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

[alsa-devel] [PATCH v6 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 19 Oct '15

19 Oct '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 | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..8ad6f1b 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,13 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers
by Luis de Bethencourt 15 Oct '15

15 Oct '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/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 03fa1cb..8c435be 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 = { -- 2.4.6

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[alsa-devel] [PATCH v5 0/10] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 14 Oct '15

14 Oct '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 v5: - add Mark's tag for spi changes - remove unnecessary whitespace change - use switch statement for I2S dma_quirk's manipulation Changes in v4: - remove spi & i2s dts changes and query quirk from dmaengine API suggeseted by Mark. - fix typo - Add dmaengine_get_quirk hook and implement it for pl330 Changes in v3: - add Sunny's tag - 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 (5): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform dmaengine: add API for getting dma controller's quirk DMA: pl330: implement dmaengine_get_quirks hook snd: dmaengine-pcm: add snd_dmaengine_pcm_get_quirks interface Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + arch/arm/boot/dts/rk3288.dtsi | 3 + arch/arm/boot/dts/rk3xxx.dtsi | 3 + drivers/dma/pl330.c | 110 +++++++++++++++------ drivers/spi/spi-rockchip.c | 13 ++- include/linux/dmaengine.h | 9 ++ sound/soc/rockchip/rockchip_i2s.c | 21 ++++ sound/soc/soc-generic-dmaengine-pcm.c | 24 +++++ 8 files changed, 153 insertions(+), 31 deletions(-) -- 2.3.7

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[alsa-devel] [PATCH v2] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 13 Oct '15

13 Oct '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 | 57 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 10 +++++++ 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..642dfbd 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 +721,9 @@ 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; + BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS); + for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); 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..89dc7d6 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,18 @@ #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. + */ +#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

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[alsa-devel] [PATCH V2 0/7] BAT: Add Basic Audio Tester command line tool
by han.lu＠intel.com 13 Oct '15

13 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> BAT (Basic Audio Tester) is a simple command line utility intended to automate audio driver and sound server QA testing with no human interaction. BAT works by playing an audio stream and capturing the same stream in either a digital or analog loop back. It then compares the captured stream to the original to determine if the test case passes or fails. The main idea of frequency detecting is: The analysis function reads data from WAV file, run FFT against the data to get magnitude of frequency vectors, and then calculates the average value and standard deviation of frequency vectors. After that, we define a threshold: threshold = 3 * standard_deviation + average_value Frequencies with amplitude larger than threshold will be recognized as a peak, and the frequency with largest peak value will be recognized as a detected frequency. BAT then compares the detected frequency to target frequency, to decide if the detecting passes or fails. BAT supports 4 working modes: 1. Single line playback; 2. Single line capture and analysis; 3. Playback and capture in loop, and analyze captured data; 4. Local analyze without actual playback or capture. BAT will check devices input by user to decide which mode to use. BAT will create threads for playback and record, and will run spectrum analysis for mode 2, 3 or 4. Changes on V2: 1. Adopt Caleb Crome's method to generate sine signal to avoids unwanted distortion on waveform. Lu, Han (7): BAT: Add initial functions BAT: Add common definitions and functions BAT: Add playback and record functions BAT: Add signal generator BAT: Add converting functions BAT: Add spectrum analysis functions BAT: Add Makefile and configures Makefile.am | 3 + bat/Makefile.am | 14 ++ bat/alsa.c | 594 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsa.h | 20 ++ bat/analyze.c | 314 +++++++++++++++++++++++++++++ bat/analyze.h | 16 ++ bat/bat.c | 603 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/common.c | 198 +++++++++++++++++++ bat/common.h | 176 +++++++++++++++++ bat/convert.c | 113 +++++++++++ bat/convert.h | 23 +++ bat/signal.c | 182 +++++++++++++++++ bat/signal.h | 30 +++ configure.ac | 19 +- 14 files changed, 2304 insertions(+), 1 deletion(-) create mode 100644 bat/Makefile.am create mode 100644 bat/alsa.c create mode 100644 bat/alsa.h create mode 100644 bat/analyze.c create mode 100644 bat/analyze.h create mode 100644 bat/bat.c create mode 100644 bat/common.c create mode 100644 bat/common.h create mode 100644 bat/convert.c create mode 100644 bat/convert.h create mode 100644 bat/signal.c create mode 100644 bat/signal.h -- 1.9.1

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[alsa-devel] [RFC] i2c: Revert back to old device naming for ACPI enumerated I2C slaves
by Jarkko Nikula 12 Oct '15

12 Oct '15

Commit 70762abb9f89 ("i2c: Use stable dev_name for ACPI enumerated I2C slaves") broke the lm-sensors which relies on I2C hwmon slave devices under /sys/bus/i2c/devices/ to be named as "x-00yz". However if those hwmon devices are ACPI 5 enumerated their name became "i2c-INTABCD:ij" and sysfs code in lm-sensors does not find them anymore: lib/sysfs.c:665: if ((!subsys || !strcmp(subsys, "i2c")) && sscanf(dev_name, "%hd-%x", &entry.chip.bus.nr, &entry.chip.addr) == 2) { This patch fixes this by reverting back the old device naming in i2c-core but at the same avoids regression to ALSA SoC drivers that depend on stable device binding. Reverted I2C slave device naming is handled in ALSA SoC core by using the ACPI name as a component name if device is ACPI probed. This keeps the component binding in ALSA SoC stable and requires only minimal changes to affected machine drivers. Fixes: 70762abb9f89 i2c: Use stable dev_name for ACPI enumerated I2C slaves Reported-by: Dustin Byford <dustin(a)cumulusnetworks.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- This is on top of 4.2.0-rc8. I didn't check exact kernel versions where this still applies but I know we would need a few different versions for older stable versions. Mainly because of added and moved drivers in sound/soc/intel/. This is for discussion so I didn't cc stable(a)vger.kernel.org yet. I was thinking would it work if we'd keep the stable name but have an another symlink in /sys/bus/i2c/devices/ that uses "x-00yz" name. However this feels ill-use of devices directory and probably causes more troubles elsewhere. I don't know how common ACPI enumerated I2C hwmon devices are but I guess they exists since Dustin notices this issue and wrote "With that change, /sys/bus/i2c/devices/i2c-0-004a, for example, became /sys/bus/i2c/devices/i2c-PNPXXXX:xx". It could be possible there is use for the new device naming. I found a comment from Documentation/hid/hid-sensor.txt introduced by commit b2eafd7282fd ("HID: sensor: Update document for custom sensor") that seems to indicate current i2c-INTABCD:xy naming may be used from userspace too but I didn't find examples of that. --- drivers/i2c/i2c-core.c | 21 ++++----------------- sound/soc/intel/boards/broadwell.c | 6 +++--- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +++--- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/soc-core.c | 10 ++++++++++ 10 files changed, 28 insertions(+), 31 deletions(-) diff --git a/drivers/i2c/i2c-core.c b/drivers/i2c/i2c-core.c index c83e4d13cfc5..e9c227b9dc92 100644 --- a/drivers/i2c/i2c-core.c +++ b/drivers/i2c/i2c-core.c @@ -913,22 +913,6 @@ void i2c_unlock_adapter(struct i2c_adapter *adapter) } EXPORT_SYMBOL_GPL(i2c_unlock_adapter); -static void i2c_dev_set_name(struct i2c_adapter *adap, - struct i2c_client *client) -{ - struct acpi_device *adev = ACPI_COMPANION(&client->dev); - - if (adev) { - dev_set_name(&client->dev, "i2c-%s", acpi_dev_name(adev)); - return; - } - - /* For 10-bit clients, add an arbitrary offset to avoid collisions */ - dev_set_name(&client->dev, "%d-%04x", i2c_adapter_id(adap), - client->addr | ((client->flags & I2C_CLIENT_TEN) - ? 0xa000 : 0)); -} - /** * i2c_new_device - instantiate an i2c device * @adap: the adapter managing the device @@ -987,7 +971,10 @@ i2c_new_device(struct i2c_adapter *adap, struct i2c_board_info const *info) client->dev.of_node = info->of_node; client->dev.fwnode = info->fwnode; - i2c_dev_set_name(adap, client); + /* For 10-bit clients, add an arbitrary offset to avoid collisions */ + dev_set_name(&client->dev, "%d-%04x", i2c_adapter_id(adap), + client->addr | ((client->flags & I2C_CLIENT_TEN) + ? 0xa000 : 0)); status = device_register(&client->dev); if (status) goto out_err; diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 8bafaf6ceab1..3abcf0d7682e 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -205,7 +205,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, - .codec_name = "i2c-INT343A:00", + .codec_name = "INT343A:00", .codec_dai_name = "rt286-aif1", .init = broadwell_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | @@ -223,7 +223,7 @@ static int broadwell_suspend(struct snd_soc_card *card){ struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + if (!strcmp(codec->component.name, "INT343A:00")) { dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); rt286_mic_detect(codec, NULL); break; @@ -236,7 +236,7 @@ static int broadwell_resume(struct snd_soc_card *card){ struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + if (!strcmp(codec->component.name, "INT343A:00")) { dev_dbg(codec->dev, "enabling jack detect for resume.\n"); rt286_mic_detect(codec, &broadwell_headset); break; diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 7ab8cc9fbfd5..ae336b8522a8 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -116,7 +116,7 @@ static struct snd_soc_dai_link byt_max98090_dais[] = { .stream_name = "Audio", .cpu_dai_name = "baytrail-pcm-audio", .codec_dai_name = "HiFi", - .codec_name = "i2c-193C9890:00", + .codec_name = "193C9890:00", .platform_name = "baytrail-pcm-audio", .init = byt_max98090_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index ae89b9b966d9..275b056c495c 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -186,7 +186,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .stream_name = "Audio", .cpu_dai_name = "baytrail-pcm-audio", .codec_dai_name = "rt5640-aif1", - .codec_name = "i2c-10EC5640:00", + .codec_name = "10EC5640:00", .platform_name = "baytrail-pcm-audio", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7f55d59024a8..973f07548b08 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link byt_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "rt5640-aif1", - .codec_name = "i2c-10EC5640:00", + .codec_name = "10EC5640:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .be_hw_params_fixup = byt_codec_fixup, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 70f832114a5a..1a5348771e03 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -232,7 +232,7 @@ static struct snd_soc_ops cht_be_ssp2_ops = { static struct snd_soc_aux_dev cht_max98090_headset_dev = { .name = "Headset Chip", .init = cht_max98090_headset_init, - .codec_name = "i2c-104C227E:00", + .codec_name = "104C227E:00", }; static struct snd_soc_dai_link cht_dailink[] = { @@ -265,7 +265,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "HiFi", - .codec_name = "i2c-193C9890:00", + .codec_name = "193C9890:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf467842a..c05841264dc7 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -290,7 +290,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "rt5645-aif1", - .codec_name = "i2c-10EC5645:00", + .codec_name = "10EC5645:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, @@ -365,7 +365,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } } card->dev = &pdev->dev; - sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + sprintf(codec_name, "%s:00", drv->acpi_card->codec_id); /* set correct codec name */ strcpy((char *)card->dai_link[2].codec_name, codec_name); snd_soc_card_set_drvdata(card, drv); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5be439e..632926d0c595 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -280,7 +280,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .no_pcm = 1, .nonatomic = true, .codec_dai_name = "rt5670-aif1", - .codec_name = "i2c-10EC5670:00", + .codec_name = "10EC5670:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, @@ -296,7 +296,7 @@ static int cht_suspend_pre(struct snd_soc_card *card) struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + if (!strcmp(codec->component.name, "10EC5670:00")) { dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); rt5670_jack_suspend(codec); break; @@ -310,7 +310,7 @@ static int cht_resume_post(struct snd_soc_card *card) struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + if (!strcmp(codec->component.name, "10EC5670:00")) { dev_dbg(codec->dev, "enabling jack detect for resume.\n"); rt5670_jack_resume(codec); break; diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 22558572cb9c..6fd6491e368d 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -160,7 +160,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, - .codec_name = "i2c-INT33CA:00", + .codec_name = "INT33CA:00", .codec_dai_name = "rt5640-aif1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0e1e69c7abd5..811d63e31a7a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -34,6 +34,7 @@ #include <linux/ctype.h> #include <linux/slab.h> #include <linux/of.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -2476,10 +2477,18 @@ static char *fmt_single_name(struct device *dev, int *id) { char *found, name[NAME_SIZE]; int id1, id2; + struct acpi_device *adev; if (dev_name(dev) == NULL) return NULL; + if (ACPI_HANDLE(dev) && + !acpi_bus_get_device(ACPI_HANDLE(dev), &adev)) { + /* use ACPI name as a component name for ACPI probed devices */ + strlcpy(name, acpi_dev_name(adev), NAME_SIZE); + goto out; + } + strlcpy(name, dev_name(dev), NAME_SIZE); /* are we a "%s.%d" name (platform and SPI components) */ @@ -2508,6 +2517,7 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } +out: return kstrdup(name, GFP_KERNEL); } -- 2.5.0

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[alsa-devel] Boss GT-001
by maillist＠superlative.org 10 Oct '15

10 Oct '15

I recently got access to a Boss GT-001. This is a small USB audio Interface for guitar with onboard FX which doesn't work with ALSA under Linux Mint 17 (running kernel 3.16.0-38-generic). I've not tried it with the latest ALSA development branch, but scanning the recent ALSA Dev archives, I can't see anything which might fix this. # lsusb ... Bus 001 Device 011: ID 0582:0187 Roland Corp (detailed lsusb output is towards the bottom of this email) The unit is moderately quirky compared to most audio devices. This unit includes on-board Boss guitar amp emulations and effects and the USB connection allows you to simultaneously record both the "wet" (with effects) and "dry" (without effects) signal. Thus the USB carries two separate stereo pairs to the computer. It also allows "re-amping" which allows you to apply effects to the output signal. The way this seems to do this is by having two separate stereo pair outputs from the computer one of which is processed via the onboard effects, and one of which is not. This is all just background, but the upshot of this is that the unit is a 2x stereo pair in, and 2x stereo pair out. However, this not only does not show up properly under Linux, but it will not play or record anything. ALSA sees it thusly: # aplay -l ... card 4: GT001 [GT-001], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 PLaying a CD-format file: # aplay hw:4 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1233: Sample format not available Available formats: - S32_LE # aplay hw:4 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo No audio is output For input: # arecord -l card 4: GT001 [GT-001], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 The unit also supports MIDI which is used for both control of the unit itself via a PC editing application (editing settings, changing patches, etc.) and for simple DAW control. There are three USB MIDI ports for this which, on Windows, are labelled "GT-001", "GT-001 DAW CTRL" and "GT-001 CTRL" On Linux these show up as follows: # amidi -l IO hw:4,0,0 GT-001 MIDI 1 IO hw:4,0,1 GT-001 MIDI 2 IO hw:4,0,2 GT-001 MIDI 3 Any advice and pointers to get this working would be appreciated. Dmesg output: [17530.759008] usb 1-1: new high-speed USB device number 12 using ehci-pci [17530.893299] usb 1-1: New USB device found, idVendor=0582, idProduct=0187 [17530.893303] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [17530.893305] usb 1-1: Product: GT-001 [17530.893307] usb 1-1: Manufacturer: BOSS [17530.894412] snd-usb-audio: probe of 1-1:1.0 failed with error -5 [17530.927591] usb 1-1: Unable to change format on ep #8e: already in use [17530.927614] usb 1-1: Unable to change format on ep #8e: already in use [17530.927764] usb 1-1: Unable to change format on ep #8e: already in use [17530.927945] usb 1-1: Unable to change format on ep #8e: already in use [17530.928161] usb 1-1: Unable to change format on ep #8e: already in use [17530.928600] usb 1-1: Unable to change format on ep #8e: already in use [17530.928620] usb 1-1: Unable to change format on ep #8e: already in use [17530.928693] usb 1-1: Unable to change format on ep #8e: already in use [17530.928832] usb 1-1: Unable to change format on ep #8e: already in use [17530.928958] usb 1-1: Unable to change format on ep #8e: already in use [17530.929350] usb 1-1: Unable to change format on ep #8e: already in use [17530.929377] usb 1-1: Unable to change format on ep #8e: already in use [17530.929500] usb 1-1: Unable to change format on ep #8e: already in use [17530.929743] usb 1-1: Unable to change format on ep #8e: already in use [17530.929988] usb 1-1: Unable to change format on ep #8e: already in use [17530.930577] usb 1-1: Unable to change format on ep #8e: already in use [17530.930617] usb 1-1: Unable to change format on ep #8e: already in use [17530.930808] usb 1-1: Unable to change format on ep #8e: already in use [17530.931064] usb 1-1: Unable to change format on ep #8e: already in use [17530.931213] usb 1-1: Unable to change format on ep #8e: already in use [17531.330778] usb 1-1: Unable to change format on ep #8e: already in use [17531.334500] usb 1-1: Unable to change format on ep #8e: already in use lsusb output: lsusb -vv -d 0582:0187 Bus 001 Device 011: ID 0582:0187 Roland Corp. Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x0187 bcdDevice 0.00 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 176 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 04 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x0d EP 13 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 04 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8e EP 14 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 03 03 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Cheers, Keith

4 19

[alsa-devel] Dummy module and sample rates
by Uli Franke 10 Oct '15

10 Oct '15

Hi Clemens, I fiddled around with the Dummy module and it seems to be the thing we're looking for. I observed a rather strange behavior though: I set the hw params sample rate to 44k1 without error, but a subsequent call of snd_pcm_start returns EPIPE. When setting hw params sample rate to 4k this does not happen. Did I miss anything? Regards Uli ____ uli@ankerklause:~/.../dlna/wdmr$ modinfo snd-dummy filename: /lib/modules/3.16.0-46-generic/kernel/sound/drivers/snd-dummy.ko license: GPL description: Dummy soundcard (/dev/null) author: Jaroslav Kysela <perex(a)perex.cz> srcversion: BB2C5716F580F3D7D0F53D1 depends: snd-pcm,snd intree: Y vermagic: 3.16.0-46-generic SMP mod_unload modversions 686 signer: Magrathea: Glacier signing key sig_key: 6C:3B:5C:5E:34:A0:6D:7D:5F:7D:A3:D0:50:98:75:D9:6C:6D:E6:C4 sig_hashalgo: sha512 parm: index:Index value for dummy soundcard. (array of int) parm: id:ID string for dummy soundcard. (array of charp) parm: enable:Enable this dummy soundcard. (array of bool) parm: model:Soundcard model. (array of charp) parm: pcm_devs:PCM devices # (0-4) for dummy driver. (array of int) parm: pcm_substreams:PCM substreams # (1-128) for dummy driver. (array of int) parm: fake_buffer:Fake buffer allocations. (bool) parm: hrtimer:Use hrtimer as the timer source. (bool)

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[alsa-devel] [PATCH 0/2] ASoC: rockchip: i2s: add 8 channels capture and lrck-mode
by Sugar Zhang 08 Oct '15

08 Oct '15

Support max 8 channels capture. support lrck clk mode configuration, there are 3 modes: - txrx: lrck_tx and lrck_rx are different. - tx_share: lrck_tx is shared with lrck_rx. - rx_share: lrck_rx is shared with lrck_tx. Sugar Zhang (2): ASoC: rockchip: i2s: add 8 channels capture and lrck-mode support Documentation: DT bindings: rockchip-i2s: add capture and lrck-mode .../devicetree/bindings/sound/rockchip-i2s.txt | 5 +++ sound/soc/rockchip/rockchip_i2s.c | 48 +++++++++++++++++++++- sound/soc/rockchip/rockchip_i2s.h | 23 +++++++++++ 3 files changed, 74 insertions(+), 2 deletions(-) -- 2.3.6

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[alsa-devel] [PATCH v3 0/8] Add DSP topology management for SKL
by Vinod Koul 07 Oct '15

07 Oct '15

The SKL driver does not code DSP topology in driver. With this series the ASoC 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. changes in v3: - removeno/less logic wrappers - remove most of debug prints, few kept - use asoc widget list macros instead of using list_for_xxx - fix one instance of no empty line between two functions - fix tplg event ID > 255 - add more comments for locking, usage of widget list and few other places, fixed few error prints, add file name etc changes in v2: - add more explanation in changelog and code - add explanation of MCPS - fix whitespace issues - remove skl widget check on event handler as it is redundant - add switch for depth - add else for direction based code - remove cast for NHLT blob query - make debug prints to error - remove dump info - clean some debug prints Jeeja KP (6): 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: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table Vinod Koul (2): ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 143 +++- sound/soc/intel/skylake/skl-topology.c | 1185 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 28 + sound/soc/intel/skylake/skl-tplg-interface.h | 80 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 8 files changed, 1443 insertions(+), 37 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 1.9.1

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