December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH - plugins] oss: avoid conflict between reclev and igain mixer ctls
by Luca Bayer 04 Dec '13

04 Dec '13

To fix the following error that happens with Intel HDA driver. $ amixer ALSA lib simple_none.c:1549:(simple_add1) helem (MIXER,'Capture Volume',0,0,0) appears twice or more amixer: Mixer default load error: Invalid argument Signed-off-by: Luca Bayer <kometto(a)web.de> diff --git a/oss/ctl_oss.c b/oss/ctl_oss.c index b8847b1..b05ba18 100644 --- a/oss/ctl_oss.c +++ b/oss/ctl_oss.c @@ -52,7 +52,7 @@ static const char *const vol_devices[SOUND_MIXER_NRDEVICES] = { [SOUND_MIXER_CD] = "CD Playback Volume", [SOUND_MIXER_IMIX] = "Monitor Mix Playback Volume", [SOUND_MIXER_ALTPCM] = "Headphone Playback Volume", - [SOUND_MIXER_RECLEV] = "Capture Volume", + [SOUND_MIXER_RECLEV] = "Master Capture Volume", [SOUND_MIXER_IGAIN] = "Capture Volume", [SOUND_MIXER_OGAIN] = "Playback Volume", [SOUND_MIXER_LINE1] = "Aux Playback Volume",

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[alsa-devel] [PATCH] ASoC: dmaengine: remove SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME
by Stephen Warren 04 Dec '13

04 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> struct snd_dmaengine_pcm_config's new chan_names field implements the same feature, and nothing uses SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME, so remove it. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This follows on from the patch "ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config" which I posted earlier today. include/sound/dmaengine_pcm.h | 6 ------ sound/soc/soc-generic-dmaengine-pcm.c | 13 +------------ 2 files changed, 1 insertion(+), 18 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 87e9d481d7b6..634b0fca3afa 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,7 +61,6 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. - * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { @@ -70,7 +69,6 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; - const char *chan_name; unsigned int fifo_size; }; @@ -100,10 +98,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ac16259490ee..819d5997eece 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -188,8 +188,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; - struct device *dev = rtd->platform->dev; - struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -210,13 +208,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!pcm->chan[i] && - (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) - pcm->chan[i] = dma_request_slave_channel(dev, - dma_data->chan_name); - if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); @@ -290,9 +281,7 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, unsigned int i; const char *name; - if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || - !dev->of_node) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) return; if (config->dma_dev) { -- 1.8.1.5

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[alsa-devel] [PATCH 1/2] ASoC: jz4740-i2s: Use managed resources
by Lars-Peter Clausen 03 Dec '13

03 Dec '13

Makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 85 ++++++++--------------------------------- 1 file changed, 15 insertions(+), 70 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 4c849a4..a0b6a85 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -432,91 +432,36 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; + struct resource *mem; int ret; - i2s = kzalloc(sizeof(*i2s), GFP_KERNEL); - + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; - i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!i2s->mem) { - ret = -ENOENT; - goto err_free; - } - - i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem), - pdev->name); - if (!i2s->mem) { - ret = -EBUSY; - goto err_free; - } - - i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem)); - if (!i2s->base) { - ret = -EBUSY; - goto err_release_mem_region; - } + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); - i2s->phys_base = i2s->mem->start; + i2s->phys_base = mem->start; - i2s->clk_aic = clk_get(&pdev->dev, "aic"); - if (IS_ERR(i2s->clk_aic)) { - ret = PTR_ERR(i2s->clk_aic); - goto err_iounmap; - } + i2s->clk_aic = devm_clk_get(&pdev->dev, "aic"); + if (IS_ERR(i2s->clk_aic)) + return PTR_ERR(i2s->clk_aic); - i2s->clk_i2s = clk_get(&pdev->dev, "i2s"); - if (IS_ERR(i2s->clk_i2s)) { - ret = PTR_ERR(i2s->clk_i2s); - goto err_clk_put_aic; - } + i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s"); + if (IS_ERR(i2s->clk_i2s)) + return PTR_ERR(i2s->clk_i2s); platform_set_drvdata(pdev, i2s); - ret = snd_soc_register_component(&pdev->dev, &jz4740_i2s_component, - &jz4740_i2s_dai, 1); - - if (ret) { - dev_err(&pdev->dev, "Failed to register DAI\n"); - goto err_clk_put_i2s; - } - - return 0; - -err_clk_put_i2s: - clk_put(i2s->clk_i2s); -err_clk_put_aic: - clk_put(i2s->clk_aic); -err_iounmap: - iounmap(i2s->base); -err_release_mem_region: - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); -err_free: - kfree(i2s); - - return ret; -} - -static int jz4740_i2s_dev_remove(struct platform_device *pdev) -{ - struct jz4740_i2s *i2s = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - - clk_put(i2s->clk_i2s); - clk_put(i2s->clk_aic); - iounmap(i2s->base); - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); - - kfree(i2s); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &jz4740_i2s_component, &jz4740_i2s_dai, 1); } static struct platform_driver jz4740_i2s_driver = { .probe = jz4740_i2s_dev_probe, - .remove = jz4740_i2s_dev_remove, .driver = { .name = "jz4740-i2s", .owner = THIS_MODULE, -- 1.7.10.4

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[alsa-devel] [PATCH - oxygen 1/1] Patch for the Xonar DG Patch by Roman Volkov
by Marius Graefe 03 Dec '13

03 Dec '13

I just tested Roman Volkovs oxygen patch for the xonar dg and it works like a charm, however I needed to change two lines in xonar_dg_mixer.c: 1) #include <linux/export.h> was missing which resulted in a warning when compiling. 2) I had to assign a .shortname in xonar_dg_model, otherwise my kernel (unmodified ubuntu 3.2.0-56-generic-pae) throws an oops, the backtrace led me to the missing string. Actually this is weird because the shortname is assigned in get_oxygen_model in oxygen.c, and alsamixer also shows that name later, but at some point in the kernel startup sequence it's NULL. Volkovs Xonar DG 4-Patch series is required! Signed-off-by: Marius Graefe <alsa(a)mgraefe.de> diff --git a/sound/pci/oxygen/xonar_dg_mixer.c b/sound/pci/oxygen/xonar_dg_mixer.c index 39a0da7..102cfce 100644 --- a/sound/pci/oxygen/xonar_dg_mixer.c +++ b/sound/pci/oxygen/xonar_dg_mixer.c @@ -26,6 +26,7 @@ #include <linux/pci.h> #include <linux/delay.h> +#include <linux/export.h> #include <sound/control.h> #include <sound/core.h> #include <sound/info.h> @@ -456,6 +457,7 @@ static int xonar_control_filter(struct snd_kcontrol_new *template) } struct oxygen_model model_xonar_dg = { + .shortname = "Xonar DG/DGX", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: hda - fixup ALC262 to skip depop delay before D3 on Intel BayleyBay
by mengdong.lin＠intel.com 03 Dec '13

03 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch sets a 0ms depop delay in fixup funtion 'alc_fixup_no_depop_delay'. And Realteck ALC262 applies this on Intel Baytrail BayleyBay platform to reduce codec suspend time. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Reviewed-by: Kailang Yang <kailang(a)realtek.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 21dcb7b..69f11e2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -374,8 +374,10 @@ static void alc_fixup_no_depop_delay(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PROBE) + if (action == HDA_FIXUP_ACT_PROBE) { spec->no_depop_delay = 1; + codec->depop_delay = 0; + } } static int alc_auto_parse_customize_define(struct hda_codec *codec) -- 1.8.1.2

2 1

[alsa-devel] [TINYCOMPRESS PATCH 1/2] crec: Initial version of a compressed capture utility
by Charles Keepax 03 Dec '13

03 Dec '13

This version only supports capture of PCM streams over a compressed device and saves these as WAV files. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Android.mk | 10 ++ crec.c | 425 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 448 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index 104dfef..3cdd3d0 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..d2fd873 --- /dev/null +++ b/crec.c @@ -0,0 +1,425 @@ +/* + * BSD LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "sound/compress_offload.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SNDRV_PCM_FORMAT_S16_LE; + +struct riff_chunk { + char desc[4]; + uint32_t size; +} __attribute__((__packed__)); + +struct wave_header { + struct { + struct riff_chunk chunk; + char format[4]; + } __attribute__((__packed__)) riff; + + struct { + struct riff_chunk chunk; + uint16_t type; + uint16_t channels; + uint32_t rate; + uint32_t byterate; + uint16_t blockalign; + uint16_t samplebits; + } __attribute__((__packed__)) fmt; + + struct { + struct riff_chunk chunk; + } __attribute__((__packed__)) data; +} __attribute__((__packed__)); + +const struct wave_header blank_wave_header = { + .riff = { + .chunk = { + .desc = "RIFF", + }, + .format = "WAVE", + }, + .fmt = { + .chunk = { + .desc = "fmt ", /* Note the space is important here */ + .size = sizeof(blank_wave_header.fmt) - + sizeof(blank_wave_header.fmt.chunk), + }, + .type = 0x01, /* PCM */ + }, + .data = { + .chunk = { + .desc = "data", + }, + }, +}; + +static void init_wave_header(struct wave_header *header, uint16_t channels, + uint32_t rate, uint16_t samplebits) +{ + memcpy(header, &blank_wave_header, sizeof(blank_wave_header)); + + header->fmt.channels = channels; + header->fmt.rate = rate; + header->fmt.byterate = channels * rate * (samplebits / 8); + header->fmt.blockalign = channels * (samplebits / 8); + header->fmt.samplebits = samplebits; +} + +static void size_wave_header(struct wave_header *header, uint32_t size) +{ + header->riff.chunk.size = sizeof(*header) - + sizeof(header->riff.chunk) + size; + header->data.chunk.size = size; +} + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n" + "-v\tverbose mode\n" + "-l\tlength of record in seconds\n" + "-h\tPrints this help list\n\n" + "-C\tSpecify the number of channels (default %u)\n" + "-R\tSpecify the sample rate (default %u)\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.wav\n" + "\tcrec -f 5 test.wav\n", + DEFAULT_CHANNELS, DEFAULT_RATE); + + exit(EXIT_FAILURE); +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else { + printf("DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, + unsigned int length, unsigned int rate, + unsigned int channels, unsigned int format) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + struct wave_header header; + FILE *file; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SNDRV_PCM_FORMAT_S32_LE: + samplebits = 32; + break; + default: + samplebits = 16; + break; + } + + /* Convert length from seconds to bytes */ + length = length * rate * (samplebits / 8) * channels; + + if (verbose) + printf("%s: entry, reading %u bytes\n", __func__, length); + + file = fopen(name, "w+b"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + /* Write a header, will update with size once record is complete */ + init_wave_header(&header, channels, rate, samplebits); + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error writing output file header: %d\n", + ferror(file)); + goto file_exit; + } + + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = compress_get_alsa_rate(rate); + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.bit_rate = 0; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto file_exit; + }; + + if (verbose) + printf("%s: Opened compress device\n", __func__); + + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto comp_exit; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, rate); + + compress_start(compress); + + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = compress_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto buf_exit; + } + if ((unsigned int)read != size) { + fprintf(stderr, "We read %d, DSP sent %d\n", + size, read); + } + + if (read > 0) { + total_read += read; + + written = fwrite(buffer, 1, read, file); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %d\n", + ferror(file)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = compress_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + } + + /* Update file header now we know file size */ + size_wave_header(&header, total_read); + ret = fseek(file, 0, SEEK_SET); + if (ret < 0) { + fprintf(stderr, "Error seeking: %s\n", stderror(errno)); + goto buf_exit; + } + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error updating output file header: %d\n", + ferror(file)); + goto buf_exit; + } + + if (verbose) + printf("%s: exit success\n", __func__); + + free(buffer); + fclose(file); + + compress_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + compress_close(compress); +file_exit: + fclose(file); + + if (verbose) + printf("%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, length = 0; + unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; + unsigned int format = DEFAULT_FORMAT; + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + case 'l': + length = strtol(optarg, NULL, 10); + break; + case 'R': + rate = strtol(optarg, NULL, 10); + break; + case 'C': + channels = strtol(optarg, NULL, 10); + break; + case 'F': + if (strcmp(optarg, "S16_LE") == 0) { + format = SNDRV_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SNDRV_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + printf("Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 677878d..81782d0 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c utils.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c utils.c +RECSRC = crec.c utils.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm8731: fix dsp mode configuration
by Bo Shen 03 Dec '13

03 Dec '13

According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0. So, fix LRP for DSP mode as the datesheet specification. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8731.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 456bb8c..bc7472c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: soc-pcm: Only do snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 03 Dec '13

03 Dec '13

This patch removed the redundant snd_soc_dai_digital_mute() in close() since it is better to mute in hw_free() which is slightly earlier and symmetrical for the case of reconfiguration, 'aplay 44100.wav 48000.wav' for example: open()->hw_params()->prepare(unmute)->playing->hw_free(mute)->hw_params()-> parepare(unmute)->playing->hw_free(mute)->close() This patch also corrects the condition of the snd_soc_dai_digital_mute() in hw_free() because it would not be executed as we only decrease codec->active in snd_soc_close(). Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/soc-pcm.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb340a8..eac82858 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -474,11 +474,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); - if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -689,7 +684,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -707,7 +702,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - if (!codec->active) + if ((playback && codec_dai->playback_active == 1) || + (!playback && codec_dai->capture_active == 1)) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ -- 1.8.4

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[alsa-devel] [PATCH 1/2] ASoC: ux500_pcm: Differentiate between pdata and DT initialisation
by Lee Jones 03 Dec '13

03 Dec '13

If booting with full DT support (i.e. DMA too, the last piece of the puzzle), then we don't need to use the compatible_request_channel call back or require some of the historical bumph which probably isn't required by a platform data start-up now either. This will also be ripped out in upcoming commits. Cc: alsa-devel(a)alsa-project.org Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_pcm.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 2f1bdb7..ad0d420 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -134,15 +134,24 @@ static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { .prepare_slave_config = ux500_pcm_prepare_slave_config, }; +static const struct snd_dmaengine_pcm_config ux500_dmaengine_of_pcm_config = { + .prepare_slave_config = ux500_pcm_prepare_slave_config, +}; + int ux500_pcm_register_platform(struct platform_device *pdev) { + const struct snd_dmaengine_pcm_config *pcm_config; + struct device_node *np = pdev->dev.of_node; int ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, - &ux500_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_COMPAT | - SND_DMAENGINE_PCM_FLAG_NO_DT); + if (np) + pcm_config = &ux500_dmaengine_of_pcm_config; + else + pcm_config = &ux500_dmaengine_pcm_config; + + ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT); if (ret < 0) { dev_err(&pdev->dev, "%s: ERROR: Failed to register platform '%s' (%d)!\n", -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: ssm2602: Use core for applying symmetry constraints
by Lars-Peter Clausen 03 Dec '13

03 Dec '13

Let the core take care of applying sample rate and sample bits constraints instead of open-coding this in the driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm2602.c | 50 ++-------------------------------------------- 1 file changed, 2 insertions(+), 48 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 480074d..c6dd485 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,8 +53,6 @@ enum ssm2602_type { struct ssm2602_priv { unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; struct regmap *regmap; @@ -277,11 +275,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); unsigned int iface; - if (substream == ssm2602->slave_substream) { - dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - if (srate < 0) return srate; @@ -314,33 +307,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct snd_pcm_runtime *master_runtime; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - * TODO: the ssm2602 allows pairs of non-matching PB/REC rates - */ - if (ssm2602->master_substream) { - master_runtime = ssm2602->master_substream->runtime; - dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - if (master_runtime->rate != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - if (master_runtime->sample_bits != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - ssm2602->slave_substream = substream; - } else - ssm2602->master_substream = substream; if (ssm2602->sysclk_constraints) { snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -351,19 +317,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - - if (ssm2602->master_substream == substream) - ssm2602->master_substream = ssm2602->slave_substream; - - ssm2602->slave_substream = NULL; -} - - static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec); @@ -530,7 +483,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops ssm2602_dai_ops = { .startup = ssm2602_startup, .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, @@ -551,6 +503,8 @@ static struct snd_soc_dai_driver ssm2602_dai = { .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, .ops = &ssm2602_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, }; static int ssm2602_suspend(struct snd_soc_codec *codec) -- 1.8.0

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