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

[alsa-devel] [PATCH 1/2] ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The SND_AT91_SOC_SAM9G20_WM8731 and SND_AT91_SOC_SAM9X5_WM8731 machine driver symbols select SND_SOC_WM8731 which depends on SND_SOC_I2C_AND_SPI. So the machine driver symbols need to depend on SND_SOC_I2C_AND_SPI as well, otherwise we might end up with a invalid configuration, which will sooner or later upset the randconfig auto-builders. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/atmel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1579e99..2f185e5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,7 +25,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -45,7 +45,7 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 -- 1.8.0

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[alsa-devel] [PATCH] ASoC: Intel: remove unneeded variable
by Julia Lawall 12 Apr '15

12 Apr '15

Removes unneeded variable used to store return value. Generated by: scripts/coccinelle/misc/returnvar.cocci CC: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- sst-baytrail-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -511,7 +511,6 @@ static int sst_byt_process_notification( struct sst_byt_stream *stream; u64 header; u8 msg_id, stream_id; - int handled = 1; header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); msg_id = sst_byt_header_msg_id(header); @@ -531,7 +530,7 @@ static int sst_byt_process_notification( break; } - return handled; + return 1; } static irqreturn_t sst_byt_irq_thread(int irq, void *context)

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[alsa-devel] [PATCH 0/2 RFC] control core refactoring
by Takashi Sakamoto 12 Apr '15

12 Apr '15

This patchset is for refactoring control core code. The first patch is re-send http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087700.h… The second patch is new. As long as I investigated, the changed code was written in 1999 and its basic design have never been changed. In my opinion, the design is a bit complicated and refactoring it is better to decrease maintaining cost. Takashi Sakamoto (2): ALSA: ctl: evaluate macro instead of numerical value ALSA: ctl: refactoring for read operation sound/core/control.c | 87 ++++++++++++++++++++++++++++++++-------------------- 1 file changed, 54 insertions(+), 33 deletions(-) -- 2.1.0

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Re: [alsa-devel] Playback - Overwrite buffer with silence
by Nilhcraiv 12 Apr '15

12 Apr '15

Hello Clemens, Fisrt of all thanks for your answer. In my code there are error checking, of course. The code of this email thread is only just to simplify the question. Clemens wrote: It is impossible for the boundar to be zero. I am sure, if you compile this code, "boundary" variable is always zero. But the question is, How I can put silence data in buffer if the received data is not available at time? (to avoid under-run) Is the doc wrong? #define FRAMES 240 #define PERIOD 1*FRAMES #define BUFFER_SIZE PERIOD*2 #define CHANNELS 1 #define SAMPLE_RATE 48000 snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int exact_rate,dir; snd_pcm_uframes_t boundary=0; snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; snd_pcm_hw_params_alloca(&hwparams); snd_pcm_sw_params_alloca(&swparams); /*PlayBack hw */ snd_pcm_open(&pcm_handle, "default", stream, 0); snd_pcm_hw_params_any(pcm_handle, hwparams); snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE); exact_rate = SAMPLE_RATE; snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0); snd_pcm_hw_params_set_channels(pcm_handle, hwparams, CHANNELS); snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, BUFFER_SIZE); snd_pcm_hw_params(pcm_handle, hwparams); /*Avoid under-run sw*/ snd_pcm_sw_params_current(pcm_handle, swparams); snd_pcm_sw_params_get_boundary(swparams, &boundary); /*Here, boundary is allways ZERO*/ snd_pcm_sw_params_set_silence_threshold(pcm_handle, swparams, 0); /*According to the doc.*/ snd_pcm_sw_params_set_silence_size(pcm_handle, swparams, boundary); /*According to the doc.*/ snd_pcm_sw_params(pcm_handle,swparams); /*END Avoid under-run*/ snd_pcm_prepare(pcm_handle); Thanks! Nilhcraiv wrote: > If I obtain the boundary for ring pointers in frames at the start > (snd_pcm_sw_params_get_boundary), the boundary is always zero. It is impossible for the boundar to be zero. > Here you can see a light version of my code (without error checking): Error checking would be important. Regards, Clemens

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[alsa-devel] [PATCH v5 0/5] ALSA: jack: Refactoring for jack kctls
by Jie Yang 12 Apr '15

12 Apr '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For hda, a. call snd_jack_kctl_new()(in __snd_hda_jack_add_kctl()) to new a jack_kctl(also new kctl); and use snd_hda_ctl_add to add kctl to card; b. new a hda_jack_tbl, appoint jack_tbl->jack_kctl = jack_kctl; c. for non-phantom jack, call snd_jack_new() with this jack_kctl, then the jack_kctl will be added to the new created jack; Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (5): ALSA: jack: create jack kcontrols for every jack input ALSA: jack: add a parameter to pass kctl for snd_jack_new ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove export snd_kctl_jack_new() include/sound/jack.h | 25 ++++++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 1 - sound/core/jack.c | 152 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_jack.c | 52 +++++++------- sound/pci/hda/hda_jack.h | 4 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 10 files changed, 201 insertions(+), 45 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/4] ALSA: ctl: fix userspace control element operations
by Takashi Sakamoto 12 Apr '15

12 Apr '15

This patchset is to fix some issues related to operations for userspace control element. Three patches are related to identical information, the other is related to several element added by one operation. Unfortunately, current alsa-lib has no functionality to achieve the operation to add several element. To demonstrate this and test these patches, I prepare for a small GObject introspection library, alsa-gir. https://github.com/takaswie/alsa-gir This reposiroty includes sample scripts written by python. The file of sample/ctl.py demonstrates userspace control elements. If this library helps your review for this patchset, I'm feeling happy. Takashi Sakamoto (4): ALSA: ctl: confirm to return all identical information in 'activate' event ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls ALSA: ctl: fill identical information to return value when adding userspace elements ALSA: ctl: fix to handle several elements added by one operation for userspace element sound/core/control.c | 44 +++++++++++++++++++++++++++++++++++++------- 1 file changed, 37 insertions(+), 7 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH] ASoC: rnsd: fix build regression without CONFIG_OF
by Arnd Bergmann 12 Apr '15

12 Apr '15

The r-car sound driver only works when CONFIG_OF is set, and after a recent change has a compile-time dependency as well: sound/built-in.o: In function `rsnd_dma_request_channel': :(.text+0x9fb84): undefined reference to `of_dma_request_slave_channel' This could be fixed either by adding a static inline wrapper for the function, or by adding a Kconfig dependency. This implements the second approach, which seems appropriate because the driver in fact has a hard dependency. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 72adc61f4637aa3 ("ASoC: rsnd: 1st DMAC dma-names cares subnode") diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 2b3030415573..07114b0b0dc1 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" + depends on DMA_OF select SND_SIMPLE_CARD select REGMAP_MMIO help

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[alsa-devel] [GIT PULL] Last minute ASoC fix for v4.0
by Mark Brown 12 Apr '15

12 Apr '15

The following changes since commit c517d838eb7d07bbe9507871fab3931deccff539: Linux 4.0-rc1 (2015-02-22 18:21:14 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.0-rc7 for you to fetch changes up to 14f0413ce3c400a74213a861bc8f47dca5e4d231: ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4 (2015-04-11 22:09:14 +0100) ---------------------------------------------------------------- ASoC: Last minute fix for v4.0 This patch backs out a change that came in during the merge window which selects a configuration for GPIO4 on pcm512x CODECs that may not be suitable for all systems using the device. Changes for v4.1 will make this properly configurable but for now it's safest to revert to the v3.19 behaviour and leave the pin configuration alone. Sorry for sending this direct at the last minute but due to the GPIO misuse it'd be really good to get it in the release and I'd not realised it hadn't been sent yet - between some travel, a job change and other non-urgent fixes coming in I'd lost track of the urgency. It's been in -next for several weeks now, is isolated to the driver and fairly clear to inspection. ---------------------------------------------------------------- Howard Mitchell (1): ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4 sound/soc/codecs/pcm512x.c | 19 ------------------- 1 file changed, 19 deletions(-)

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[alsa-devel] Playback - Overwrite buffer with silence
by Nilhcraiv 11 Apr '15

11 Apr '15

Hello, I am working an application and I have problems playing the audio with ALSA. In my application, I receive the sound data, this data must be decoded to be reproduce. In this process, always obtain an under-runerror. To fix this problem, I followed the documentation i.e. I use snd_pcm_sw_params_set_silence_size() and snd_pcm_sw_params_set_silence_threshold() to overwrite the buffer with silence. --> http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___s_w___params.… but this does not work. Does the doc is wrong? If I obtain the boundary for ring pointers in frames at the start (snd_pcm_sw_params_get_boundary), the boundary is always zero. If I obtain the boundary each time (in the received_data() function) it is more than the buffer size and I get error when I put it in snd_pcm_sw_params_set_silence_size(), ¿Why?, Is it correct to use the boundary value?, ¿How I can avoid under-run by putting silence if I have not data at time? Here you can see a light version of my code (without error checking): #define FRAMES 240 #define PERIOD 1*FRAMES #define BUFFER_SIZE PERIOD*2 #define CHANNELS 1 #define SAMPLE_RATE 48000 snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int exact_rate,dir; snd_pcm_uframes_t boundary=0; snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; snd_pcm_hw_params_alloca(&hwparams); snd_pcm_sw_params_alloca(&swparams); /*PlayBack hw */ snd_pcm_open(&pcm_handle, "default", stream, 0); snd_pcm_hw_params_any(pcm_handle, hwparams); snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE); exact_rate = SAMPLE_RATE; snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0); snd_pcm_hw_params_set_channels(pcm_handle, hwparams, CHANNELS); snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, BUFFER_SIZE); snd_pcm_hw_params(pcm_handle, hwparams); /*Avoid under-run sw*/ snd_pcm_sw_params_current(pcm_handle, swparams); snd_pcm_sw_params_get_boundary(swparams, &boundary); /*Here, boundary is allways ZERO*/ snd_pcm_sw_params_set_silence_threshold(pcm_handle, swparams, 0); /*According to the doc.*/ snd_pcm_sw_params_set_silence_size(pcm_handle, swparams, boundary); /*According to the doc.*/ snd_pcm_sw_params(pcm_handle,swparams); /*END Avoid under-run*/ snd_pcm_prepare(pcm_handle); void received_data(void *t_data)/*short version of the function*/ { short *decoded_data = NULL; int len=0; decoded_data = (short *)malloc(PERIOD * FRAME_SIZE); len = decoder(&decoded_data,t_data); snd_pcm_sw_params_get_boundary(swparams, &boundary); /*Here, boundary is always a large number*/ snd_pcm_sw_params_set_silence_size(pcm_handle, swparams, boundary);/*Here, boundary in frames is always a large number, and there are an error...*/ snd_pcm_sw_params(pcm_handle,swparams); pcmreturn = snd_pcm_writei(pcm_handle, decoded_data, len);/*In most cases there are UNDERRUN...*/ xrun_recovery(&pcm_handle, pcmreturn);/*Recover in case of an error.*/ } Thanks in advance! Vicente

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[alsa-devel] [PATCH v2] ALSA: emu10k1: add toggles for E-mu 1010 optical ports
by Michael Gernoth 11 Apr '15

11 Apr '15

The optical ports on the E-mu 1010 (and dock) can be configured for ADAT- or S/PDIF-mode, which is currently hardcoded to ADAT. Add two mixer elements to expose this setting. Tested on an E-mu 1010 PCIe with connected Micro Dock. Signed-off-by: Michael Gernoth <michael(a)gernoth.net> --- sound/pci/emu10k1/emumixer.c | 118 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 118 insertions(+) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 1de3302..55e5716 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -806,6 +806,108 @@ static struct snd_kcontrol_new snd_emu1010_internal_clock = .put = snd_emu1010_internal_clock_put }; +static int snd_emu1010_optical_out_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { + "SPDIF", "ADAT" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_emu1010_optical_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->emu1010.optical_out; + return 0; +} + +static int snd_emu1010_optical_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + u32 tmp; + int change = 0; + + val = ucontrol->value.enumerated.item[0]; + /* Limit: uinfo->value.enumerated.items = 2; */ + if (val >= 2) + return -EINVAL; + change = (emu->emu1010.optical_out != val); + if (change) { + emu->emu1010.optical_out = val; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + } + return change; +} + +static struct snd_kcontrol_new snd_emu1010_optical_out = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Optical Output Mode", + .count = 1, + .info = snd_emu1010_optical_out_info, + .get = snd_emu1010_optical_out_get, + .put = snd_emu1010_optical_out_put +}; + +static int snd_emu1010_optical_in_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { + "SPDIF", "ADAT" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_emu1010_optical_in_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->emu1010.optical_in; + return 0; +} + +static int snd_emu1010_optical_in_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + u32 tmp; + int change = 0; + + val = ucontrol->value.enumerated.item[0]; + /* Limit: uinfo->value.enumerated.items = 2; */ + if (val >= 2) + return -EINVAL; + change = (emu->emu1010.optical_in != val); + if (change) { + emu->emu1010.optical_in = val; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + } + return change; +} + +static struct snd_kcontrol_new snd_emu1010_optical_in = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Optical Input Mode", + .count = 1, + .info = snd_emu1010_optical_in_info, + .get = snd_emu1010_optical_in_get, + .put = snd_emu1010_optical_in_put +}; + static int snd_audigy_i2c_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2051,6 +2153,14 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_out, emu)); + if (err < 0) + return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_in, emu)); + if (err < 0) + return err; } else if (emu->card_capabilities->emu_model) { /* all other e-mu cards for now */ @@ -2086,6 +2196,14 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_out, emu)); + if (err < 0) + return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_in, emu)); + if (err < 0) + return err; } if ( emu->card_capabilities->i2c_adc) { -- 2.1.4

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