January 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: pxa: remove incorrect do_div() call
by Arnd Bergmann 26 Feb '16

26 Feb '16

The new optimized do_div implementation (now in asm-generic/next) exposes a glitch in the brownstone audio driver by producing a compile-time warning: sound/soc/pxa/brownstone.c: In function 'brownstone_wm8994_hw_params': sound/soc/pxa/brownstone.c:67:85: warning: comparison of distinct pointer types lacks a cast sound/soc/pxa/brownstone.c:67:10125: warning: right shift count >= width of type [-Wshift-count-overflow] sound/soc/pxa/brownstone.c:67:10254: warning: passing argument 1 of '__div64_32' from incompatible pointer type [-Wincompatible-pointer-types] The driver just divides two plain integer values, so it should not use do_div to start with, but has apparently done so ever since the code was first merged. This replaces do_div with a simple division operator. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- I think this one is harmless, so we only need it in 4.5 to avoid the warning, but not backported to earlier kernels. Found today on ARM allmodconfig diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 6147e86e9b0f..416ea646c3b1 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -63,8 +63,7 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, sysclk = params_rate(params) * 512; sspa_mclk = params_rate(params) * 64; } - sspa_div = freq_out; - do_div(sspa_div, sspa_mclk); + sspa_div = freq_out / sspa_mclk; snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);

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[alsa-devel] [PATCH 0/1] ALSA: hda - Use acpi_dev_present()
by Lukas Wunner 25 Feb '16

25 Feb '16

Hi Takashi, the acpi_dev_present() API has now landed in Linus' tree. Thus, after Linus' tree gets merged back into yours, it would be possible to use the API in the Thinkpad hda drivers as per the following patch. I've also pushed it to GitHub in case anyone prefers perusing it in a browser: https://github.com/l1k/linux/commit/a1473d726b57eaf97c4de8812c5967603068e261 An ack for this patch was kindly provided by Hui Wang with: Message-ID: <5653C291.9090607(a)canonical.com> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100962.h… Thanks, Lukas Lukas Wunner (1): ALSA: hda - Use acpi_dev_present() sound/pci/hda/thinkpad_helper.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH 3/3] ASoC: fsl_ssi: remove register defaults
by Maciej S. Szmigiero 22 Feb '16

22 Feb '16

There is no guarantee that on fsl_ssi module load SSI registers will have their power-on-reset values. In fact, if the driver is reloaded the values in registers will be whatever they were set to previously. This fixes hard lockup on fsl_ssi module reload, at least in AC'97 mode. Fixes: 05cf237972fe ("ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast") Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 40dfd8a36484..f6ef7d64acb3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,20 +112,6 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; -static const struct reg_default fsl_ssi_reg_defaults[] = { - {CCSR_SSI_SCR, 0x00000000}, - {CCSR_SSI_SIER, 0x00003003}, - {CCSR_SSI_STCR, 0x00000200}, - {CCSR_SSI_SRCR, 0x00000200}, - {CCSR_SSI_STCCR, 0x00040000}, - {CCSR_SSI_SRCCR, 0x00040000}, - {CCSR_SSI_SACNT, 0x00000000}, - {CCSR_SSI_STMSK, 0x00000000}, - {CCSR_SSI_SRMSK, 0x00000000}, - {CCSR_SSI_SACCEN, 0x00000000}, - {CCSR_SSI_SACCDIS, 0x00000000}, -}; - static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -190,8 +176,6 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .reg_defaults = fsl_ssi_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg,

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[alsa-devel] [PATCH v5 00/14] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 20 Feb '16

20 Feb '16

This patch series adds DP audio and hotplug notification support. In skylake platform the DP is on a different port which is not enabled by default. A vendor widget is programmed to enable playback on all ports. This also enables all the converters. Codec Dais, widgets, graph are now created dynamically based on the nodes enumerated, thus removing the hardcoding. Mux controls are used to establish routing stream to a specific port. For DP audio support, infoframe needs to be different. Based on ELD queried we either pack HDMI or DP infoframe. Also with this series, Jack notification support is added and Jack event is reported to userspace as is done in case of legacy driver. Few fixes are added to check for possible memory leak, reconfiguring register during resume from D3 etc. The 8th patch is adding a small macro for getting connection type in drm header. It is reviewed by drm folks, CCed them. Please merge it through sound trees. Changes in v5: - Added more comments and reworded some commit messages - Jack reporting support to user space added - Fix to check for possible memory leak in hw_params - Fix not to fail in dai startup to make userland APIs happy Cc: Jani Nikula <jani.nikula(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Daniel Vetter <daniel.vetter(a)intel.com> Jeeja KP (2): ASoC: hdac_hdmi: Add jack reporting ASoC: hdac_hdmi: Add PM support Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Subhransu S. Prusty (11): ASoC: hdac_hdmi: Add hotplug notification and read eld ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi: create dais based on number of cvts ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Enable playback on all enumerated ports drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume include/drm/drm_edid.h | 12 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1197 +++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/hdac_hdmi.h | 6 + 4 files changed, 1094 insertions(+), 122 deletions(-) create mode 100644 sound/soc/codecs/hdac_hdmi.h -- 1.9.1

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[alsa-devel] [RFC v2 0/6] sti: add audio interface to the hdmi driver
by Arnaud Pouliquen 19 Feb '16

19 Feb '16

V2: - patch: ALSA: pcm: add IEC958 channel status control helper - Return 1 instead of 0 in snd_pcm_iec958_put - Add .access field in control structure - I have kept condition on mutex for flexibility (but could be cleaned to force user to use a mutex) - patch: video: hdmi: add helper function for N and CTS - Duplicate function to have a separte treatment for coherent and non-coherent clcocks - Add ratio fiel for alternate CTS value - Clock frequency in Hz for TMDS and audio clocks - Add information concerning clocks and CTS calculation. V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Arnaud Pouliquen (6): video: hdmi: add helper function for N and CTS ALSA: pcm: add IEC958 channel status control helper ASoC: core: add code to complete dai init after pcm creation drm: sti: Add ASoC generic hdmi codec support. ASoc: hdmi-codec: add IEC control. ARM: DT: b2120: add audio HDMI dai link in audio card arch/arm/boot/dts/stihxxx-b2120.dtsi | 15 ++- drivers/gpu/drm/sti/Kconfig | 1 + drivers/gpu/drm/sti/sti_hdmi.c | 248 ++++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 10 ++ drivers/video/hdmi.c | 222 +++++++++++++++++++++++++++++++ include/linux/hdmi.h | 22 ++++ include/sound/hdmi-codec.h | 1 + include/sound/pcm_iec958.h | 16 +++ include/sound/soc-dai.h | 7 + sound/core/pcm_iec958.c | 99 ++++++++++++++ sound/soc/codecs/hdmi-codec.c | 59 +++++++-- sound/soc/soc-core.c | 14 ++ 12 files changed, 682 insertions(+), 32 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2 0/3] ASoC: topology: Add support for FE DAI & DAI links
by mengdong.lin＠linux.intel.com 16 Feb '16

16 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows topology to create FE DAI and DAI links from the PCM topology objects defined by the user space. History: v2: Remove first 2 patches in v1 that were accepted. Add Ack by Takashi to patch 'ALSA: pcm: Add snd_pcm_rate_range_to_bits()' and add his comments on the function's assumption to c file. Mengdong Lin (3): ALSA: pcm: Add snd_pcm_rate_range_to_bits() ASoC: topology: Add FE DAIs dynamically ASoC: topology: Add FE DAI links dynamically include/sound/pcm.h | 2 + include/sound/soc-topology.h | 21 +++-- include/sound/soc.h | 2 +- sound/core/pcm_misc.c | 30 +++++++ sound/soc/soc-topology.c | 195 ++++++++++++++++++++++++++++++++++--------- 5 files changed, 197 insertions(+), 53 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v2 00/14] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 15 Feb '16

15 Feb '16

This patch series adds DP audio and hotplug notification support. On Skylake two DP ports are available and to enable DP on both ports all pins need to be enabled. There is a special vendor widget which need to be programmed to enable all pins and converters. This series adds hotplug notification, read/set constraint based on ELD, enable all pin/cvts, DP1.2, programs the audio infoframe for DP. There is a one to one mapping between converter and stream, so the dais are created based on the no of streams supported on hdmi codec. Even though cvts can be mapped dynamically to the streams, currently it is statically mapped as simultaneous playback on both DP and HDMI is not supported as of now. Pin muxes and controls are created dynamically to map converter to pin widget. So at run time specific pin is mapped to the dai based on the control selected (based on the display type DP/HDMI connected). Finally the DP audio infoframe programming is added to support the DP feature. Also with hotplug notification support, ELD is read and capabilities are set for rate, formats and channels. drm_eld sound/core framework is updated to limit the formats based on ELD. There are few fixes one fixing the static checker warning and other one not to fail if no connection list is found for a pin widget. Pls note, the 11th patch is adding a small macro for getting connection type in drm header, we have CCed drm folks on that and this one. Pls ack so that we can have this series merged thru sound trees changes in v2: - Address review comments - Dropped devm_xxx for cvt and pin memory allocation. - Dropped the ELD helper APIs from core and copied to driver. - Updated notification patch to add PM fixes from legacy hda. - commit message and change log updated for display power fix. - Updated DRM helper APIs and the caller Jeeja KP (1): ASoC: hdac_hdmi: Add codec suspend/resume handler Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep display active while enumerating codec Subhransu S. Prusty (12): ASoC: hdac_hdmi: Fix to check num nodes correctly ASoC: hdac_hdmi: Fix to warn instead of err for no connected nids ASoC: hdac_hdmi - Use list to add pins and converters ASoC: hdac_hdmi: Add hotplug notification and read eld ALSA: pcm: Add DRM helper to set constraint for format ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi - create dais based on number of streams ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Assign pin for stream based on dapm connection drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio include/drm/drm_edid.h | 12 + sound/core/pcm_drm_eld.c | 42 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1019 ++++++++++++++++++++++++++++++++++++------ 4 files changed, 925 insertions(+), 149 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: topology: Add support for FE DAI & DAI links
by mengdong.lin＠linux.intel.com 15 Feb '16

15 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows topology to create FE DAI and DAI links from the PCM topology objects defined by the user space. Mengdong Lin (5): ASoC: Define soc_add_dai() to add a DAI to a component ASoC: Support registering a DAI dynamically ALSA: pcm: Add snd_pcm_rate_range_to_bits() ASoC: topology: Add FE DAIs dynamically ASoC: topology: Add FE DAI links dynamically include/sound/pcm.h | 2 + include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 21 +++-- include/sound/soc.h | 5 +- sound/core/pcm_misc.c | 27 ++++++ sound/soc/soc-core.c | 128 ++++++++++++++++++++-------- sound/soc/soc-topology.c | 195 ++++++++++++++++++++++++++++++++++--------- 7 files changed, 291 insertions(+), 88 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH V2 0/3] ASOC: bcm2835: move bcm2835-i2s to use clock framework
by kernel＠martin.sperl.org 13 Feb '16

13 Feb '16

From: Martin Sperl <kernel(a)martin.sperl.org> This patchset enables the bcm2835-i2s driver to use the clock framework which was introduced with commit 94cb7f76caa0 ("ARM: bcm2835: Switch to using the new clock driver support."). This commit resulted in the fact that the bcm2835-i2s driver was no longer working due to some register addresses used by 2 drivers (clk-bcm2835 and bcm2835-i2s). This patchset requires that the patchset for PCM-clock support as well as fractional/mash support is applied to the clk-bcm2835 driver and the corresponding device tree, but as the current version of the driver is not working, that should not be a problem. Note that there is one change: right now the current driver tries to calculate an optimal bclk_ratio based on its knowledge that it is using the 19.2Mhz oscillator. This computation would recommend the use of 40 or 80 bits instead of 32/64 bits that are required. Some of the DACs can not handle this, so most downstream drivers would set snd_soc_dai_set_bclk_ratio explicitly to disable this "non-power-of-2" automatic selection. So it seems wise to leave it out of the current patchset. If it is deemed necessary, then I can provide a separate patch that implements this again. Changelog: V1->V2: * moving clock patches into a separate patchset, which fixes also other issues in the clock framework * remove unnecessary bclk_size assignments Martin Sperl (3): ASoC: bcm2835: move to use the clock framework ARM: bcm2835: I2S: use new register-range and clock framework dt-bindings: bsm2835: fix bindings documentation to use new clock framework .../devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 7 +- arch/arm/boot/dts/bcm2835.dtsi | 5 +- sound/soc/bcm/bcm2835-i2s.c | 284 +++++--------------- 3 files changed, 69 insertions(+), 227 deletions(-) -- 1.7.10.4

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[alsa-devel] Usb audio - unsuported sampling rates
by Gordon Dilan 12 Feb '16

12 Feb '16

Hello, I'm using an usb mic which(according to it's producer should support up to 48kHz sampling rate). The problem is ALSA allows me to use only 16 kHz for this device. I do not want to resample the data, I want to sample it at 48kHz in order to be able to record some high pitch sounds 18-20kHz. I would want to try to do any modifications necessary in order to try support 48kHz, but I'm not sure where to start. Some information: cat /proc/asound/modules ... 1 snd_usb_audio arecord --list-devices **** List of CAPTURE Hardware Devices **** .... card 1: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 when I try to use: arecord -v -D hw:1,0 -f S16_LE -c 4 -r 48000 gigi.wav I get this: Recording WAVE 'gigi.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Warning: rate is not accurate (requested = 48000Hz, got = 16000Hz) please, try the plug plugin Hardware PCM card 1 'USB Camera-B4.09.24.1' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 1 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 Any sampling rate I would try higher or lower than 16kHz, I will always get 16000 kHz. Some more information: lsusb Bus 001 Device 007: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 7, If 0, Class=vend., Driver=ov534, 480M |__ Port 2: Dev 7, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 7, If 2, Class=audio, Driver=snd-usb-audio, 480M At the moment I'm looking inside ALSA's source code to try to see from where this limitation of 16000kHz for my device appears, but I have to admit that I'm kind of lost. Any pointers on what direction should I go to try make my desired rate supported would be greatly appreciated! Thank you and best regards, Gordon

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