March 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] rawmidi: virtual: fix reading into a small buffer
by Clemens Ladisch 20 Mar '17

20 Mar '17

In the special case for handling partial messages, the pointer calculations were wrong, which would result in data corruption. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- alsa-lib/src/rawmidi/rawmidi_virt.c +++ alsa-lib/src/rawmidi/rawmidi_virt.c @@ -263,8 +263,8 @@ static ssize_t snd_rawmidi_virtual_read(snd_rawmidi_t *rmidi, void *buffer, size } size1 = virt->in_buf_size - virt->in_buf_ofs; if ((size_t)size1 > size) { - virt->in_buf_ofs += size1 - size; - memcpy(buffer, virt->in_buf_ptr, size); + memcpy(buffer, virt->in_buf_ptr + virt->in_buf_ofs, size); + virt->in_buf_ofs += size; result += size; break; }

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[alsa-devel] [PATCH] ALSA: ctxfi: Fix the incorrect check of dma_set_mask() call
by Takashi Iwai 20 Mar '17

20 Mar '17

In the commit [15c75b09f8d1: ALSA: ctxfi: Fallback DMA mask to 32bit], I forgot to put "!" at dam_set_mask() call check in cthw20k1.c (while cthw20k2.c is OK). This patch fixes that obvious bug. (As a side note: although the original commit was completely wrong, it's still working for most of machines, as it sets to 32bit DMA mask in the end. So the bug severity is low.) Fixes: 15c75b09f8d1 ("ALSA: ctxfi: Fallback DMA mask to 32bit") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ctxfi/cthw20k1.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index ab4cdab5cfa5..79edd88d5cd0 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1905,7 +1905,7 @@ static int hw_card_start(struct hw *hw) return err; /* Set DMA transfer mask */ - if (dma_set_mask(&pci->dev, DMA_BIT_MASK(dma_bits))) { + if (!dma_set_mask(&pci->dev, DMA_BIT_MASK(dma_bits))) { dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(dma_bits)); } else { dma_set_mask(&pci->dev, DMA_BIT_MASK(32)); -- 2.11.1

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[alsa-devel] [PATCH] ALSA: x86: Select CONFIG_HDMI_LPE_AUDIO as default
by Ian W MORRISON 20 Mar '17

20 Mar '17

The upstream kernel builds for distributions such as Ubuntu which now includes binary packages for v4.11 mainline kernel release candidates are promoted as a way of testing upstream kernels to to confirm that upstream has fixed a specific issue (see https://wiki.ubuntu.com/Kernel/MainlineBuilds). Unfortunately the long awaited patch for providing HDMI audio support for Bay Trail and Cherry Trail devices does not include this support through a module built by default. By defining the default of the two associated CONFIG settings (SND_X86 and HDMI_LPE_AUDIO) as a module, upstream kernel builds would automatically provide the much desired HDMI audio support by default. This patch uses a Kconfig 'default' statement to include the driver as default. Signed-off-by: Ian W Morrison <linuxium(a)linuxium.com.au> --- sound/x86/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 84c8f8fc..439850c 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_X86 tristate "X86 sound devices" depends on X86 + default m ---help--- X86 sound devices that don't fall under SoC or PCI categories @@ -9,6 +10,7 @@ if SND_X86 config HDMI_LPE_AUDIO tristate "HDMI audio without HDaudio on Intel Atom platforms" depends on DRM_I915 + default m select SND_PCM help Choose this option to support HDMI LPE Audio mode -- 1.9.1

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[alsa-devel] [PATCH] ASoC: omap: constify snd_soc_ops structures
by Bhumika Goyal 20 Mar '17

20 Mar '17

Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. Cross compiled the .o files for arm architecture. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 2 +- sound/soc/omap/omap-twl4030.c | 2 +- sound/soc/omap/omap3pandora.c | 2 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/rx51.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 25a33e9..d565102 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -49,7 +49,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops am3517evm_ops = { +static const struct snd_soc_ops am3517evm_ops = { .hw_params = am3517evm_hw_params, }; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fdecb70..71e5f31 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -124,7 +124,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; } -static struct snd_soc_ops n810_ops = { +static const struct snd_soc_ops n810_ops = { .startup = n810_startup, .hw_params = n810_hw_params, .shutdown = n810_shutdown, diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 89fe95e..614b18d 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -70,7 +70,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops omap_abe_ops = { +static const struct snd_soc_ops omap_abe_ops = { .hw_params = omap_abe_hw_params, }; diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 7431314..a24b0de 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -73,7 +73,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream, return snd_soc_runtime_set_dai_fmt(rtd, fmt); } -static struct snd_soc_ops omap_twl4030_ops = { +static const struct snd_soc_ops omap_twl4030_ops = { .hw_params = omap_twl4030_hw_params, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 732e749..4e3de71 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -184,7 +184,7 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_ops omap3pandora_ops = { +static const struct snd_soc_ops omap3pandora_ops = { .hw_params = omap3pandora_hw_params, }; diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index aa4053b..e409677 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -68,7 +68,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, return err; } -static struct snd_soc_ops osk_ops = { +static const struct snd_soc_ops osk_ops = { .startup = osk_startup, .hw_params = osk_hw_params, .shutdown = osk_shutdown, diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index a768457..66ac358 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -123,7 +123,7 @@ static int rx51_hw_params(struct snd_pcm_substream *substream, SND_SOC_CLOCK_IN); } -static struct snd_soc_ops rx51_ops = { +static const struct snd_soc_ops rx51_ops = { .startup = rx51_startup, .hw_params = rx51_hw_params, }; -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: rt5665: fix wrong shift rt5665_if2_1_adc_in_enum
by Bard Liao 20 Mar '17

20 Mar '17

The shift is RT5665_IF2_1_ADC_IN_SFT not RT5665_IF3_ADC_IN_SFT. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 5545d08..21b07ca 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2256,7 +2256,7 @@ static const char * const rt5665_if2_1_adc_in_src[] = { static const SOC_ENUM_SINGLE_DECL( rt5665_if2_1_adc_in_enum, RT5665_DIG_INF2_DATA, - RT5665_IF3_ADC_IN_SFT, rt5665_if2_1_adc_in_src); + RT5665_IF2_1_ADC_IN_SFT, rt5665_if2_1_adc_in_src); static const struct snd_kcontrol_new rt5665_if2_1_adc_in_mux = SOC_DAPM_ENUM("IF2_1 ADC IN Source", rt5665_if2_1_adc_in_enum); -- 2.7.4

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[alsa-devel] USB Focusrite Scarlett 6i6 2nd Gen - Mixer not working (new VID, quirks and probably matrix needed)
by Matthias Andree 19 Mar '17

19 Mar '17

Greetings, I recently acquired a Focusrite Scarlett 6i6 *2nd Gen*. It appears to work for sound output (haven't tested much else yet), but mixing/routing won't work, and to add insult to injury, Pulseaudio misrepresents it as a multi-channel sound card, defaults to 5.1 and deprives it of the bass. Selecting 5.0 improves the situation as a workaround for stereo output. Hardware differences between 1st and 2nd Gen aside (where pad and inst/line switches moved into software), the interface does not create useful mixer entries. Digging around I figured we didn't have new stuff in the current kernel Git tree, so I chose to continue with what I had and found: in Ubuntu 16.04's 4.8.0 HWE kernel "Advanced Linux Sound Architecture Driver Version k4.8.0-42-generic.", I figured that the USB IDs (VID 0x1235 as usual, PID 0x8203 for the 6i6 2nd Gen) aren't recognized. I added them to the quirks section, and to the mixer_scarlett.c, to no avail. I get this on load, and the second message identically for each attempt to access the card through alsamixer, which alsamixer terminates with SIGPIPE. [10834.242351] usbcore: registered new interface driver snd-usb-audio [10834.329623] usb 2-6: cannot get ctl value: req = 0x81, wValue = 0x901, wIndex = 0x100, type = 4 With the patched module, sound stops after a few notes but I assume that's pulseaudio f*cking up, while it works with the unpatched module. So it appears we need to dig up and implement a bit of its innards and teach mixer_scarlett about it. How do we best go about getting this 2nd Gen Scarlett 6i6 fully supported? Do I need to sniff the USB communications on Windows while poking at the interface with its mixer control tool, or do we have other options? How did the 1st gen driver come to pass, and who did it and might be available for assistance? Is there any user-space poking we can do to dig up the needed information? Thanks. Cheers, Matthias

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[alsa-devel] [RESEND PATCH v2 0/2] Add STM32 SAI support
by olivier moysan 18 Mar '17

18 Mar '17

This patch-set handles the Serial Audio Interface (SAI) IP on STM32 platforms. The SAI IP is composed of two Sub-block A and B. (see the figure below) Each sub-block makes use of individual and shared resources. Shared resources: - Reset line. Reset common and sub-block registers. - Bus interface clocks. This clock is not exposed in device as it it managed by clock driver, according to kernel clocks gating. - Common register. SAI IP exhibits a common configuration register to manage synchronization modes. NB: These modes are not yet implemented in this version of the SAI driver. - Interrupt. Sub-blocks have their own interrupt status registers but they share the same interrupt line. Individual resources: - Register set - DMA request line. - Communication interface. Each sub-block has its own GPIOs and associated bus lines. - Kernel clock. Each sub-block has its own dedicated clock for its communication interface. To reflect this architecture Sub-block A and B are handled by 2 child devices. Sub-block A and B devices can be configured independently either as transmitter or receiver. A PCM device is associated to each sub-block. A sub-block has to select the appropriated parent clock at runtime, depending on the audio stream sampling rate to be played or captured. Two parent clocks must be provided to support sampling rates multiples of 8 kHz or 11.025kHz. Interface clock +--------------------------------+ +---------------> | SAI IP | Reset | +-----------------+ | +---------------> | |common registers | | | +-----------------+ | | | | +----------------------+ | | | Sub-block A | | Kernel clock A | | | | Bus A +--------------------> | +----------------+ +----------------> | | |A registers | | | | | +----------------+ | | | +----------------------+ | | | | +----------------------+ | | | Sub-block B | | Kernel clock B | | | | Bus B +--------------------> | +----------------+ +-----------------> | | |B registers | | | | | +----------------+ | | | +----------------------+ | | | +--------------------------------+ Changes in v2: - Fix warnings olivier moysan (2): ASoC: stm32: add bindings for SAI ASoC: stm32: add SAI driver .../devicetree/bindings/sound/st,stm32-sai.txt | 79 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/stm/Kconfig | 8 + sound/soc/stm/Makefile | 6 + sound/soc/stm/stm32_sai.c | 123 +++ sound/soc/stm/stm32_sai.h | 202 +++++ sound/soc/stm/stm32_sai_sub.c | 899 +++++++++++++++++++++ 8 files changed, 1319 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_sai.c create mode 100644 sound/soc/stm/stm32_sai.h create mode 100644 sound/soc/stm/stm32_sai_sub.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: mediatek: constify snd_soc_ops structures
by Bhumika Goyal 18 Mar '17

18 Mar '17

Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 1e7e8d4..aa5b31b 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -129,7 +129,7 @@ static int mt2701_cs42448_fe_ops_startup(struct snd_pcm_substream *substream) return 0; } -static struct snd_soc_ops mt2701_cs42448_48k_fe_ops = { +static const struct snd_soc_ops mt2701_cs42448_48k_fe_ops = { .startup = mt2701_cs42448_fe_ops_startup, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 46c8e6a..e0c2b23 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -67,7 +67,7 @@ static int mt8173_max98090_hw_params(struct snd_pcm_substream *substream, SND_SOC_CLOCK_IN); } -static struct snd_soc_ops mt8173_max98090_ops = { +static const struct snd_soc_ops mt8173_max98090_ops = { .hw_params = mt8173_max98090_hw_params, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 467f704..5e383eb 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -75,7 +75,7 @@ static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mt8173_rt5650_rt5514_ops = { +static const struct snd_soc_ops mt8173_rt5650_rt5514_ops = { .hw_params = mt8173_rt5650_rt5514_hw_params, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 1b8b2a7..fed1f15 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -79,7 +79,7 @@ static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mt8173_rt5650_rt5676_ops = { +static const struct snd_soc_ops mt8173_rt5650_rt5676_ops = { .hw_params = mt8173_rt5650_rt5676_hw_params, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ba65f41..a784708 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -105,7 +105,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mt8173_rt5650_ops = { +static const struct snd_soc_ops mt8173_rt5650_ops = { .hw_params = mt8173_rt5650_hw_params, }; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: samsung: constify snd_soc_ops structures
by Bhumika Goyal 18 Mar '17

18 Mar '17

Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. Cross compiled the files arndale_rt5631.o, h1940_uda1380.o, jive_wm8750.o, littlemill.o, neo1973_wm8753.o and rx1950_uda1380.o for arm architecture. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/samsung/arndale_rt5631.c | 2 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/tm2_wm5110.c | 2 +- sound/soc/samsung/tobermory.c | 2 +- 15 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index ee1fda9..6f40587 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -53,7 +53,7 @@ static int arndale_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops arndale_ops = { +static const struct snd_soc_ops arndale_ops = { .hw_params = arndale_hw_params, }; diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 5f5825f..3c8d58a 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -120,7 +120,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops h1940_ops = { +static const struct snd_soc_ops h1940_ops = { .startup = h1940_startup, .hw_params = h1940_hw_params, }; diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 7fcb51f..529b10d 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -79,7 +79,7 @@ static int jive_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops jive_ops = { +static const struct snd_soc_ops jive_ops = { .hw_params = jive_hw_params, }; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 7cb204e..d057eae 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -135,7 +135,7 @@ static int littlemill_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops littlemill_ops = { +static const struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 65602b9..8e097f2 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -114,7 +114,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) /* * Neo1973 WM8753 HiFi DAI opserations. */ -static struct snd_soc_ops neo1973_hifi_ops = { +static const struct snd_soc_ops neo1973_hifi_ops = { .hw_params = neo1973_hifi_hw_params, .hw_free = neo1973_hifi_hw_free, }; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index fa096ab..324e16a 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -73,7 +73,7 @@ static struct snd_soc_jack_gpio hp_jack_gpios[] = { }, }; -static struct snd_soc_ops rx1950_ops = { +static const struct snd_soc_ops rx1950_ops = { .startup = rx1950_startup, .hw_params = rx1950_hw_params, }; diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index dcc008d..6de63f3 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -211,7 +211,7 @@ static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) return 0; } -static struct snd_soc_ops simtec_snd_ops = { +static const struct snd_soc_ops simtec_snd_ops = { .hw_params = simtec_hw_params, }; diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 81a7894..a6a4bc7 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -199,7 +199,7 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops s3c24xx_uda134x_ops = { +static const struct snd_soc_ops s3c24xx_uda134x_ops = { .startup = s3c24xx_uda134x_startup, .shutdown = s3c24xx_uda134x_shutdown, .hw_params = s3c24xx_uda134x_hw_params, diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 425ee2b..72408ae 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -77,7 +77,7 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, /* * SmartQ WM8987 HiFi DAI operations. */ -static struct snd_soc_ops smartq_hifi_ops = { +static const struct snd_soc_ops smartq_hifi_ops = { .hw_params = smartq_hifi_hw_params, }; diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index a2f2363..7fc7cc6 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -144,7 +144,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops smdk_spdif_ops = { +static const struct snd_soc_ops smdk_spdif_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 6e4dfa7..5c3980b 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -93,7 +93,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8580 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index ff57b19..307122f 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -80,7 +80,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8994 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 2e62149..7bbb822 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -91,7 +91,7 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops smdk_wm8994_pcm_ops = { +static const struct snd_soc_ops smdk_wm8994_pcm_ops = { .hw_params = smdk_wm8994_pcm_hw_params, }; diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 24cc9d6..25e5be9 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -131,7 +131,7 @@ static int tm2_aif1_hw_params(struct snd_pcm_substream *substream, return tm2_start_sysclk(rtd->card); } -static struct snd_soc_ops tm2_aif1_ops = { +static const struct snd_soc_ops tm2_aif1_ops = { .hw_params = tm2_aif1_hw_params, }; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 3310eda..097663d 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -110,7 +110,7 @@ static int tobermory_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tobermory_ops = { +static const struct snd_soc_ops tobermory_ops = { .hw_params = tobermory_hw_params, }; -- 2.7.4

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[alsa-devel] dmix/dshare: discrepancy between snd_pcm_{status, state, delay}
by Cheng Sun 18 Mar '17

18 Mar '17

Hi all, I've been tracking down the cause of an indefinite hang in PortAudio whenever an xrun occurs, and bisected the issue to the following. After commits faf53c19 (dshare) and 38a2d2ed (dmix), the semantics of snd_pcm_state and snd_pcm_delay are no longer the same as a query to snd_pcm_status, and a subsequent snd_pcm_status_get_*. Note that these two commits remove the lines that set the state field in the snd_pcm_status_t: - status->state = snd_pcm_state(dshare->spcm); This causes the behaviour where, when a dmix's slave suffers an xrun, and the dmix's status is queried, snd_pcm_status_get_state returns RUNNING, and yet snd_pcm_state returns XRUN. The reason why I am not attaching a patch right now is because I am confused as to the commit message in faf53c19, which explicitly states: Note: snd_pcm_dshare_delay() is not updated. Which leads me to think that this change in semantics was perhaps deliberate (if not documented anywhere). However I'm not sure what the rationale is, so I'm not clear on the right way to go about fixing it. If anyone could shed any further light into this, I'd be very grateful. Cheers, Cheng

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