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

[alsa-devel] [PATCH 2/2] ASoC: cs4341: Add DT bindings documentation for CS4341 CODEC
by Alexander Shiyan 16 Sep '16

16 Sep '16

This patch adds DT bindings documentation for Cirrus Logic CS4341 CODEC. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- Documentation/devicetree/bindings/sound/cs4341.txt | 15 +++++++++++++++ 1 file changed, 15 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4341.txt diff --git a/Documentation/devicetree/bindings/sound/cs4341.txt b/Documentation/devicetree/bindings/sound/cs4341.txt new file mode 100644 index 0000000..78d45be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4341.txt @@ -0,0 +1,15 @@ +Cirrus Logic CS4341 DT bindings + +Required properties: + - compatible: "cirrus,cs4341a" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Example: + codec: cs4341@0 { + #sound-dai-cells = <0>; + compatible = "cirrus,cs4341a"; + reg = <0>; + spi-max-frequency = <6000000>; + }; -- 2.7.3

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Re: [alsa-devel] Novation Twitch broken on Linux 4.7?
by Juan Pedro Bolívar Puente 16 Sep '16

16 Sep '16

Hi, Mark Hills <mark(a)xwax.org> writes: > On Fri, 16 Sep 2016, Juan Pedro Bolívar Puente wrote: > >> I am trying to use again my Novation Twitch after some time, and audio >> playback is not working anymore. I have tried with Jackd and various >> other programs, and it seems to always fail to install the hardware >> params. > > [...] > > If it doesn't work then maybe the attached alsacap.c is useful and you > could post the output back (and CC alsa-devel); it should show us the > advertised rates and formats. Here is the ouput: $ ./alsacap *** Scanning for playback devices *** ... Card 1, ID `Twitch', name `Twitch' Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available) 4 channels, sampling rate 44100..48000 Hz Sample formats: S24_3LE Subdevice 0, name `subdevice #0' $ ./alsacap -d hw:Twitch *** Exploring configuration space of device `hw:Twitch' for playback *** 4 channels Sampling rate 44100..48000 Hz Sample formats: S24_3LE Significant bits: 24 For people reading this message for the first time in `alsa-devel`, here are the error output of various commands when trying to use the device: $ aplay -D hw:Twitch -f S24_3LE -c 4 -r 44100 test.wav Playing raw data 'prueba.wav' : Signed 24 bit Little Endian in 3bytes, Rate 44100 Hz, Channels 4 aplay: set_params:1361: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_3LE SUBFORMAT: STD SAMPLE_BITS: 24 FRAME_BITS: 96 CHANNELS: 4 RATE: 44100 PERIOD_TIME: (125011 125012) PERIOD_SIZE: 5513 PERIOD_BYTES: 66156 PERIODS: (3 4) BUFFER_TIME: 500000 BUFFER_SIZE: 22050 BUFFER_BYTES: 264600 TICK_TIME: 0 $ mplayer -ao alsa:device=hw=Twitch test.mp3 MPlayer 1.3.0 (Debian), built with gcc-5.4.0 (C) 2000-2016 MPlayer Team Playing test.mp3. ... [AO_ALSA] Format s16le is not supported by hardware, trying default. [AO_ALSA] Unable to set format: Invalid argument Failed to initialize audio driver 'alsa:device=hw=Twitch' $ mplayer -ao alsa:device=plughw=Twitch test.mp3 MPlayer 1.3.0 (Debian), built with gcc-5.4.0 (C) 2000-2016 MPlayer Team Playing test.mp3. ... [AO_ALSA] alsa-lib: pcm_hw.c:603:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE failed (-32): Broken pipe [AO_ALSA] alsa-lib: pcm_direct.c:1054:(snd1_pcm_direct_initialize_slave) unable to install hw params [AO_ALSA] alsa-lib: pcm_dmix.c:1053:(snd_pcm_dmix_open) unable to initialize slave [AO_ALSA] Playback open error: Broken pipe Failed to initialize audio driver 'alsa:device=hw=Twitch' $ jackd -n twitch -d alsa -d hw:Twitch jackdmp 1.9.11 ... creating alsa driver ... hw:Twitch|hw:Twitch|1024|2|48000|0|0|nomon|swmeter|-|32bit Using ALSA driver USB-Audio running on card 1 - Novation Twitch at usb-0000:00:1d.0-1.2, full speed ALSA: Cannot open PCM device alsa_pcm for capture. Falling back to playback-only mode configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods ALSA: final selected sample format for playback: 24bit little-endian in 3bytes format ALSA: use 2 periods for playback ALSA: cannot set hardware parameters for playback ALSA: cannot configure playback channel Thanks a lot! -- ∿∿∿∿ https://sinusoid.es ∿∿∿∿

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[alsa-devel] [PATCH] ASoC: rockchip: Support headset button function on rk3399
by Cheng-Yi Chiang 16 Sep '16

16 Sep '16

Set the mapping between button and media key event. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 164b6da..ee06489 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -23,6 +23,7 @@ #include <linux/of_gpio.h> #include <linux/delay.h> #include <linux/spi/spi.h> +#include <linux/input.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -209,6 +210,14 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + snd_jack_set_key(rockchip_sound_jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key( + rockchip_sound_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key( + rockchip_sound_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key( + rockchip_sound_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(codec, &rockchip_sound_jack); return 0; -- 2.6.6

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[alsa-devel] [PATCH] ASoC: rt5663: refine error handling
by Arnd Bergmann 16 Sep '16

16 Sep '16

The newly added rt5663 codec driver introduces a couple of warnings when built with -Wmaybe-uninitialized: codecs/rt5663.c: In function 'rt5663_set_tdm_slot': codecs/rt5663.c:2680:2: error: 'reg' may be used uninitialized in this function [-Werror=maybe-uninitialized] codecs/rt5663.c: In function 'rt5663_set_dai_pll': codecs/rt5663.c:2556:12: error: 'shift' may be used uninitialized in this function [-Werror=maybe-uninitialized] codecs/rt5663.c:2596:2: error: 'mask' may be used uninitialized in this function [-Werror=maybe-uninitialized] We can avoid those warnings by always returning an error from the switch() statements instead of falling through with an uninitialized variable when we hit an unexpected case. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: df7c52168ee1 ("ASoC: add rt5663 codec driver") --- sound/soc/codecs/rt5663.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 7963a2736a91..8078abcaf728 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1831,6 +1831,7 @@ static int rt5663_i2s_use_asrc(struct snd_soc_dapm_widget *source, break; default: dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + return 1; } if (da_asrc_en || ad_asrc_en) @@ -2579,6 +2580,7 @@ static int rt5663_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + return -EINVAL; } switch (source) { @@ -2674,6 +2676,7 @@ static int rt5663_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; default: dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + return -EINVAL; } snd_soc_update_bits(codec, reg, RT5668_TDM_MODE_MASK | -- 2.9.0

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[alsa-devel] [PATCH] ALSA: compress: fix some missing and misplaced \n in messages
by Colin King 16 Sep '16

16 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Fix a missing \n in a pr_debug message and move the \n to the end of a pr_err message. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/compress_offload.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 583d407..fec1dfd 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -780,7 +780,7 @@ static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) ret = wait_event_interruptible(stream->runtime->sleep, (stream->runtime->state != SNDRV_PCM_STATE_DRAINING)); if (ret == -ERESTARTSYS) - pr_debug("wait aborted by a signal"); + pr_debug("wait aborted by a signal\n"); else if (ret) pr_debug("wait for drain failed with %d\n", ret); @@ -962,7 +962,7 @@ static int snd_compress_dev_register(struct snd_device *device) compr->card, compr->device, &snd_compr_file_ops, compr, &compr->dev); if (ret < 0) { - pr_err("snd_register_device failed\n %d", ret); + pr_err("snd_register_device failed %d\n", ret); return ret; } return ret; -- 2.9.3

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[alsa-devel] [PATCH] ALSA: hdac: add missing \n to end of dev_err messages
by Colin King 16 Sep '16

16 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trival fix, some dev_err messages are missing a \n, so add it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/hda/ext/hdac_ext_stream.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 2441273..3be051a 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -424,7 +424,7 @@ void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { - dev_err(bus->dev, "Address of SPB capability is NULL"); + dev_err(bus->dev, "Address of SPB capability is NULL\n"); return; } @@ -453,7 +453,7 @@ int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { - dev_err(bus->dev, "Address of SPB capability is NULL"); + dev_err(bus->dev, "Address of SPB capability is NULL\n"); return -EINVAL; } @@ -476,7 +476,7 @@ int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { - dev_err(bus->dev, "Address of SPB capability is NULL"); + dev_err(bus->dev, "Address of SPB capability is NULL\n"); return -EINVAL; } @@ -516,7 +516,7 @@ void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { - dev_err(bus->dev, "Address of DRSM capability is NULL"); + dev_err(bus->dev, "Address of DRSM capability is NULL\n"); return; } @@ -545,7 +545,7 @@ int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { - dev_err(bus->dev, "Address of DRSM capability is NULL"); + dev_err(bus->dev, "Address of DRSM capability is NULL\n"); return -EINVAL; } -- 2.9.3

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[alsa-devel] [PATCH] Revert "ASoC: core: mark SND_SOC_BYTES_EXT as deprecated"
by Takashi Sakamoto 16 Sep '16

16 Sep '16

SND_SOC_BYTES_EXT was added by commit d98812082c87 ("ASoC: add SND_SOC_BYTES_EXT") to allow drivers to transfer arbitrary byte-aligned data to/from devices via ALSA control interface. Later, SND_SOC_BYTES_TLV was added by commit 7523a271682f ("ASoC: core: add a helper for extended byte controls using TLV") was added for the same purpose and developers expected this to obsolete the former. However, SND_SOC_BYTES_TLV brought confusions to user land because it doesn't follow to a protocol of ALSA control interface. At least, this macro and related kernel APIs include two misunderstandings: - 'struct snd_ctl_elem_info.count' can also represent the length of TLV packet paylaod, snd_soc_bytes_info_ext() performs in this way. - 'struct snd_ctl_tlv.tlv' can include arbitrary data regardless of TLV packet structure, snd_soc_bytes_tlv_callback() performs in this way. In a policy of kernel land development, it's quite worse to break protocols for applications. Therefore, developers are discouraged to use these kernel APIs. In the first place, SND_SOC_BYTES_TLV was added to satisfy a request of developers who need to add control elements which transfer data larger than the size which 'struct snd_ctl_elem_value' can represent; e.g. over 512 bytes. As long as the size is less than the size, SND_SOC_BYTES_EXT is still available. Although there is the limitation of maximum data size, it's better to use this API for stable application interface till better alternative ways are implemented in future. For these reasons, this commit reverts the previous commit which lead developers to the worse behaviour. Unfortunately, SND_SOC_BYTES_TLV was already merged in development cycle for kernel 3.17 and published, so this commit doesn't touch it for any compatibilities. Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069483.h… Reference: commit d98812082c87 ("ASoC: add SND_SOC_BYTES_EXT") Reference: commit 7523a271682f ("ASoC: core: add a helper for extended byte controls using TLV") Cc: Vinod Koul <vinod.koul(a)intel.com> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/soc.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6144882..18f7d21 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -311,9 +311,6 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } -/* - * SND_SOC_BYTES_EXT is deprecated, please USE SND_SOC_BYTES_TLV instead - */ #define SND_SOC_BYTES_EXT(xname, xcount, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info_ext, \ -- 2.7.4

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[alsa-devel] [PATCH] ALSA: au88x0: Add missing \n to end of dev_err message
by Colin King 16 Sep '16

16 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trival fix, the dev_err message is missing a \n so add it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/au88x0/au88x0_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index d3125c1..e1af24f 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1043,7 +1043,7 @@ static void vortex_fifo_init(vortex_t * vortex) for (x = NR_ADB - 1; x >= 0; x--) { hwwrite(vortex->mmio, addr, (FIFO_U0 | FIFO_U1)); if (hwread(vortex->mmio, addr) != (FIFO_U0 | FIFO_U1)) - dev_err(vortex->card->dev, "bad adb fifo reset!"); + dev_err(vortex->card->dev, "bad adb fifo reset!\n"); vortex_fifo_clearadbdata(vortex, x, FIFO_SIZE); addr -= 4; } -- 2.9.3

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[alsa-devel] [PATCH v2 0/9] ASoC: simple-card: tidyup and synchronize patches
by Kuninori Morimoto 16 Sep '16

16 Sep '16

Hi Mark These are v2 of cleanup/tidyup and synchronize patches for simple-card. We have now simple-card and simple-scu-card. These are almost same, difference should be very small. But, because of historical reason, these have "code" difference. This patch cleanup and synchronize these. Kuninori Morimoto (9): ASoC: simple-card: call of_node_put() for dai_link ASoC: simple-card: tidyup mclk-fs of_property_read_u32() usage ASoC: simple-card: code sync: tidyup simple_priv_to_xxx() macro ASoC: simple-card: code sync: use simple_priv_to_props() macro ASoC: simple-card: is GPL v2 ASoC: simple-card: code sync: tidyup white line ASoC: simple-card: code sync: use tag ASoC: simple-card: code sync: rename num_link to num ASoC: simple-card: use kzalloc() for dai_props / dai_link sound/soc/generic/simple-card.c | 87 ++++++++++++++++++++--------------------- 1 file changed, 43 insertions(+), 44 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] Revert "ASoC: core: mark SND_SOC_BYTES_EXT as deprecated"
by Takashi Sakamoto 16 Sep '16

16 Sep '16

SND_SOC_BYTES_TLV brings confusions to user land because it doesn't follow to a protocol of ctl and tlv operation. At least, this macro and related kernel APIs include two misunderstandings: - 'struct snd_ctl_elem_info.count' can also represent the length of TLV packet paylaod, snd_soc_bytes_info_ext() performs in this way. - 'struct snd_ctl_tlv.tlv' can include arbitrary data regardless of TLV packet structure, snd_soc_bytes_tlv_callback() performs in this way. In a policy of kernel land development, it's quite worse to break protocols for applications. Therefore, developers are discouraged to use these kernel APIs. In the first place, SND_SOC_BYTES_TLV was added to satisfy a request of developers who need to add control elements which transfer data larger than the size which 'struct snd_ctl_elem_value' can represent; e.g. over 512 bytes. However, as long as the size is less than the size; e.g. 512 bytes, SND_SOC_BYTES_EXT is still available. Although there is actually the limitation of maximum data size, it's better to use this API for stable application interface till better alternative ways are implemented in future. For these reasons, this commit reverts the previous commit which lead developers to the worse behaviour. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/soc.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6144882..18f7d21 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -311,9 +311,6 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } -/* - * SND_SOC_BYTES_EXT is deprecated, please USE SND_SOC_BYTES_TLV instead - */ #define SND_SOC_BYTES_EXT(xname, xcount, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info_ext, \ -- 2.7.4

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