wrong way (logical AND instead of logical OR), thus usb_autopm_get_interface() is never called, leading to unbalance of runtime PM refcount. This patch fixes it by correcting the logic. Reported-by: Hans Yang <hansy(a)nvidia.com> Fixes: 1ee23fe07ee8 ('ALSA: usb-audio: Fix deadlocks at resuming') Cc: <stable(a)vger.kernel.org> [v3.15+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 1fab9778807a..0450593980fd 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -638,7 +638,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip) int err = -ENODEV; down_read(&chip->shutdown_rwsem); - if (chip->probing && chip->in_pm) + if (chip->probing || chip->in_pm) err = 0; else if (!chip->shutdown) err = usb_autopm_get_interface(chip->pm_intf); -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: wm8741: Drop misleading readable_reg callback implementation
by Axel Lin 18 Aug '15

18 Aug '15

regmap_readable() returns false if map->format.format_write is set. For .reg_bits = 7, .val_bits = 9, setting, map->format.format_write = regmap_format_7_9_write; Even current code has implemented map->readable_reg, regmap_readable() still returns false anyway. Thus drop the misleading readable_reg callback implementation. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8741.c | 21 --------------------- 1 file changed, 21 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9723e44..de42c03 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -61,25 +61,6 @@ static const struct reg_default wm8741_reg_defaults[] = { { 32, 0x0002 }, /* R32 - ADDITONAL_CONTROL_1 */ }; -static bool wm8741_readable(struct device *dev, unsigned int reg) -{ - switch (reg) { - case WM8741_DACLLSB_ATTENUATION: - case WM8741_DACLMSB_ATTENUATION: - case WM8741_DACRLSB_ATTENUATION: - case WM8741_DACRMSB_ATTENUATION: - case WM8741_VOLUME_CONTROL: - case WM8741_FORMAT_CONTROL: - case WM8741_FILTER_CONTROL: - case WM8741_MODE_CONTROL_1: - case WM8741_MODE_CONTROL_2: - case WM8741_ADDITIONAL_CONTROL_1: - return true; - default: - return false; - } -} - static int wm8741_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8741_RESET, 0); @@ -541,8 +522,6 @@ static const struct regmap_config wm8741_regmap = { .reg_defaults = wm8741_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8741_reg_defaults), .cache_type = REGCACHE_RBTREE, - - .readable_reg = wm8741_readable, }; static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: topology: Disable use from userspace
by Mark Brown 18 Aug '15

18 Aug '15

Since the topology API is still in sufficient flux for changes to be identified disable the use of the userspace ABI by adding #error statements to the code, ensuring that nobody relies on the headers as currently defined. It is expected that this change will be reverted for v4.3. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/asoc.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 51b8066..247c50b 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -18,6 +18,12 @@ #include <linux/types.h> #include <sound/asound.h> +#ifndef __KERNEL__ +#error This API is an early revision and not enabled in the current +#error kernel release, it will be enabled in a future kernel version +#error with incompatible changes to what is here. +#endif + /* * Maximum number of channels topology kcontrol can represent. */ -- 2.5.0

1 1

[alsa-devel] [PATCH] ASoC: tegra: Convert to managed resources
by Vaishali Thakkar 18 Aug '15

18 Aug '15

Use managed resource functions devm_clk_put and devm_snd_soc_register_component to simplify error handling. To be compatible with the change various gotos are replaced with direct returns, and unneeded labels are dropped. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/tegra/tegra20_spdif.c | 37 +++++++++++++------------------------ 1 file changed, 13 insertions(+), 24 deletions(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 9141477..f69b2e4 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -273,45 +273,40 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) GFP_KERNEL); if (!spdif) { dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } dev_set_drvdata(&pdev->dev, spdif); - spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + spdif->clk_spdif_out = devm_clk_get(&pdev->dev, "spdif_out"); if (IS_ERR(spdif->clk_spdif_out)) { pr_err("Can't retrieve spdif clock\n"); ret = PTR_ERR(spdif->clk_spdif_out); - goto err; + return ret; } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; + return -ENODEV; } dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; + return -ENODEV; } memregion = devm_request_mem_region(&pdev->dev, mem->start, resource_size(mem), DRV_NAME); if (!memregion) { dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; + return -EBUSY; } regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!regs) { dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_clk_put; + return -ENOMEM; } spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs, @@ -319,7 +314,7 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); ret = PTR_ERR(spdif->regmap); - goto err_clk_put; + return ret; } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; @@ -334,8 +329,9 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = snd_soc_register_component(&pdev->dev, &tegra20_spdif_component, - &tegra20_spdif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &tegra20_spdif_component, + &tegra20_spdif_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -345,21 +341,17 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_component; + return ret; } return 0; -err_unregister_component: - snd_soc_unregister_component(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_spdif_runtime_suspend(&pdev->dev); err_pm_disable: pm_runtime_disable(&pdev->dev); -err_clk_put: - clk_put(spdif->clk_spdif_out); -err: + return ret; } @@ -372,9 +364,6 @@ static int tegra20_spdif_platform_remove(struct platform_device *pdev) tegra20_spdif_runtime_suspend(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - - clk_put(spdif->clk_spdif_out); return 0; } -- 1.9.1

3 8

[alsa-devel] Segmentation Fault in snd_pcm_rate_hw_free()
by Alexandru Costache 18 Aug '15

18 Aug '15

Hi, I was reproducing the crash that Valentin Corfu reported earlier with his test. Valgrind and gdb show that expand_linear_s16() is somehow corrupting the area used for memory management by libc for rate->pareas[0].addr More exactly, the line that seems to corrupt the heap in static void expand_linear_s16() is: *dst = (old_sample * old_weight + new_sample * new_weight) >> 16; As Valgrind says: ==11161== Invalid write of size 2 ==11161== at 0x40A8A82: linear_expand_s16 (pcm_rate_linear.c:179) ==11161== by 0x40A85A9: linear_convert (pcm_rate_linear.c:320) ==11161== by 0x40A6915: do_convert (pcm_rate.c:537) ==11161== by 0x40A6C7D: snd_pcm_rate_write_areas1 (pcm_rate.c:550) ==11161== by 0x40A6C7D: snd_pcm_rate_commit_area (pcm_rate.c:749) ==11161== by 0x40A7014: snd_pcm_rate_drain (pcm_rate.c:1105) ==11161== by 0x4084511: snd_pcm_drain (pcm.c:1122) ==11161== by 0x8049288: main (in /root/pcm_min_daisuke) ==11161== Address 0x43d6258 is 0 bytes after a block of size 86,016 alloc'd ==11161== at 0x40261B1: malloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==11161== by 0x4099801: snd_pcm_mmap (pcm_mmap.c:425) ==11161== by 0x4093CA7: sndrv_pcm_hw_params (pcm_params.c:2366) ==11161== by 0x409A59C: snd1_pcm_generic_hw_params (pcm_generic.c:104) ==11161== by 0x4091E7F: snd1_pcm_hw_params_slave (pcm_params.c:2250) ==11161== by 0x40A54AC: snd_pcm_rate_hw_params (pcm_rate.c:246) ==11161== by 0x4093AE3: sndrv_pcm_hw_params (pcm_params.c:2326) ==11161== by 0x40ABC37: snd_pcm_plug_hw_params (pcm_plug.c:1045) ==11161== by 0x4093AE3: sndrv_pcm_hw_params (pcm_params.c:2326) ==11161== by 0x4084281: snd_pcm_hw_params (pcm.c:830) ==11161== by 0x8048F84: main (in /root/pcm_min_daisuke) ==11161== ==11161== Invalid free() / delete / delete[] / realloc() ==11161== at 0x402726D: free (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==11161== by 0x40A53C0: snd_pcm_rate_hw_free (pcm_rate.c:344) ==11161== by 0x4083CDA: snd_pcm_hw_free (pcm.c:853) ==11161== by 0x40A9E8D: snd_pcm_plug_hw_free (pcm_plug.c:1061) ==11161== by 0x4083CDA: snd_pcm_hw_free (pcm.c:853) ==11161== by 0x80492AC: main (in /root/pcm_min_daisuke) ==11161== Address 0xe705004b is not stack'd, malloc'd or (recently) free'd I'm not very familiar with alsa in general and was wondering if someone who knows more on it can drop a look on this one? Thank you, Alexandru

2 1

[alsa-devel] [PATCH v4 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 18 Aug '15

18 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..aabebcb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,7 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

1 3

[alsa-devel] [PATCH] ASoC: topology: Add Kconfig option for topology
by Mark Brown 18 Aug '15

18 Aug '15

Allow the topology code to be compiled out so that users who don't need topology don't need to havve the code compiled in, saving them some memory. Some more configuration could be added to remove some of the hooks into the core data structures but that is probably best done with some refactoring to use functions to do the updates of the data structures rather than ifdefing in the code as we'd need to do at the minute. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-topology.h | 12 ++++++++++++ sound/soc/Kconfig | 3 +++ sound/soc/Makefile | 3 +++ 3 files changed, 18 insertions(+) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 865a141..427bc41 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -141,6 +141,8 @@ struct snd_soc_tplg_ops { int io_ops_count; }; +#ifdef CONFIG_SND_SOC_TOPOLOGY + /* gets a pointer to data from the firmware block header */ static inline const void *snd_soc_tplg_get_data(struct snd_soc_tplg_hdr *hdr) { @@ -165,4 +167,14 @@ int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, const struct snd_soc_tplg_widget_events *events, int num_events, u16 event_type); +#else + +static inline int snd_soc_tplg_component_remove(struct snd_soc_component *comp, + u32 index) +{ + return 0; +} + +#endif + #endif diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 467ee1a..225bfda 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -30,6 +30,9 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_TOPOLOGY + bool + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 19fd711..134aca1 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,9 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o + +ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) snd-soc-core-objs += soc-topology.o +endif ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o -- 2.5.0

2 4

[alsa-devel] [PATCH v2 00/10] ASoC: support adding PCM dynamically from topology
by mengdong.lin＠intel.com 18 Aug '15

18 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This series allows the topology core to create PCM devices dynamically. The user can define different DAIs in the topology files for different versions of firmware, but share a generic platform and machine driver. A dummy DAI and DAI link can be used to register the soc card and specify the platform with topology. Then real DAIs are created in platform probing phase by the topology core, and the machine driver will be notified to create relavant DAI links. We tested using topology to create FE DAI/DAI links on Broadwell. Mengdong Lin (10): ASoC: Change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list ASoC: Add support for dummy DAI links and PCM runtimes ASoC: Bind new DAI links after probing components ASoC: Support adding a DAI dynamically ASoC: topology: Change pass number of DAI smaller than graph ASoC: topology: Change stream formats to bitwise flag ASOC: topology: Add PCM DAIs dynamically when loading them include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 1 - include/sound/soc.h | 28 +- include/uapi/sound/asoc.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/soc-core.c | 642 +++++++++++++++++++-------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 99 ++++- sound/soc/tegra/tegra_wm8903.c | 3 +- 22 files changed, 701 insertions(+), 291 deletions(-) -- 1.9.1

6 33

Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel driver
by Takashi Iwai 18 Aug '15

18 Aug '15

On Tue, 18 Aug 2015 01:03:31 +0200, Paulo Roberto de Oliveira Castro wrote: > > Hello again, > > I am using Arch now, and today kernel 4.1.5 was released on Arch's repo. > However, the sound isn't working yet. Am I missing something? The patch is included in 4.1.5. Make sure that you tested really properly. If not, check again whether 4.2 works in the same condition. Takashi > > 2015-07-29 14:40 GMT-03:00 Takashi Iwai <tiwai(a)suse.de>: > > > On Wed, 29 Jul 2015 18:48:29 +0200, > > Paulo Roberto de Oliveira Castro wrote: > > > > > > Thanks! Is there a way to track these changes? > > > > Track git tree :) > > > > At first, it's found in my sound git tree, then it'll be merged to > > Linus git tree, and then to each stable tree. > > > > > > Takashi > > > > > > > > 2015-07-29 4:08 GMT-03:00 Takashi Iwai <tiwai(a)suse.de>: > > > > > > > On Wed, 29 Jul 2015 00:32:08 +0200, > > > > Paulo Roberto de Oliveira Castro wrote: > > > > > > > > > > I'm not an expert on applying patches to the kernel (never did that > > > > > before), therefore, I don't know if I did everything right. > > > > > > > > > > What I did was: downloaded 4.2-rc4 from kernel.org[1], added the > > line by > > > > > hand (couldn't apply the patch using patch), copied the config file > > from > > > > > /boot/ (the config fedora 22 uses for its kernel) to .config on the > > root > > > > of > > > > > the kernel directory, runned "make oldconfig" (typing ENTER for every > > > > > question, is that right?), "make", "make modules_install", "make > > > > install". > > > > > Rebooted, and chose the new kernel I just installed. > > > > > > > > > > The sound worked like a charm, the speakers worked as expected, and > > the > > > > > volume was as loud as on Windows. > > > > > > > > Good to hear, I queued the fix now. > > > > > > > > > Is the patch going to be applied on master? Any chance that fix > > reaches > > > > my > > > > > Fedora 22 (with backport to older kernel versions)? > > > > > > > > The commit is with Cc to stable, so it'll be backported to stable > > > > kernels once when landed to upstream. Then Fedora will pick it up > > > > eventually later. > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > > > > > > > > > Thanks for the help, > > > > > > > > > > Paulo > > > > > > > > > > [1] > > > > > > > > > > > https://www.kernel.org/pub/linux/kernel/v4.x/testing/linux-4.2-rc4.tar.xz > > > > > > > > > > 2015-07-28 10:49 GMT-03:00 Paulo Roberto de Oliveira Castro < > > > > > p.oliveira.castro(a)gmail.com>: > > > > > > > > > > > > > > > > > > > > > > > ---------- Forwarded message --------- > > > > > > From: Takashi Iwai <tiwai(a)suse.de> > > > > > > Date: Tue, Jul 28, 2015, 03:35 > > > > > > Subject: Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) > > > > kernel > > > > > > driver > > > > > > To: Paulo Roberto de Oliveira Castro <p.oliveira.castro(a)gmail.com> > > > > > > Cc: <alsa-devel(a)alsa-project.org> > > > > > > > > > > > > > > > > > > On Tue, 28 Jul 2015 05:35:34 +0200, > > > > > > Paulo Roberto de Oliveira Castro wrote: > > > > > > > > > > > > > > Hi everyone, > > > > > > > > > > > > > > After installing Fedora 22 on my Dell Vostro 5480, the speakers > > don't > > > > > > work, > > > > > > > and the sound only comes out of the subwoofer. Here is the > > output of > > > > > > > alsa-info.sh. After talking to "debianuser" on IRC, we found this > > > > > > > workaround: > > > > > > > > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_PIN_WIDGET_CONTROL 0x40 # > > > > enable, > > > > > > > set it to OUT > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_CONNECT_SEL 0 # switch to > > 0x0c > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0xa000 # > > unmute > > > > left > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0x9000 # > > unmute > > > > > > right > > > > > > > > > > > > > > We suspect that it's a kernel driver bug. This makes the speakers > > > > work > > > > > > > alongside the headphones (both at the same time) but I need help > > > > making > > > > > > it > > > > > > > working when headphones are not plugged in. > > > > > > > > > > > > Does the patch below work? Try with 4.2-rc kernel at best. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > --- > > > > > > diff --git a/sound/pci/hda/patch_realtek.c > > > > b/sound/pci/hda/patch_realtek.c > > > > > > index 4ae877c3b6a1..18ae17ebb356 100644 > > > > > > --- a/sound/pci/hda/patch_realtek.c > > > > > > +++ b/sound/pci/hda/patch_realtek.c > > > > > > @@ -5185,6 +5185,7 @@ static const struct snd_pci_quirk > > > > alc269_fixup_tbl[] > > > > > > = { > > > > > > SND_PCI_QUIRK(0x1028, 0x064a, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x064b, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", > > > > > > ALC288_FIXUP_DELL_XPS_13), > > > > > > + SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", > > > > > > ALC290_FIXUP_SUBWOOFER_HSJACK), > > > > > > SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", > > > > > > ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x06da, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > > > > > > _______________________________________________ > > > > > Alsa-devel mailing list > > > > > Alsa-devel(a)alsa-project.org > > > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > > > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] > > > > >

1 0

[alsa-devel] [PATCH] ASoC: davinci-vcif: Use devm_snd_soc_register_component
by Vaishali Thakkar 17 Aug '15

17 Aug '15

Use resource managed function devm_snd_soc_register_component for component registration instead of snd_soc_register_component. Also, remove davinci_vcif_remove as it is now redundant. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/davinci/davinci-vcif.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index fabd05f..c77d921 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -231,8 +231,9 @@ static int davinci_vcif_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, davinci_vcif_dev); - ret = snd_soc_register_component(&pdev->dev, &davinci_vcif_component, - &davinci_vcif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &davinci_vcif_component, + &davinci_vcif_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "could not register dai\n"); return ret; @@ -241,23 +242,14 @@ static int davinci_vcif_probe(struct platform_device *pdev) ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - snd_soc_unregister_component(&pdev->dev); return ret; } return 0; } -static int davinci_vcif_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; -} - static struct platform_driver davinci_vcif_driver = { .probe = davinci_vcif_probe, - .remove = davinci_vcif_remove, .driver = { .name = "davinci-vcif", }, -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: mediatek: Remove AIF widgets for backend DAIs
by Koro Chen 17 Aug '15

17 Aug '15

DAPM core already creates widgets for DAIs. It is not necessary to declare them by SND_SOC_DAPM_AIF_IN/SND_SOC_DAPM_AIF_OUT. Furthermore, original codes use backend DAI's stream name to be the AIF widget name. It causes the same widget to be created twice, and after commit 92fa12426741 ("ASoC: dapm: Add new widgets to the end of the widget list") the first created widget (by snd_soc_dapm_new_controls) is used, not the 2nd created one (by snd_soc_dapm_new_dai_widgets), so audio path is broken. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 55471ee..d190fe0 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -897,10 +897,6 @@ static const struct snd_kcontrol_new mtk_afe_o10_mix[] = { }; static const struct snd_soc_dapm_widget mtk_afe_pcm_widgets[] = { - /* Backend DAIs */ - SND_SOC_DAPM_AIF_IN("I2S Capture", NULL, 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT("I2S Playback", NULL, 0, SND_SOC_NOPM, 0, 0), - /* inter-connections */ SND_SOC_DAPM_MIXER("I05", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I06", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -932,11 +928,6 @@ static const struct snd_soc_dapm_route mtk_afe_pcm_routes[] = { { "O10", "I18 Switch", "I18" }, }; -static const struct snd_soc_dapm_widget mtk_afe_hdmi_widgets[] = { - /* Backend DAIs */ - SND_SOC_DAPM_AIF_OUT("HDMIO Playback", NULL, 0, SND_SOC_NOPM, 0, 0), -}; - static const struct snd_soc_dapm_route mtk_afe_hdmi_routes[] = { {"HDMIO Playback", NULL, "HDMI"}, }; @@ -951,8 +942,6 @@ static const struct snd_soc_component_driver mtk_afe_pcm_dai_component = { static const struct snd_soc_component_driver mtk_afe_hdmi_dai_component = { .name = "mtk-afe-hdmi-dai", - .dapm_widgets = mtk_afe_hdmi_widgets, - .num_dapm_widgets = ARRAY_SIZE(mtk_afe_hdmi_widgets), .dapm_routes = mtk_afe_hdmi_routes, .num_dapm_routes = ARRAY_SIZE(mtk_afe_hdmi_routes), }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH RFC v3 0/7] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 17 Aug '15

17 Aug '15

This is my third RFC patch series on the subject. This time also the tda998x driver patches are a serious attempt to come up with an initial implementation for HDMI audio ASoC support. Russell's "drm/edid: add function to help find SADs" is here just to produce a working patch-set on top of mainline. It should be getting in trough drm-next at some point. The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (5): ASoC: hdmi: Remove obsolete dummy HDMI codec ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support Russell King - ARM Linux (1): drm/edid: add function to help find SADs, DO NOT MERGE .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 336 +++++++++++++-- include/drm/drm_edid.h | 19 + include/sound/hdmi-codec.h | 99 +++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 467 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ----- 10 files changed, 1026 insertions(+), 153 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

3 20

[alsa-devel] [PATCH v2] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 17 Aug '15

17 Aug '15

V2: Update after comments from Lars-Peter yesterday: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../devicetree/bindings/sound/ics43432.txt | 17 ++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 101 ++++++++++++++++++++ 5 files changed, 125 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..fe0af4a --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,101 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPL2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 5

[alsa-devel] [PATCH v2] ASoC: ssm2518: Add explicit device tree support
by Ricard Wanderlof 17 Aug '15

17 Aug '15

Add OF match table to SSM2518 to allow direct matching without going through I2C subsystem. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/ssm2518.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..2d199b1 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +811,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +829,7 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ssm2518_dt_ids), }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 5

Re: [alsa-devel] probe issue for Dice/OXFW drivers
by Clemens Ladisch 17 Aug '15

17 Aug '15

Takashi Sakamoto wrote: > Recently, I received an issue from a user of Mackie Onyx-i series. > According to the user, snd-dice is not applied to his Dice model. > [...] According to the log, I cannot find the first > transaction which snd-dice performs. Additionally, no error messages in > syslog. There are some checks performed before the first actual transaction. The kernel loads the driver if an entry in dice_id_table[] matches, i.e., if version==1. This values indeed is in the unit directory: > ROM header and bus information block > ----------------------------------------------------------------- > 400 0404f507 bus_info_length 4, crc_length 4, crc 62727 > 404 31333934 bus_name "1394" > 408 e0008102 irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 0, max_rec 8 (512) > 40c 000ff210 company_id 000ff2 | > 410 01800af7 device_id 1001800af7 | EUI-64 000ff21001800af7 > > root directory > ----------------------------------------------------------------- > 414 000639d3 directory_length 6, crc 14803 > 418 03000ff2 vendor > 41c 8100000a --> descriptor leaf at 444 > 420 17000006 model > 424 8100000d --> descriptor leaf at 458 > 428 0c0087c0 node capabilities per IEEE 1394 > 42c d1000001 --> unit directory at 430 > > unit directory at 430 > ----------------------------------------------------------------- > 430 00045246 directory_length 4, crc 21062 > 434 12000ff2 specifier id > 438 13000001 version > 43c 17000006 model > 440 8100000b --> descriptor leaf at 46c The dice_interface_check() function then makes other checks: /* * Check that GUID and unit directory are constructed according to DICE * rules, i.e., that the specifier ID is the GUID's OUI, and that the * GUID chip ID consists of the 8-bit category ID, the 10-bit product * ID, and a 22-bit serial number. */ These values are correct, except for the category ID. The driver already has an category ID exception for Weiss devices. It would be possible to add another exception, but given that firmware writers apparently like to change this value without good reason, it might be a better idea to just remove the category ID check. Regards, Clemens

2 1

[alsa-devel] [PATCH 00/25] ALSA: support AMDTP variants
by Takashi Sakamoto 17 Aug '15

17 Aug '15

This patchset updates my previous RFC, excluding control/status message for TASCAM FireWire series and a MOTU driver, [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094789.html Unfortunately, I've got no dedicated testers to this patchset. While, the supported models have little time to live. I think it's time to merge them. This patchset supports playbacking/capturing PCM samples/MIDI messages. Model specific functionalities such as MIDI mapping to physical controls should be done in userspace, with propper asynchronous transactions. I hope users to remind that my work is done by reasoning and assuming with a result of investigating packets on IEEE 1394 bus. There's no published specification, thus these drivers cannot fully satisfies users' expectation which comes from Windows or OS X. Changes: * Fix minor bugs, mainly for asynchronous transaction helper for MIDI port. * Regain PCM hardware constraint (I missed it). * Fix TASCAM quirk for MIDI transactions, as much as possible. Rest of work (but may not be done in this developing period): * Handling control/status messages for TASCAM driver. * Supporting MIDI exclusive message for outgoing packets. * Researching a reason to de-synchronize for Digi00x family. * Improving MIDI throttle for asynchronous transaction helper Takashi Sakamoto (25): ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' for functional separation ALSA: firewire-lib: functional separation for packet transmission layer and data processing layer ALSA: firewire-lib: add helper functions for asynchronous MIDI port ALSA: firewire-lib: add a restriction for a transaction at once ALSA: firewire-lib: schedule tasklet again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error ALSA: firewire-digi00x: add skeleton for Digi 002/003 family ALSA: firewire-digi00x: add data processing layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add MIDI functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging ALSA: firewire-digi00x: add support for MIDI ports for machine control ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add data processing layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add transaction functionality ALSA: firewire-tascam: add MIDI functionality ALSA: firewire-tascam: add hwdep interface include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 9 + sound/firewire/Kconfig | 27 ++ sound/firewire/Makefile | 4 +- sound/firewire/amdtp-am824.c | 451 ++++++++++++++++++++++++ sound/firewire/amdtp-am824.h | 47 +++ sound/firewire/{amdtp.c => amdtp-stream.c} | 375 ++++---------------- sound/firewire/{amdtp.h => amdtp-stream.h} | 113 ++---- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 16 +- sound/firewire/dice/dice-pcm.c | 12 +- sound/firewire/dice/dice-stream.c | 41 ++- sound/firewire/dice/dice.h | 2 +- sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 436 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++ sound/firewire/digi00x/digi00x-midi.c | 198 +++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 353 +++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 ++++++ sound/firewire/digi00x/digi00x-stream.c | 422 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 137 ++++++++ sound/firewire/digi00x/digi00x.c | 174 ++++++++++ sound/firewire/digi00x/digi00x.h | 157 +++++++++ sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 16 +- sound/firewire/fireworks/fireworks_pcm.c | 10 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/lib.c | 129 +++++++ sound/firewire/lib.h | 52 +++ sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 10 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- sound/firewire/tascam/Makefile | 4 + sound/firewire/tascam/amdtp-tascam.c | 248 +++++++++++++ sound/firewire/tascam/tascam-hwdep.c | 202 +++++++++++ sound/firewire/tascam/tascam-midi.c | 162 +++++++++ sound/firewire/tascam/tascam-pcm.c | 298 ++++++++++++++++ sound/firewire/tascam/tascam-proc.c | 88 +++++ sound/firewire/tascam/tascam-stream.c | 497 +++++++++++++++++++++++++++ sound/firewire/tascam/tascam-transaction.c | 291 ++++++++++++++++ sound/firewire/tascam/tascam.c | 230 +++++++++++++ sound/firewire/tascam/tascam.h | 140 ++++++++ 48 files changed, 5277 insertions(+), 498 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h rename sound/firewire/{amdtp.c => amdtp-stream.c} (70%) rename sound/firewire/{amdtp.h => amdtp-stream.h} (77%) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam-transaction.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

3 37

[alsa-devel] [PATCH 0/4] Minor cleanups to hda code
by Takashi Iwai 17 Aug '15

17 Aug '15

Hi, this is a series of trivial cleanup patches for HDA driver, mostly for procfs code. Takashi Iwai (4): ALSA: hda - Make some helper functions local ALSA: hda/proc - Add const to possible places ALSA: hda/proc - Fix racy string access for power states ALSA: hda/eld - Add const to possible places sound/pci/hda/hda_codec.c | 70 ------------------------------ sound/pci/hda/hda_codec.h | 7 --- sound/pci/hda/hda_eld.c | 10 ++--- sound/pci/hda/hda_proc.c | 107 +++++++++++++++++++++++++++++++++++----------- 4 files changed, 86 insertions(+), 108 deletions(-) -- 2.5.0

1 4

[alsa-devel] [PATCH] ASoC: topology: Add support for TLV bytes control
by mengdong.lin＠intel.com 17 Aug '15

17 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Allow vendor drivers to define bespoke bytes ext handlers and IDs for TLV bytes controls. And the topology core will bind these handlers by matching IDs defined by the vendor driver and user space topology data file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 865a141..3e412c6 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -89,6 +89,13 @@ struct snd_soc_tplg_kcontrol_ops { struct snd_ctl_elem_info *uinfo); }; +/* Bytes ext operations, for TLV byte controls */ +struct snd_soc_tplg_bytes_ext_ops { + u32 id; + int (*get)(unsigned int __user *bytes, unsigned int size); + int (*put)(const unsigned int __user *bytes, unsigned int size); +}; + /* * DAPM widget event handlers - used to map handlers onto widgets. */ @@ -139,6 +146,10 @@ struct snd_soc_tplg_ops { /* bespoke kcontrol handlers available for binding */ const struct snd_soc_tplg_kcontrol_ops *io_ops; int io_ops_count; + + /* bespoke bytes ext handlers available for binding */ + const struct snd_soc_tplg_bytes_ext_ops *bytes_ext_ops; + int bytes_ext_ops_count; }; /* gets a pointer to data from the firmware block header */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 607f98b..b5a2868 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -70,6 +70,10 @@ struct soc_tplg { const struct snd_soc_tplg_kcontrol_ops *io_ops; int io_ops_count; + /* bespoke bytes ext handlers, for TLV bytes controls */ + const struct snd_soc_tplg_bytes_ext_ops *bytes_ext_ops; + int bytes_ext_ops_count; + /* optional fw loading callbacks to component drivers */ struct snd_soc_tplg_ops *ops; }; @@ -513,6 +517,15 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, { int i; + /* TLV bytes controls don't need get/put ops */ + if (k->iface & SNDRV_CTL_ELEM_IFACE_MIXER + && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE + && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK + && hdr->ops.info == SND_SOC_TPLG_CTL_BYTES) { + k->info = snd_soc_bytes_info_ext; + return 0; + } + /* try and map standard kcontrols handler first */ for (i = 0; i < num_ops; i++) { @@ -547,6 +560,32 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, return -EINVAL; } +/* bind bytes ext get/put handlers for TLV bytes controls */ +static int soc_tplg_bytes_ext_bind_io(struct soc_tplg *tplg, + struct snd_soc_tplg_io_ops *ops_id, + struct soc_bytes_ext *sbe) +{ + const struct snd_soc_tplg_bytes_ext_ops *ops = tplg->bytes_ext_ops; + int num_ops = tplg->bytes_ext_ops_count; + int i; + + /* try bespoke handlers */ + for (i = 0; i < num_ops; i++) { + + if (sbe->put == NULL && ops[i].id == ops_id->put) + sbe->put = ops[i].put; + if (sbe->get == NULL && ops[i].id == ops_id->get) + sbe->get = ops[i].get; + } + + /* bespoke handlers found ? */ + if (sbe->put && sbe->get) + return 0; + + /* nothing to bind */ + return -EINVAL; +} + /* bind a widgets to it's evnt handlers */ int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, const struct snd_soc_tplg_widget_events *events, @@ -690,6 +729,14 @@ static int soc_tplg_dbytes_create(struct soc_tplg *tplg, unsigned int count, continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc, &be->hdr); + /* map bytes ext io handlers for TLV bytes controls */ + if (kc.access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + soc_tplg_bytes_ext_bind_io(tplg, + &be->ext_ops, sbe); + } + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc, (struct snd_soc_tplg_ctl_hdr *)be); @@ -1315,6 +1362,14 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc[i], &be->hdr); + /* map bytes ext io handlers for TLV bytes controls */ + if (kc[i].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + soc_tplg_bytes_ext_bind_io(tplg, + &be->ext_ops, sbe); + } + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc[i], (struct snd_soc_tplg_ctl_hdr *)be); @@ -1736,6 +1791,8 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, tplg.req_index = id; tplg.io_ops = ops->io_ops; tplg.io_ops_count = ops->io_ops_count; + tplg.bytes_ext_ops = ops->bytes_ext_ops; + tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count; return soc_tplg_load(&tplg); } -- 1.9.1

3 5

[alsa-devel] [PATCH v2 0/2] USB audio fixes
by Pierre-Louis Bossart 16 Aug '15

16 Aug '15

While trying to understand how asynchronous endpoints are supported, I identified two major issues. On my M-Audio transit, the following script tries to play using the 3 different endpoints supported and fails (error and no audio) after the first playback without the patches. With the patches on top of 4.1.5 or Takashi's for-next (4.2.0-rc5), I stopped the audio after an hour. while true do aplay -d5 -Dhw:2,0 ~/16_48.wav # play 48kHz, 16 bit on async ep aplay -d5 -Dhw:2,0 ~/24_48.wav # play 48kHz, 24 bit on 'none'/async ep aplay -d5 -Dhw:2,0 ~/24_96.wav # play 96 kHz, 24 bit on adaptive ep aplay -d5 -Dhw:2,0 ~/24_48.wav aplay -d5 -Dhw:2,0 ~/16_48.wav aplay -d5 -Dhw:2,0 ~/24_96.wav done The first issue is pretty bad with corrupted pointers. When transitioning from a alternate setting with a sync_endpoint to another which doesn't have one by construction (synchronous, adaptive, async on capture), the pointers are not reset and the code tries to set the parameters of a non-existent endpoint. The second issue is that the M-audio Transit descriptors are illegal but the logic can be changed without harm. I sometimes hear noise on playback start, probably the code needs to be changed to add enough silent samples to let the PLL lock. I haven't had time to work on this and this is more of an enhancement. v2: applied Takashi's comments unconditional clean-up of sync_endpoint pointers better logical tests and removal of goto Pierre-Louis Bossart (2): ALSA: usb: fix corrupted pointers due to interface setting change ALSA: usb: handle descriptor with SYNC_NONE illegal value sound/usb/pcm.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) -- 1.9.1

2 3

[alsa-devel] BugReport: Dice works partially [Broken]
by Stanislav Sergiienko 16 Aug '15

16 Aug '15

Hi, I am a user of TC Electronic DesktopKonnekt6 and latest kernel update 3.19.2 add partial support for my device and thus broke my usual sound workflow. So i want to make good bug report to make device usable on latest kernels. With 3.19.2 kernel device is successfully recognized by system and even sometimes can play sound(!). But always, after some time sound is breaks (usually after 30-90 seconds) and to restore it i must restart pulseaudio(v6.0) and application. So device becomes unusable on latests kernels. Earlier i use my sound card through ffado + jackdbus + pulseaudio. Now it's impossible, so i have to downgrade kernel. Please request any additional info that you need to fix problem and i will post it here soon. Also i can test any "bleeding edge" changes thru dkms as described here: https://github.com/takaswie/snd-firewire-improve Related bug(not mine): https://bugs.archlinux.org/task/44258 Some useful info: cat /proc/asound/cards 0 [DesktopKonnekt6]: DICE - DesktopKonnekt6 TC Electronic DesktopKonnekt6 (serial 12941) at fw1.0, S400 cat /proc/asound/card0/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc1:000100000000 notification: 00000040 nick name: DesktopKonnekt6 clock select: internal 44100 enable: 1 status: locked 44100 ext status: 00000000 sample rate: 44100 version: 1.0.4.0 clock caps: 44100 48000 88200 96000 176400 192000 arx1 arx2 internal clock source names: Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\INTERNAL\\ tx 0: iso channel: 0 audio channels: 6 midi ports: 0 speed: S400 names: ch 1\ch 2\ - \ - \ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 rx 0: iso channel: 1 sequence start: 0 audio channels: 6 midi ports: 0 names: main out 1\main out 2\phones out 1\phones out 2\ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 ext status: clock source: internal locked: 1 rate: 44100 adat user data: 0

3 4

[alsa-devel] [PATCH 0/9] Add DSP topology management for SKL
by Subhransu S. Prusty 15 Aug '15

15 Aug '15

[RESEND of the series after removing the extra patches] The SKL driver does not code DSP topology in driver. With this series the soc topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, compute resources for modules, initialize and bind the pipe modules. It uses the SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules. Last patch in this series removes the unused dais. Jeeja KP (9): ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Remove CPU dai that is not used sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 173 ++-- sound/soc/intel/skylake/skl-topology.c | 1264 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 21 + sound/soc/intel/skylake/skl-tplg-interface.h | 78 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 7 files changed, 1511 insertions(+), 68 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 2.4.3

3 28

[alsa-devel] [asoc:topic/fsl-card 2/2] sound/soc/fsl/fsl-asoc-card.c:486:29: error: 'WM8960_SYSCLK_AUTO' undeclared
by kbuild test robot 15 Aug '15

15 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/fsl-card head: 50e0ee01382b8e08289d3db209738c5856fd25cf commit: 50e0ee01382b8e08289d3db209738c5856fd25cf [2/2] ASoC: fsl-asoc-card: add wm8960 support config: xtensa-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 50e0ee01382b8e08289d3db209738c5856fd25cf # save the attached .config to linux build tree make.cross ARCH=xtensa All error/warnings (new ones prefixed by >>): sound/soc/fsl/fsl-asoc-card.c: In function 'fsl_asoc_card_probe': >> sound/soc/fsl/fsl-asoc-card.c:486:29: error: 'WM8960_SYSCLK_AUTO' undeclared (first use in this function) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; ^ sound/soc/fsl/fsl-asoc-card.c:486:29: note: each undeclared identifier is reported only once for each function it appears in vim +/WM8960_SYSCLK_AUTO +486 sound/soc/fsl/fsl-asoc-card.c 480 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; 481 priv->codec_priv.pll_id = WM8962_FLL; 482 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 483 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { 484 codec_dai_name = "wm8960-hifi"; 485 priv->card.set_bias_level = fsl_asoc_card_set_bias_level; > 486 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; 487 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; 488 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 489 } else { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: fsl: fix typos for sound/soc/fsl/*
by Xiubo Li 15 Aug '15

15 Aug '15

There are too much noise about the typos for fsl's drivers. So I fix all the typos here in this patch in almost every file I touched. Signed-off-by: Xiubo Li <lixiubo(a)cmss.chinamobile.com> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_sai.h | 12 ++++++------ sound/soc/fsl/fsl_spdif.c | 6 +++--- sound/soc/fsl/fsl_ssi.c | 4 ++-- 4 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e1aa3834..883087f 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -182,7 +182,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else { if (np) { - /* The eukrea,asoc-tlv320 driver was explicitely + /* The eukrea,asoc-tlv320 driver was explicitly * requested (through the device tree). */ dev_err(&pdev->dev, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..b4666fd 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -45,7 +45,7 @@ #define FSL_SAI_xFR(tx) (tx ? FSL_SAI_TFR : FSL_SAI_RFR) #define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) -/* SAI Transmit/Recieve Control Register */ +/* SAI Transmit/Receive Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) #define FSL_SAI_CSR_SR BIT(24) @@ -67,10 +67,10 @@ #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) -/* SAI Transmit and Recieve Configuration 1 Register */ +/* SAI Transmit and Receive Configuration 1 Register */ #define FSL_SAI_CR1_RFW_MASK 0x1f -/* SAI Transmit and Recieve Configuration 2 Register */ +/* SAI Transmit and Receive Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) #define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 @@ -82,12 +82,12 @@ #define FSL_SAI_CR2_BCD_MSTR BIT(24) #define FSL_SAI_CR2_DIV_MASK 0xff -/* SAI Transmit and Recieve Configuration 3 Register */ +/* SAI Transmit and Receive Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f -/* SAI Transmit and Recieve Configuration 4 Register */ +/* SAI Transmit and Receive Configuration 4 Register */ #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -97,7 +97,7 @@ #define FSL_SAI_CR4_FSP BIT(1) #define FSL_SAI_CR4_FSD_MSTR BIT(0) -/* SAI Transmit and Recieve Configuration 5 Register */ +/* SAI Transmit and Receive Configuration 5 Register */ #define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) #define FSL_SAI_CR5_WNW_MASK (0x1f << 24) #define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8e93221..400d60e 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -707,7 +707,7 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, return ret; } -/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */ +/* Q-subcode information. The byte size is SPDIF_UBITS_SIZE/8 */ static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -739,7 +739,7 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, return ret; } -/* Valid bit infomation */ +/* Valid bit information */ static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -767,7 +767,7 @@ static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, return 0; } -/* DPLL lock infomation */ +/* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c0b940e..b759448 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -156,7 +156,7 @@ struct fsl_ssi_soc_data { * * @dbg_stats: Debugging statistics * - * @soc: SoC specifc data + * @soc: SoC specific data */ struct fsl_ssi_private { struct regmap *regs; @@ -1210,7 +1210,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, } } - /* For those SLAVE implementations, we ingore non-baudclk cases + /* For those SLAVE implementations, we ignore non-baudclk cases * and, instead, abandon MASTER mode that needs baud clock. */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: adav80x: Remove .read_flag_mask setting from adav80x_regmap_config
by Axel Lin 15 Aug '15

15 Aug '15

Don't set .read_flag_mask for adav803, it's for adav801 only. Fixes: 0c2d69645628 ("ASoC: adav80x: Split SPI and I2C code into different modules") Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/adav80x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 88fd37c..198c924 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -865,7 +865,6 @@ const struct regmap_config adav80x_regmap_config = { .val_bits = 8, .pad_bits = 1, .reg_bits = 7, - .read_flag_mask = 0x01, .max_register = ADAV80X_PLL_OUTE, -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: ssm4567: Add sense support
by Lars-Peter Clausen 15 Aug '15

15 Aug '15

The ssm4567 has sensing circuitry that can be used to monitor the current and voltage on the speaker amplifier output has well as the VBAT input. This data can be output over the I2S interface so it can be processed by a DSP or similar. This patch adds the sense capture output stream to the CODEC DAI as well as DAPM widgets that ensure that the sensing circuitry is powered up when the capture stream is active. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm4567.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index ac242b2..e619d56 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -174,6 +174,12 @@ static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Amplifier Boost", SSM4567_REG_POWER_CTRL, 3, 1, &ssm4567_amplifier_boost_control), + SND_SOC_DAPM_SIGGEN("Sense"), + + SND_SOC_DAPM_PGA("Current Sense", SSM4567_REG_POWER_CTRL, 4, 1, NULL, 0), + SND_SOC_DAPM_PGA("Voltage Sense", SSM4567_REG_POWER_CTRL, 5, 1, NULL, 0), + SND_SOC_DAPM_PGA("VBAT Sense", SSM4567_REG_POWER_CTRL, 6, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("OUT"), }; @@ -181,6 +187,13 @@ static const struct snd_soc_dapm_route ssm4567_routes[] = { { "OUT", NULL, "Amplifier Boost" }, { "Amplifier Boost", "Switch", "DAC" }, { "OUT", NULL, "DAC" }, + + { "Current Sense", NULL, "Sense" }, + { "Voltage Sense", NULL, "Sense" }, + { "VBAT Sense", NULL, "Sense" }, + { "Capture Sense", NULL, "Current Sense" }, + { "Capture Sense", NULL, "Voltage Sense" }, + { "Capture Sense", NULL, "VBAT Sense" }, }; static int ssm4567_hw_params(struct snd_pcm_substream *substream, @@ -388,6 +401,14 @@ static struct snd_soc_dai_driver ssm4567_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32, }, + .capture = { + .stream_name = "Capture Sense", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32, + }, .ops = &ssm4567_dai_ops, }; -- 2.1.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: cs35l32: Use case range at appropriate places
by Axel Lin 15 Aug '15

15 Aug '15

Use case range syntax makes the code shorter with better readability when we have a large number of continuous switch cases. Below are the summary of readable/volatile/precious registers. The readable registers: 0x01 ~ 0x0D, 0x0F ~ 0x1C The volatile registers: 0x01 ~ 0x05, 0x15 ~ 0x18 The precious registers: 0x15 ~ 0x18 No functional change with this patch. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l32.c | 45 +++++---------------------------------------- 1 file changed, 5 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 203c720..44c30fe 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -74,33 +74,8 @@ static const struct reg_default cs35l32_reg_defaults[] = { static bool cs35l32_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_PWRCTL1: - case CS35L32_PWRCTL2: - case CS35L32_CLK_CTL: - case CS35L32_BATT_THRESHOLD: - case CS35L32_VMON: - case CS35L32_BST_CPCP_CTL: - case CS35L32_IMON_SCALING: - case CS35L32_AUDIO_LED_MNGR: - case CS35L32_ADSP_CTL: - case CS35L32_CLASSD_CTL: - case CS35L32_PROTECT_CTL: - case CS35L32_INT_MASK_1: - case CS35L32_INT_MASK_2: - case CS35L32_INT_MASK_3: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: - case CS35L32_FLASH_MODE: - case CS35L32_MOVIE_MODE: - case CS35L32_FLASH_TIMER: - case CS35L32_FLASH_INHIBIT: + case CS35L32_DEVID_AB ... CS35L32_AUDIO_LED_MNGR: + case CS35L32_ADSP_CTL ... CS35L32_FLASH_INHIBIT: return true; default: return false; @@ -110,15 +85,8 @@ static bool cs35l32_readable_register(struct device *dev, unsigned int reg) static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_DEVID_AB ... CS35L32_REV_ID: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; @@ -128,10 +96,7 @@ static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) static bool cs35l32_precious_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; -- 2.1.0

3 7

[alsa-devel] Applied "ASoC: cs42l56: Use case range at appropriate place" to the asoc tree
by Mark Brown 15 Aug '15

15 Aug '15

The patch ASoC: cs42l56: Use case range at appropriate place has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7f325bfc933d140337e55507821d6bb021321059 Mon Sep 17 00:00:00 2001 From: Axel Lin <axel.lin(a)ingics.com> Date: Wed, 12 Aug 2015 11:09:39 +0800 Subject: [PATCH] ASoC: cs42l56: Use case range at appropriate place The readable registers are in continuous range: 0x01 ~ 0x2e. Use case range syntax makes the code shorter with better readability when we have a large number of continuous switch cases. No functional change with this patch. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 47 +--------------------------------------------- 1 file changed, 1 insertion(+), 46 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 1e11ba4..98a68c6 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -115,52 +115,7 @@ static const struct reg_default cs42l56_reg_defaults[] = { static bool cs42l56_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS42L56_CHIP_ID_1: - case CS42L56_CHIP_ID_2: - case CS42L56_PWRCTL_1: - case CS42L56_PWRCTL_2: - case CS42L56_CLKCTL_1: - case CS42L56_CLKCTL_2: - case CS42L56_SERIAL_FMT: - case CS42L56_CLASSH_CTL: - case CS42L56_MISC_CTL: - case CS42L56_INT_STATUS: - case CS42L56_PLAYBACK_CTL: - case CS42L56_DSP_MUTE_CTL: - case CS42L56_ADCA_MIX_VOLUME: - case CS42L56_ADCB_MIX_VOLUME: - case CS42L56_PCMA_MIX_VOLUME: - case CS42L56_PCMB_MIX_VOLUME: - case CS42L56_ANAINPUT_ADV_VOLUME: - case CS42L56_DIGINPUT_ADV_VOLUME: - case CS42L56_MASTER_A_VOLUME: - case CS42L56_MASTER_B_VOLUME: - case CS42L56_BEEP_FREQ_ONTIME: - case CS42L56_BEEP_FREQ_OFFTIME: - case CS42L56_BEEP_TONE_CFG: - case CS42L56_TONE_CTL: - case CS42L56_CHAN_MIX_SWAP: - case CS42L56_AIN_REFCFG_ADC_MUX: - case CS42L56_HPF_CTL: - case CS42L56_MISC_ADC_CTL: - case CS42L56_GAIN_BIAS_CTL: - case CS42L56_PGAA_MUX_VOLUME: - case CS42L56_PGAB_MUX_VOLUME: - case CS42L56_ADCA_ATTENUATOR: - case CS42L56_ADCB_ATTENUATOR: - case CS42L56_ALC_EN_ATTACK_RATE: - case CS42L56_ALC_RELEASE_RATE: - case CS42L56_ALC_THRESHOLD: - case CS42L56_NOISE_GATE_CTL: - case CS42L56_ALC_LIM_SFT_ZC: - case CS42L56_AMUTE_HPLO_MUX: - case CS42L56_HPA_VOLUME: - case CS42L56_HPB_VOLUME: - case CS42L56_LOA_VOLUME: - case CS42L56_LOB_VOLUME: - case CS42L56_LIM_THRESHOLD_CTL: - case CS42L56_LIM_CTL_RELEASE_RATE: - case CS42L56_LIM_ATTACK_RATE: + case CS42L56_CHIP_ID_1 ... CS42L56_LIM_ATTACK_RATE: return true; default: return false; -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: Intel: fix runtime pm imbalance on error
by Junjie Mao 15 Aug '15

15 Aug '15

pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Junjie Mao <junjie.mao(a)enight.me> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d1b9e3..27a9653aa3d9 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -152,6 +152,7 @@ static int sst_power_control(struct device *dev, bool state) dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { + pm_runtime_put_sync(dev); dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } @@ -204,8 +205,10 @@ static int sst_cdev_open(struct device *dev, struct intel_sst_drv *ctx = dev_get_drvdata(dev); retval = pm_runtime_get_sync(ctx->dev); - if (retval < 0) + if (retval < 0) { + pm_runtime_put_sync(ctx->dev); return retval; + } str_id = sst_get_stream(ctx, str_params); if (str_id > 0) { @@ -672,8 +675,10 @@ static int sst_send_byte_stream(struct device *dev, if (NULL == bytes) return -EINVAL; ret_val = pm_runtime_get_sync(ctx->dev); - if (ret_val < 0) + if (ret_val < 0) { + pm_runtime_put_sync(ctx->dev); return ret_val; + } ret_val = sst_send_byte_stream_mrfld(ctx, bytes); sst_pm_runtime_put(ctx); -- 1.9.3

2 1

[alsa-devel] [PATCH 1/6] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..9c46c7d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

2 1

[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9c46c7d..2ce9e1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

2 1

[alsa-devel] [PATCH 3/6] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2ce9e1d..f3034b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

3 3

[alsa-devel] [PATCH 4/6] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f3034b9..0b4fcd9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

2 1

[alsa-devel] [PATCH 5/6] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0b4fcd9..e79dc16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

2 1

[alsa-devel] [PATCH 6/6] ASoC: fsl_ssi: adjust set DAI format in AC'97 mode
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Adjust set DAI format function in fsl_ssi driver so it doesn't fail and clears RXDIR in AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8e5ff5e..37aabe3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; }

3 9

[alsa-devel] [RFC PATCH 08/10] ASoC: topology: change pass number of DAI smaller than graph
by mengdong.lin＠intel.com 15 Aug '15

15 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The PCM DAIs need to be loaded and added to ASoC core ealier than the graph (route). Otherwise, adding routes will fail for missing DAIs. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1b6728b..0f8fa0d 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -45,12 +45,12 @@ #define SOC_TPLG_PASS_VENDOR 1 #define SOC_TPLG_PASS_MIXER 2 #define SOC_TPLG_PASS_WIDGET 3 -#define SOC_TPLG_PASS_GRAPH 4 -#define SOC_TPLG_PASS_PINS 5 -#define SOC_TPLG_PASS_PCM_DAI 6 +#define SOC_TPLG_PASS_PCM_DAI 4 +#define SOC_TPLG_PASS_GRAPH 5 +#define SOC_TPLG_PASS_PINS 6 #define SOC_TPLG_PASS_START SOC_TPLG_PASS_MANIFEST -#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PCM_DAI +#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PINS struct soc_tplg { const struct firmware *fw; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] USB audio fixes
by Pierre-Louis Bossart 14 Aug '15

14 Aug '15

While trying to understand how asynchronous endpoints are supported, I identified two major issues. On my M-Audio transit, the following script tries to play using the 3 different endpoints supported and fails (error and no audio) after the first playback without the patches. With the patches on top of 4.1.5 or Takashi's for-next (4.2.0-rc5), I stopped the audio after an hour. while true do aplay -d5 -Dhw:2,0 ~/16_48.wav # play 48kHz, 16 bit on async ep aplay -d5 -Dhw:2,0 ~/24_48.wav # play 48kHz, 24 bit on 'none'/async ep aplay -d5 -Dhw:2,0 ~/24_96.wav # play 96 kHz, 24 bit on adaptive ep aplay -d5 -Dhw:2,0 ~/24_48.wav aplay -d5 -Dhw:2,0 ~/16_48.wav aplay -d5 -Dhw:2,0 ~/24_96.wav done The first issue is pretty bad with corrupted pointers. When transitioning from a alternate setting with a sync_endpoint to another which doesn't have one by construction (synchronous, adaptive, async on capture), the pointers are not reset and the code tries to set the parameters of a non-existent endpoint. The second issue is that the M-audio Transit descriptors are illegal but the logic can be changed without harm. I sometimes hear noise on playback start, probably the code needs to be changed to add enough silent samples to let the PLL lock. I haven't had time to work on this and this is more of an enhancement. Comments welcome. Pierre-Louis Bossart (2): ALSA: usb: fix corrupted pointers due to interface setting change ALSA: usb: handle descriptor with SYNC_NONE illegal value sound/usb/pcm.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 14 Aug '15

14 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../devicetree/bindings/sound/ics43432.txt | 17 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 111 ++++++++++++++++++++ 5 files changed, 135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..902daae --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,111 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + unsigned int rate = params_rate(params); + + if (rate < ICS43432_RATE_MIN || rate > ICS43432_RATE_MAX) + return -EINVAL; + + return 0; +} + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .hw_params = ics43432_hw_params, + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPL2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 2

[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix parameter block size for UAC2 control requests
by Julian Scheel 14 Aug '15

14 Aug '15

USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- Changes in v2: - Removed superfluous int32 checks - Check control value length in function - Compare against 0 - Use a tmp variable to access audio_feature_info[control-1] Thanks for review by Takashi Iwai :) --- sound/usb/mixer.c | 64 ++++++++++++++++++++++++++++++++++++++----------------- sound/usb/mixer.h | 2 ++ 2 files changed, 46 insertions(+), 20 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4a49944..7046f54 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -282,6 +282,21 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) return val; } +static int uac2_ctl_value_size(int val_type) +{ + switch (val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + return 4; + case USB_MIXER_S16: + case USB_MIXER_U16: + return 2; + default: + return 1; + } + return 0; /* unreachable */ +} + /* * retrieve a mixer value @@ -328,14 +343,14 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = sizeof(__u16); + size = uac2_ctl_value_size(cval->val_type); } else { bRequest = UAC2_CS_RANGE; size = sizeof(buf); @@ -446,7 +461,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2]; + unsigned char buf[4]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; @@ -454,8 +469,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + val_len = uac2_ctl_value_size(cval->val_type); /* FIXME */ if (request != UAC_SET_CUR) { @@ -469,6 +483,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + buf[2] = (value_set >> 16) & 0xff; + buf[3] = (value_set >> 24) & 0xff; err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -799,24 +815,25 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { const char *name; - unsigned int type; /* control type (mute, volume, etc.) */ + int type; /* data type for uac1 */ + int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, - { "Tone Control - Bass", USB_MIXER_S8 }, - { "Tone Control - Mid", USB_MIXER_S8 }, - { "Tone Control - Treble", USB_MIXER_S8 }, - { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { "Volume", USB_MIXER_S16, -1 }, + { "Tone Control - Bass", USB_MIXER_S8, -1 }, + { "Tone Control - Mid", USB_MIXER_S8, -1 }, + { "Tone Control - Treble", USB_MIXER_S8, -1 }, + { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ + { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16 }, - { "Input Gain Pad Control", USB_MIXER_S16 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16, -1 }, + { "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1216,6 +1233,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, int readonly_mask) { struct uac_feature_unit_descriptor *desc = raw_desc; + struct usb_feature_control_info *ctl_info; unsigned int len = 0; int mapped_name = 0; int nameid = uac_feature_unit_iFeature(desc); @@ -1241,7 +1259,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + ctl_info = &audio_feature_info[control-1]; + if (state->mixer->protocol == UAC_VERSION_1) + cval->val_type = ctl_info->type; + else /* UAC_VERSION_2 */ + cval->val_type = ctl_info->type_uac2 >= 0 ? + ctl_info->type_uac2 : ctl_info->type; + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d3268f0..3417ef3 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -33,6 +33,8 @@ enum { USB_MIXER_U8, USB_MIXER_S16, USB_MIXER_U16, + USB_MIXER_S32, + USB_MIXER_U32, }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, -- 2.5.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix parameter block size for UAC2 control requests
by Julian Scheel 14 Aug '15

14 Aug '15

USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/usb/mixer.c | 83 +++++++++++++++++++++++++++++++++++++++++-------------- sound/usb/mixer.h | 2 ++ 2 files changed, 65 insertions(+), 20 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4a49944..675725b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -233,6 +233,14 @@ static int convert_signed_value(struct usb_mixer_elem_info *cval, int val) if (val >= 0x8000) val -= 0x10000; break; + case USB_MIXER_U32: + val &= 0xffffffff; + break; + case USB_MIXER_S32: + val &= 0xffffffff; + if (val >= 0x80000000) + val -= 0x100000000; + break; } return val; } @@ -253,6 +261,9 @@ static int convert_bytes_value(struct usb_mixer_elem_info *cval, int val) case USB_MIXER_S16: case USB_MIXER_U16: return val & 0xffff; + case USB_MIXER_S32: + case USB_MIXER_U32: + return val & 0xffffffff; } return 0; /* not reached */ } @@ -328,14 +339,26 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = sizeof(__u16); + switch (cval->val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + size = 4; + break; + case USB_MIXER_S16: + case USB_MIXER_U16: + size = 2; + break; + default: + size = 1; + break; + } } else { bRequest = UAC2_CS_RANGE; size = sizeof(buf); @@ -446,7 +469,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2]; + unsigned char buf[4]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; @@ -454,8 +477,19 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + switch (cval->val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + val_len = 4; + break; + case USB_MIXER_S16: + case USB_MIXER_U16: + val_len = 2; + break; + default: + val_len = 1; + break; + } /* FIXME */ if (request != UAC_SET_CUR) { @@ -469,6 +503,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + buf[2] = (value_set >> 16) & 0xff; + buf[3] = (value_set >> 24) & 0xff; err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -799,24 +835,25 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { const char *name; - unsigned int type; /* control type (mute, volume, etc.) */ + int type; /* data type for uac1 */ + int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, - { "Tone Control - Bass", USB_MIXER_S8 }, - { "Tone Control - Mid", USB_MIXER_S8 }, - { "Tone Control - Treble", USB_MIXER_S8 }, - { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { "Volume", USB_MIXER_S16, -1 }, + { "Tone Control - Bass", USB_MIXER_S8, -1 }, + { "Tone Control - Mid", USB_MIXER_S8, -1 }, + { "Tone Control - Treble", USB_MIXER_S8, -1 }, + { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ + { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16 }, - { "Input Gain Pad Control", USB_MIXER_S16 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16, -1 }, + { "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1241,7 +1278,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + if (state->mixer->protocol == UAC_VERSION_1) + cval->val_type = audio_feature_info[control-1].type; + else /* UAC_VERSION_2 */ + cval->val_type = audio_feature_info[control-1].type_uac2 > -1 ? + audio_feature_info[control-1].type_uac2 : + audio_feature_info[control-1].type; + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d3268f0..3417ef3 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -33,6 +33,8 @@ enum { USB_MIXER_U8, USB_MIXER_S16, USB_MIXER_U16, + USB_MIXER_S32, + USB_MIXER_U32, }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, -- 2.5.0

2 4

[alsa-devel] [PATCH v2 1/3] ASoC: dapm: dapm_dai_get_connected_widgets: Fix missing mutex unlock
by Lars-Peter Clausen 14 Aug '15

14 Aug '15

Make sure to unlock the DAPM mutex when dapm_widget_list_create() fails. This means the function will now generate a trace_snd_soc_dapm_connected event, even if the creation of the list fails. But that was the behavior before the patch that introduced the unlock issue, so that should be fine. Fixes: 1ce43acff0c0 ("ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- New in v2 --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7482c5d..c391bfe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1191,7 +1191,7 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, ret = dapm_widget_list_create(list, &widgets); if (ret) - return ret; + paths = ret; trace_snd_soc_dapm_connected(paths, stream); mutex_unlock(&card->dapm_mutex); -- 2.1.4

2 4

[alsa-devel] [PATCH 1/5] ASoC: dapm: Drop always true checks
by Lars-Peter Clausen 14 Aug '15

14 Aug '15

list_first_entry() always returns non NULL and since the code previously checked that list is not empty it will also be a valid pointer. Furthermore a path has always a sink or a source widget. So both checks are redundant and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7f834a0..10f7c7c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3485,11 +3485,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, list_source); - if (WARN_ON(!source_p || !sink_p) || - WARN_ON(!sink_p->source || !source_p->sink) || - WARN_ON(!source_p->source || !sink_p->sink)) - return -EINVAL; - source = source_p->source->priv; sink = sink_p->sink->priv; -- 2.1.4

2 11

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Specify codec_dai_name for DAI links
by Nicolin Chen 14 Aug '15

14 Aug '15

The dev->name of CODEC might not be identical to its codec_dai_name, so using dev->name to probe the CODEC dai is not a correct approach. This patch specifies each supporting codec_dai_name instead of using dev->name any more. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..040362f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -407,6 +407,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; struct clk *codec_clk; + const char *codec_dai_name; u32 width; int ret; @@ -459,6 +460,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + codec_dai_name = "cs42888"; priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; @@ -467,9 +469,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + codec_dai_name = "wm8962"; priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; @@ -521,7 +525,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; - priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].codec_dai_name = codec_dai_name; priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -530,7 +534,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exsits */ priv->dai_link[1].cpu_of_node = asrc_np; priv->dai_link[1].platform_of_node = asrc_np; - priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; -- 1.9.1

2 1