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

[alsa-devel] [PATCH] ucm: Add ucm files for DB410c board.
by Riku Voipio 10 Aug '16

10 Aug '16

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> DB410c board has support for both Digital and Analog audio. Digital audio is over HDMI and analog is over codec chip integrated inside the APQ8016 SOC. It can support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary - 2 Digital Microphones. - Earpiece. - Headset. - Loud Speaker. - HDMI. [Riku: squashed Srinivas's patches together and converted spaces to tabs] Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Riku Voipio <riku.voipio(a)linaro.org> Cc: nicolas.dechesne(a)linaro.org --- configure.ac | 1 + src/conf/ucm/DB410c/DB410c.conf | 8 ++ src/conf/ucm/DB410c/HDMI | 31 +++++++ src/conf/ucm/DB410c/HiFi | 185 ++++++++++++++++++++++++++++++++++++++++ src/conf/ucm/DB410c/Makefile.am | 4 + src/conf/ucm/Makefile.am | 2 +- 6 files changed, 230 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/DB410c/DB410c.conf create mode 100644 src/conf/ucm/DB410c/HDMI create mode 100644 src/conf/ucm/DB410c/HiFi create mode 100644 src/conf/ucm/DB410c/Makefile.am diff --git a/configure.ac b/configure.ac index f592e8b..014af5f 100644 --- a/configure.ac +++ b/configure.ac @@ -675,6 +675,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ src/conf/ucm/chtrt5645/Makefile \ + src/conf/ucm/DB410c/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/DB410c/DB410c.conf b/src/conf/ucm/DB410c/DB410c.conf new file mode 100644 index 0000000..590278f --- /dev/null +++ b/src/conf/ucm/DB410c/DB410c.conf @@ -0,0 +1,8 @@ +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play HiFi quality Music." +} +SectionUseCase."HDMI" { + File "HDMI" + Comment "HDMI output." +} diff --git a/src/conf/ucm/DB410c/HDMI b/src/conf/ucm/DB410c/HDMI new file mode 100644 index 0000000..8112bc0 --- /dev/null +++ b/src/conf/ucm/DB410c/HDMI @@ -0,0 +1,31 @@ +# Use case configuration for DB410c board. +# Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +SectionVerb { + EnableSequence [ + ] + + DisableSequence [ + ] + Value { + TQ "HiFi" + PlaybackPCM "plughw:0,0" + } +} + +SectionDevice."HDMI-stereo" { + #Name "HDMI-stereo" + Comment "HDMI Digital Stereo Output" + + EnableSequence [ + cdev "hw:0" + ] + + DisableSequence [ + cdev "hw:0" + ] + + Value { + PlaybackChannels "2" + } +} diff --git a/src/conf/ucm/DB410c/HiFi b/src/conf/ucm/DB410c/HiFi new file mode 100644 index 0000000..f9cbcf0 --- /dev/null +++ b/src/conf/ucm/DB410c/HiFi @@ -0,0 +1,185 @@ +# Use case configuration for DB410c board. +# Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +SectionVerb { + + EnableSequence [ + ] + + DisableSequence [ + ] + + # ALSA PCM + Value { + # ALSA PCM device for HiFi + PlaybackPCM "plughw:0,1" + CapturePCM "plughw:0,2" + } +} + + +SectionDevice."Speaker" { + Comment "Speaker playback" + + ConflictingDevice [ + "Headphones" + "Earpiece" + ] + + EnableSequence [ + cdev "hw:0" + cset "name='SPK DAC Switch' 1" + cset "name='RX3 MIX1 INP1' RX1" + ## gain to 0dB + cset "name='RX3 Digital Volume' 128" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='SPK DAC Switch' 0" + cset "name='RX3 MIX1 INP1' ZERO" + ] + + Value { + PlaybackChannels "2" + } +} + + +SectionDevice."Headphones" { + Comment "Headphones playback" + + ConflictingDevice [ + "Speaker" + "Earpiece" + ] + EnableSequence [ + cdev "hw:0" + cset "name='RX1 MIX1 INP1' RX1" + cset "name='RX2 MIX1 INP1' RX2" + cset "name='RDAC2 MUX' RX2" + cset "name='HPHL' 1" + cset "name='HPHR' 1" + ## gain to 0dB + cset "name='RX1 Digital Volume' 128" + ## gain to 0dB + cset "name='RX2 Digital Volume' 128" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='RX1 Digital Volume' 0" + cset "name='RX2 Digital Volume' 0" + cset "name='HPHL' 0" + cset "name='HPHR' 0" + cset "name='RDAC2 MUX' ZERO" + cset "name='RX1 MIX1 INP1' ZERO" + cset "name='RX2 MIX1 INP1' ZERO" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Earpiece" { + Comment "Earpiece playback" + + ConflictingDevice [ + "Speaker" + "Headphones" + ] + EnableSequence [ + cdev "hw:0" + ] + + DisableSequence [ + cdev "hw:0" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Handset" { + Comment "Headset Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC2" + cset "name='ADC2 Volume' 8" + cset "name='ADC2 MUX' INP2" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='ADC2 MUX' ZERO" + cset "name='ADC2 Volume' 0" + cset "name='DEC1 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."Primarymic" { + Comment "Primary Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC1" + cset "name='ADC1 Volume' 8" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + cset "name='ADC1 Volume' 0" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."Secondarymic" { + Comment "Secondary Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC2" + cset "name='ADC2 Volume' 8" + cset "name='ADC2 MUX' INP2" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + cset "name='ADC2 Volume' 0" + cset "name='ADC2 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."DMIC" { + Comment "Digital Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' DMIC1" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} diff --git a/src/conf/ucm/DB410c/Makefile.am b/src/conf/ucm/DB410c/Makefile.am new file mode 100644 index 0000000..e10a136 --- /dev/null +++ b/src/conf/ucm/DB410c/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/DB410c +ucm_DATA = DB410c.conf HDMI HiFi +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 8865770..7bf432e 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 DB410c -- 2.1.4

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[alsa-devel] [PATCH] sound: soc: atmel: Don't unconditionally reset SSC on stream startup
by Peter Meerwald 10 Aug '16

10 Aug '16

From: Christoph Huber <c.huber(a)bct-electronic.com> commit cbaadf0f90d6 (ASoC: atmel_ssc_dai: refactor the startup and shutdown) refactored code such that the SSC is reset on every startup; this breaks duplex audio (e.g. first start audio playback, then start record, causing the playback to stop/hang) Signed-off-by: Christoph Huber <c.huber(a)bct-electronic.com> Signed-off-by: Peter Meerwald-Stadler <p.meerwald(a)bct-electronic.com> Cc: Songjun Wu <songjun.wu(a)microchip.com> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..b3e9f0c 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -298,8 +298,9 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, clk_enable(ssc_p->ssc->clk); ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); - /* Reset the SSC to keep it at a clean status */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Reset the SSC if it is not initialized to keep it at a clean status */ + if(!ssc_p->initialized) + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; -- 2.1.4

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[alsa-devel] [PATCH v5 1/2] mfd: Add DT bindings documentation for Samsung Exynos LPASS
by Sylwester Nawrocki 10 Aug '16

10 Aug '16

This patch adds documentation of the DT bindings for the Samsung Exynos SoC Low Power Audio Subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v4: - text reformatting, added VIC to the list of sub-devices, relative paths used for the reference DT binding documentation files, added example dts. Changes since v3: - moved to Documentation/devicetree/bindings/mfd, - added #address-cells, #size-cells, ranges as required properties, - added description of the IP block subnodes and links to their binding's documentation. Changes since v2: - none. --- .../bindings/mfd/samsung,exynos5433-lpass.txt | 70 ++++++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt diff --git a/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt new file mode 100644 index 0000000..c110e11 --- /dev/null +++ b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt @@ -0,0 +1,70 @@ +Samsung Exynos SoC Low Power Audio Subsystem (LPASS) + +Required properties: + + - compatible : "samsung,exynos5433-lpass" + - reg : should contain the LPASS top SFR region location + and size + - samsung,pmu-syscon : the phandle to the Power Management Unit node + - #address-cells : should be 1 + - #size-cells : should be 1 + - ranges : must be present + +Each IP block of the Low Power Audio Subsystem should be specified as +an optional sub-node. For "samsung,exynos5433-lpass" compatible this includes: +UART, SLIMBUS, PCM, I2S, DMAC, Timers 0...4, VIC, WDT 0...1 devices. + +Bindings of the sub-nodes are described in: + ../serial/samsung_uart.txt + ../sound/samsung-i2s.txt + ../dma/arm-pl330.txt + + +Example: + +audio-subsystem { + compatible = "samsung,exynos5433-lpass"; + reg = <0x11400000 0x100>, <0x11500000 0x08>; + samsung,pmu-syscon = <&pmu_system_controller>; + #address-cells = <1>; + #size-cells = <1>; + ranges; + + adma: adma@11420000 { + compatible = "arm,pl330", "arm,primecell"; + reg = <0x11420000 0x1000>; + interrupts = <0 73 0>; + clocks = <&cmu_aud CLK_ACLK_DMAC>; + clock-names = "apb_pclk"; + #dma-cells = <1>; + #dma-channels = <8>; + #dma-requests = <32>; + }; + + i2s0: i2s0@11440000 { + compatible = "samsung,exynos7-i2s"; + reg = <0x11440000 0x100>; + dmas = <&adma 0 &adma 2>; + dma-names = "tx", "rx"; + interrupts = <0 70 0>; + clocks = <&cmu_aud CLK_PCLK_AUD_I2S>, + <&cmu_aud CLK_SCLK_AUD_I2S>, + <&cmu_aud CLK_SCLK_I2S_BCLK>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + status = "disabled"; + }; + + serial_3: serial@11460000 { + compatible = "samsung,exynos5433-uart"; + reg = <0x11460000 0x100>; + interrupts = <0 67 0>; + clocks = <&cmu_aud CLK_PCLK_AUD_UART>, + <&cmu_aud CLK_SCLK_AUD_UART>; + clock-names = "uart", "clk_uart_baud0"; + pinctrl-names = "default"; + pinctrl-0 = <&uart_aud_bus>; + status = "disabled"; + }; + }; -- 1.9.1

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[alsa-devel] Underrun with 2 periods on skylake HDA driver
by Dharageswari R 10 Aug '16

10 Aug '16

Hello, While doing the validation of Skylake HDA driver using aplay and following parameters, we noticed underrun issues. aplay -Dhw:0,0,0 -vv s_48K_16Bit_Sunshine.wav -F 100000 -B 200000 -d 20 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 9600 period_size : 4800 period_time : 100000 tstamp_mode : NONE period_step : 1 avail_min : 4800 period_event : 0 start_threshold : 9600 stop_threshold : 9600 silence_threshold: 0 silence_size : 0 boundary : 5404319552844595200 appl_ptr : 0 hw_ptr : 0 After doing further debugging, what I found is 1.DMA complete interrupt occurs for the first period. 2.Driver reads DMA position in buffer pointer and gets value of 4784. 3.Driver calls snd_pcm_period_elapsed() function to report period elapsed interrupt to application with DMA position = 4784. 4.snd_pcm_lib_write1() is waiting for “avail_min” free space to be available. In this case the avail_min = 4800 and at this interrupt the avail = 4874 and so it does not write any data. 5.DMA complete interrupt occurs for the second period. 6.Driver reads DMA position in the buffer pointer and gets value of 9600. 7.Driver calls snd_pcm_period_elapsed() function to report period elapsed interrupt. 8.Now the snd_pcm_lib_write1() gets avail > avail_min but it’s too late because DMA has already started playing data from first period, which is not yet written by the application. 9.This causes underrun. Logs: 857.870974: hwptr: pcmC0D0p/sub0: IRQ: pos=4800, old=19200, base=19200, period=4800, buf=9600 857.871237: hwptr: pcmC0D0p/sub0: POS: pos=4800, old=24000, base=19200, period=4800, buf=9600 857.970972: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=24000, base=19200, period=4800, buf=9600 857.971101: hwptr: pcmC0D0p/sub0: POS: pos=0, old=28800, base=28800, period=4800, buf=9600 858.070971: hwptr: pcmC0D0p/sub0: IRQ: pos=4784, old=28800, base=28800, period=4800, buf=9600 858.170968: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=33584, base=28800, period=4800, buf=9600 858.170988: xrun: pcmC0D0p/sub0: XRUN: old=38400, base=38400, period=4800, buf=9600 858.172240: hwptr: pcmC0D0p/sub0: POS: pos=96, old=0, base=0, period=4800, buf=9600 858.269867: hwptr: pcmC0D0p/sub0: IRQ: pos=4798, old=96, base=0, period=4800, buf=9600 858.369902: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=4798, base=0, period=4800, buf=9600 I checked with HDA architects and they are saying that when the DMA completion interrupt occurs there can be slight delay before the DMA Position In Buffer Pointer gets updated. So driver is expected to refill data based on whatever content is reflected in the DMA Position In Buffer. What is the right way to fix this issue ? Thanks, Dhara.R --

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[alsa-devel] [PATCH v4 0/5] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 10 Aug '16

10 Aug '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. The parsing of caps is made common and moved up from extended hda to hdac. The SKL driver and hdac extended core has been updated for this. Mark, The update of the driver needs to update Skl driver, so if you can ACK the 2nd patch, it can go along with extended audio change thru ALSA tree. Changes in v4: - fix code style - use hdac_readl/writel - add cpufeatures header Guneshwor Singh (2): ALSA - hda: Add support for parsing new HDA capabilities ALSA - hda: Add support for link audio time reporting Vinod Koul (3): ALSA: hda - move bus_parse_capabilities to core ALSA: convert users to core bus_parse_capabilities ALSA - Ext hda: remove bus_parse_capabilities include/sound/hda_register.h | 36 ++++++ include/sound/hdaudio.h | 13 +++ include/sound/hdaudio_ext.h | 12 -- sound/hda/ext/hdac_ext_controller.c | 91 ++------------- sound/hda/ext/hdac_ext_stream.c | 46 ++++---- sound/hda/hdac_controller.c | 75 ++++++++++++ sound/pci/hda/hda_controller.c | 203 ++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 3 + sound/pci/hda/hda_intel.c | 17 +++ sound/soc/intel/skylake/skl-messages.c | 4 +- sound/soc/intel/skylake/skl-pcm.c | 8 +- sound/soc/intel/skylake/skl.c | 6 +- 12 files changed, 386 insertions(+), 128 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: core: fix shift used for second item in snd_soc_get_enum_double
by jjassal 09 Aug '16

09 Aug '16

Incorrect shift value was being used to extract the second item. Signed-off-by: jjassal <jjassal(a)opensource.wolfsonmicro.com> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a513a34..9fc1a7b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -77,7 +77,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_r) & e->mask; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } -- 2.1.4

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[alsa-devel] [PATCH 00/39] seq: obsolete change of address limit
by Takashi Sakamoto 09 Aug '16

09 Aug '16

Hi, ALSA sequencer core has two types of client; user application and kernel driver. The core allows both types of client to do relevant operations, thus it's required for the core to handle data in both user and kernel spaces. Currently, this is achieved by changing address limit of running task. This is a well-known technique to suppress address check, while it's just a suppression and unfriendly to readers or static code parsers. This patchset obsoletes the usage. In process context, data in user space is once copied to kernel space, then operated and copied to user space. As a result, actual operations for each ioctl command handle data in kernel space, and '__user' qualifier is useless. In this series, patch 1-8 just apply above design. The rest changes each operation following to the design. Therefore, static code parser like sparce generates warnings temporarily in a way to apply these patches. There's a concern of this solution. The data for ioctl is always copied to kernel space or to kernel space even when it's read-only or write-only. But this brings no severe issue as long as I read ALSA sequencer core. (This patchset is a part of my work to introduce EPIPE into ALSA rawmidi core.) Takashi Sakamoto (39): ALSA: seq: add const qualifier to table of functions for ioctl ALSA: seq: apply shorter name for file local functions ALSA: seq: fulfill callback entry for ioctl ALSA: seq: copy ioctl data from user space to kernel stack ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: obsolete address mode in compatibility layer ALSA: seq: obsolete change of address limit in in-kernel path for ioctl ALSA: seq: obsolete address limit helper ALSA: seq: optimize pversion function to new design ALSA: seq: optimize client_id function to new design ALSA: seq: optimize system_info function to new design ALSA: seq: optimize running mode function to new design ALSA: seq: optimize client_info function to new design ALSA: seq: optimize set_client_info function to new design ALSA: seq: optimize create_port function to new design ALSA: seq: optimize delete_port function to new design ALSA: seq: optimize get_port_info function to new design ALSA: seq: optimize seq_port_info function to new design ALSA: seq: optimize subscribe_port function to new design ALSA: seq: optimize unsubscribe_port function to new design ALSA: seq: optimize create_queue function to new design ALSA: seq: optimize delete_queue function to new design ALSA: seq: optimize get_queue_info function to new design ALSA: seq: optimize seq_queue_info function to new design ALSA: seq: optimize get_named_queue function to new design ALSA: seq: optimize get_queue_status function to new design ALSA: seq: optimize get_queue_tempo function to new design ALSA: seq: optimize set_queue_tempo function to new design ALSA: seq: optimize get_queue_timer function to new design ALSA: seq: optimize seq_queue_timer function to new design ALSA: seq: optimize get_queue_client function to new design ALSA: seq: optimize set_queue_client function to new design ALSA: seq: optimize get_client_pool function to new design ALSA: seq: optimize seq_client_pool function to new design ALSA: seq: optimize remove_events function to new design ALSA: seq: optimize get_subscription function to new design ALSA: seq: optimize query_subs function to new design ALSA: seq: optimize query_next_client function to new design ALSA: seq: optimize query_next_port function to new design sound/core/seq/seq_clientmgr.c | 854 +++++++++++++++++++---------------------- sound/core/seq/seq_compat.c | 26 +- 2 files changed, 407 insertions(+), 473 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v3 0/5] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 09 Aug '16

09 Aug '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. The parsing of caps is made common and moved up from extended hda to hdac. The SKL driver and hdac extended core has been updated for this. Mark, The update of the driver needs to update Skl driver, so if you can ACK the 2nd patch, it can go along with extended audio change thru ALSA tree. Changes in v3: moved extended hda to hdac update SKL drv to use new hdac caps update patchset for HDA drv to use new caps parsing Guneshwor Singh (2): ALSA - hda: Add support for parsing new HDA capabilities ALSA - hda: Add support for link audio time reporting Vinod Koul (3): ALSA: hda - move bus_parse_capabilities to core ALSA: convert users to core bus_parse_capabilities ALSA - Ext hda: remove bus_parse_capabilities include/sound/hda_register.h | 36 ++++++ include/sound/hdaudio.h | 13 +++ include/sound/hdaudio_ext.h | 12 -- sound/hda/ext/hdac_ext_controller.c | 91 ++------------- sound/hda/ext/hdac_ext_stream.c | 46 ++++---- sound/hda/hdac_controller.c | 76 +++++++++++++ sound/pci/hda/hda_controller.c | 200 ++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 3 + sound/pci/hda/hda_intel.c | 16 +++ sound/soc/intel/skylake/skl-messages.c | 4 +- sound/soc/intel/skylake/skl-pcm.c | 8 +- sound/soc/intel/skylake/skl.c | 6 +- 12 files changed, 383 insertions(+), 128 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: remove snd_soc_pcm_set/get_drvdata()
by Kuninori Morimoto 09 Aug '16

09 Aug '16

Hi Mark snd_soc_pcm_set_drvdata() will set driver data to rtd->dev, but driver data of rtd->dev is already used as "rtd" on soc_post_component_init(). static int soc_post_component_init(xxx) { ... dev_set_drvdata(rtd->dev, rtd); ... } But, ASoC has snd_soc_pcm_set/get_drvdata() to set rtd->dev. It will overwrite existing data and will break system. This patch tries to solve this issue. Intel haswell and qcom is using these functions. ** Note ** I don't have these boards, thus, I did compile test only. Kuninori Morimoto (3): ASoC: sst-haswell-pcm: don't use snd_soc_pcm_set_drvdata() ASoC: lpass-platform: don't use snd_soc_pcm_set_drvdata() ASoC: remove snd_soc_pcm_set/get_drvdata() include/sound/soc.h | 11 ----------- sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - sound/soc/qcom/lpass-platform.c | 14 +++++++------- sound/soc/qcom/lpass.h | 2 ++ 4 files changed, 9 insertions(+), 19 deletions(-)

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[alsa-devel] [PATCH 0/6] ASoC: simple-card: add init_dai / canonicalize_dailink
by Kuninori Morimoto 09 Aug '16

09 Aug '16

Hi Mark These add init_dai / canonicalize_dailink to simple utiles. These are part of simple-card cleanup Kuninori Morimoto (6): ASoC: simple-card-utils: add asoc_simple_card_init_dai() ASoC: simple-card: use asoc_simple_card_init_dai() ASoC: rsrc-card: use asoc_simple_card_init_dai() ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() include/sound/simple_card_utils.h | 5 +++++ sound/soc/generic/simple-card-utils.c | 42 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 47 +++++------------------------------------------ sound/soc/sh/rcar/rsrc-card.c | 32 +++++--------------------------- 4 files changed, 57 insertions(+), 69 deletions(-) -- 1.9.1

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