[ALSA-UTILS] Add support for cplay and crecord

Message ID	1425483360-1806-1-git-send-email-qais.yousef@imgtec.com (mailing list archive)
State	New, archived
Headers	show Return-Path: <alsa-devel-bounces@alsa-project.org> From: Qais Yousef <qais.yousef@imgtec.com> To: <alsa-devel@alsa-project.org> Date: Wed, 4 Mar 2015 15:36:00 +0000 Message-ID: <1425483360-1806-1-git-send-email-qais.yousef@imgtec.com> MIME-Version: 1.0 Cc: Takashi Iwai <tiwai@suse.de>, Vinod Koul <vinod.koul@intel.com>, Mark Brown <broonie@kernel.org>, Qais Yousef <qais.yousef@imgtec.com> Subject: [alsa-devel] [ALSA-UTILS][PATCH] Add support for cplay and crecord Precedence: list Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Errors-To: alsa-devel-bounces@alsa-project.org Sender: alsa-devel-bounces@alsa-project.org

Dne 6.3.2015 v 11:35 Srinivasan S napsal(a): > Once again many Thanks for the quick responses, > > Could you please try & let me know whether does it works viceversa ie., arecord on loopback card first & then aplay on the loopback card as shown below > > $ arecord -D plughw:3,1,0 -f dat -c 1 a.wav > Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono > > $ aplay -D plughw:3,0,0 a.wav > Playing WAVE '/usr/share/sounds/alsa/Noise.wav' : Signed 16 bit Little > Endian, Rate 48000 Hz, Mono > > Awaiting for your replies, I'm not sure, if you understand the purpose of the loopback card. It just passes the data from one playback device to another capture device so it returns them to the userspace. So you need to have an input (source) to be loopbacked. You're probably trying to loopback the zero (silence) samples which are provided when the source is not available from the loopback card. You cannot do (simple notation): arecord Loopback,1 | aplay Loopback,0 it's endless silence loop, but you can do (both commands should be executed at same time): aplay Loopback,0 <some_wav_file_with_real_content> arecord Loopback,1 <loopbacked_result_stored_to_wav> Jaroslav > > Thanks in advance again. > Srinivasan S > > > ________________________________________ > From: Jaroslav Kysela <perex@perex.cz> > Sent: Friday, March 6, 2015 12:21 AM > To: Srinivasan S > Cc: alsa-devel@alsa-project.org > Subject: Re: snd-aloop not working in linux-3.12.10 > > Dne 5.3.2015 v 17:42 Srinivasan S napsal(a): >> Thanks for your replies, >> >> I even tried in the below linux host machine first ie., ubuntu 12.04 prior trying in embedded board >> Linux srinivasan-Latitude-3440 3.13.0-43-generic #72~precise1-Ubuntu SMP Tue Dec 9 12:14:18 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux >> >> am getting the below >> >>> card 1, device 0 >>> card 1, device 1 >> >> aplay -D hw:1,0,0 TangoForTajMusic11.wav >> >> arecord -D hw:1,1,0 record .wav >> >> Still this is not working in the linux host machine >> >> Could you please help me out in resolving this issue in host machine > > On my host, kernel 3.17.8-200.fc20.x86_64, loopback card is #3: > > $ aplay -D plughw:3,0,0 /usr/share/sounds/alsa/Noise.wav > Playing WAVE '/usr/share/sounds/alsa/Noise.wav' : Signed 16 bit Little > Endian, Rate 48000 Hz, Mono > $ arecord -D plughw:3,1,0 -f dat -c 1 a.wav > Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono > > As you see, the "capture" program must set the same parameters as the > sample provider. > > This also works: > > $ alsaloop -C hw:3,1 -P plughw:1 -t 50000 > > The "plughw:1" is the real sound hardware in this case (first device, > second soundcard #1). The aplay command is same as in the first example > with device "plughw:3,0,0". In this case, the alsaloop detects the PCM > stream paramter changes automatically using the control API: > > $ amixer -c 3 controls > numid=2,iface=PCM,name='PCM Notify' > numid=1,iface=PCM,name='PCM Rate Shift 100000' > numid=3,iface=PCM,name='PCM Slave Active' > numid=6,iface=PCM,name='PCM Slave Channels' > numid=4,iface=PCM,name='PCM Slave Format' > numid=5,iface=PCM,name='PCM Slave Rate' > numid=8,iface=PCM,name='PCM Notify',subdevice=1 > numid=7,iface=PCM,name='PCM Rate Shift 100000',subdevice=1 > numid=9,iface=PCM,name='PCM Slave Active',subdevice=1 > > > Jaroslav > > >> >> Thanks in advance >> >> >> >> ________________________________________ >> From: Jaroslav Kysela <perex@perex.cz> >> Sent: Thursday, March 5, 2015 7:43 PM >> To: Srinivasan S >> Cc: alsa-devel@alsa-project.org >> Subject: Re: snd-aloop not working in linux-3.12.10 >> >> Dne 5.3.2015 v 14:45 Srinivasan S napsal(a): >>> Dear Jaroslav >>> >>> As this feature ie., snd-aloop designed by you, could you please redirect to the respective links where my queries can be posted & get the solutions for the problem >> >> No idea. It looks like you are asking for a commercial support. >> It may be a cache coherency issue or something else. The embedded/ARM >> platforms might behave completely differently than x86 on which >> this code was developed and tested. >> >> Jaroslav >> >>> >>> As we are trying to establish GSM two way calls via stereo codec >>> >>> We are planning to use loopback module (ie.,snd-aloop and alsaloop in ti sdk 7) to connect the sink/source and source/sink GSM and Codec. >>> >>> The below is the virtual devices created after configuring snd-aloop in the linux kernel 3.12.10 >>> >>> card 0, device 0 >>> card 0, device 1 >>> >>> whatever am playing we are unable to record in the virtual device, but we are able to play & record with actual device >>> >>> aplay -D hw:0,0,0 play.wav >>> arecord -D hw:0,1,0 record.wav or alsaloop -C hw:0,1 -P hw:0,0 -t 50000 # second terminal, latency 50ms >>> >>> As per the logs below, am using the above commands to perform loopback, could you please let me know why am unable to perform the loopback with the below commands or please let me know am I missing any configurations, >>> >>> As this is feature is not working in ti sdk 7 (ie.,snd-aloop and alsaloop), >>> >>> Kindly requesting to try in your am335x-evm where it has tlv codec & verify this feature ie., snd-aloop & let me know as early as possible >>> >>> >>> logs : >>> ====== >>> root@am335x-evm:/# ls /dev/snd/ >>> by-path controlC1 pcmC0D0p pcmC0D1p pcmC1D0p >>> controlC0 pcmC0D0c pcmC0D1c pcmC1D0c timer >>> root@am335x-evm:/# cat /proc/asound/devices >>> 0: [ 0] : control >>> 16: [ 0- 0]: digital audio playback >>> 17: [ 0- 1]: digital audio playback >>> 24: [ 0- 0]: digital audio capture >>> 25: [ 0- 1]: digital audio capture >>> 32: [ 1] : control >>> 33: : timer >>> 48: [ 1- 0]: digital audio playback >>> 56: [ 1- 0]: digital audio capture >>> root@am335x-evm:/# >>> >>> Loopback device >>> ---------------- >>> root@am335x-evm:/# aplay -l >>> **** List of PLAYBACK Hardware Devices **** >>> card 0: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] >>> Subdevices: 8/8 >>> Subdevice #0: subdevice #0 >>> Subdevice #1: subdevice #1 >>> Subdevice #2: subdevice #2 >>> Subdevice #3: subdevice #3 >>> Subdevice #4: subdevice #4 >>> Subdevice #5: subdevice #5 >>> Subdevice #6: subdevice #6 >>> Subdevice #7: subdevice #7 >>> card 0: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] >>> Subdevices: 8/8 >>> Subdevice #0: subdevice #0 >>> Subdevice #1: subdevice #1 >>> Subdevice #2: subdevice #2 >>> Subdevice #3: subdevice #3 >>> Subdevice #4: subdevice #4 >>> Subdevice #5: subdevice #5 >>> Subdevice #6: subdevice #6 >>> Subdevice #7: subdevice #7 >>> card 1: UDA1345TS [TI UDA1345TS], device 0: UDA134x uda134x-hifi-0 [] >>> Subdevices: 1/1 >>> Subdevice #0: subdevice #0 >>> root@am335x-evm:/# aplay -D hw:0,0,0 TangoForTajMusic11.wav >>> Playing WAVE 'TangoForTajMusic11.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo >>> >>> >>> >>> Actual device >>> ------------- >>> root@am335x-evm:/# aplay -D hw:1,0,0 TangoForTajMusic11.wav >>> Playing WAVE 'TangoForTajMusic11.wav' : Signed 16 bit Li[ 2219.654309] DAVINCIIIIIIIIII UDA134XXXXX SYSCLK=12288000 >>> ttle Endian, Rate 48000 Hz, Stereo >>> [ 2219.663776] DAVINCIIIIIIIIII UDA134XXXXX BCLK FREQQ=1536000 >>> [ 2219.672880] DAVINCIIIIIIIIII UDA134XXXXX SYSCLK/BCLK_FREQ =8 >>> [ 2219.678982] uda134x_hw_params CLOCKS uda134x_hw_params uda134x->sysclk: 12288000, params_rate(params):48000 >>> [ 2219.689474] uda134x_hw_params FORMATS uda134x_hw_params dai_fmt: 16385, params_format:2 >>> [ 2219.698095] uda134x_hw_params FORMATS uda134x_hw_params uda134x->sysclk / params_rate(params) 256 >>> [ 2219.707644] UDA1345TSSSSSSSSSSSS SYSCLK / fs ratio is 256 >>> [ 2219.713470] uda134x_hw_params dai_fmt: 16385, params_format:2 >>> [ 2219.719639] UDA1345TSSSSSSSSSSSS FORMAT SND_SOC_DAIFMT_I2S >>> [ 2219.725716] ENTERED davinci_config_channel_size davinci_config_channel_size: tx_rotate = 4 >>> [ 2219.734620] ENTERED davinci_config_channel_size davinci_config_channel_size: MASK= 65535 >>> [ 2219.748324] uda134x_unnnnnnnnmuteeeeeeeeee uda134x_mute mute: 0 >>> [ 2219.756522] davinci_mcasp_starttttttttttttttttttttt SNDRV_PCM_STREAM_PLAYBACK >>> >>> >>> Kindly do the needful as early as possible >>> Awaiting for your replies, >>> >>> Many Thanks in advance, >> >> >> -- >> Jaroslav Kysela <perex@perex.cz> >> Linux Kernel Sound Maintainer >> ALSA Project; Red Hat, Inc. >> > > > -- > Jaroslav Kysela <perex@perex.cz> > Linux Kernel Sound Maintainer > ALSA Project; Red Hat, Inc. >

diff --git a/Makefile.am b/Makefile.am index 5bbe588a8d84..0842657530fd 100644 --- a/Makefile.am +++ b/Makefile.am @@ -19,6 +19,9 @@ if ALSALOOP SUBDIRS += alsaloop endif endif +if HAVE_COMPRESS +SUBDIRS += cplay +endif if HAVE_SEQ SUBDIRS += seq endif diff --git a/configure.ac b/configure.ac index f09aa5484d1d..c08c24b90658 100644 --- a/configure.ac +++ b/configure.ac @@ -42,6 +42,8 @@ fi dnl Check components AC_CHECK_HEADERS([alsa/pcm.h], [have_pcm="yes"], [have_pcm="no"], [#include <alsa/asoundlib.h>]) +AC_CHECK_HEADERS([alsa/compress.h], [have_compress="yes"], [have_compress="no"], + [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/mixer.h], [have_mixer="yes"], [have_mixer="no"], [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/rawmidi.h], [have_rawmidi="yes"], [have_rawmidi="no"], @@ -54,6 +56,7 @@ AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"] [#include <samplerate.h>]) AM_CONDITIONAL(HAVE_PCM, test "$have_pcm" = "yes") +AM_CONDITIONAL(HAVE_COMPRESS, test "$have_compress" = "yes") AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") AM_CONDITIONAL(HAVE_SEQ, test "$have_seq" = "yes") @@ -359,7 +362,8 @@ AC_OUTPUT(Makefile alsactl/Makefile alsactl/init/Makefile \ alsaconf/alsaconf alsaconf/Makefile \ alsaconf/po/Makefile \ alsaucm/Makefile \ - aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ + aplay/Makefile cplay/Makefile \ + include/Makefile iecset/Makefile utils/Makefile \ utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ seq/aplaymidi/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ speaker-test/Makefile speaker-test/samples/Makefile \ diff --git a/cplay/Makefile.am b/cplay/Makefile.am new file mode 100644 index 000000000000..bcb1bfce7a2d --- /dev/null +++ b/cplay/Makefile.am @@ -0,0 +1,14 @@ +LIBRT = @LIBRT@ + +AM_CPPFLAGS = -I$(top_srcdir)/include +LDADD = $(LIBINTL) $(LIBRT) + +# debug flags +#LDFLAGS = -static +#LDADD += -ldl + +bin_PROGRAMS = cplay crecord +noinst_HEADERS = tinymp3.h + +cplay_SOURCES = cplay.c +crecord_SOURCES = crec.c diff --git a/cplay/cplay.c b/cplay/cplay.c new file mode 100644 index 000000000000..a017394e5b4f --- /dev/null +++ b/cplay/cplay.c @@ -0,0 +1,294 @@ +/* + * tinyplay command line player for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * 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> +#include <alsa/asoundlib.h> +#include <sound/compress_params.h> + +#include "tinymp3.h" + +static int verbose; + +static void usage(void) +{ + fprintf(stderr, "usage: cplay [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n\n" + "-v\tverbose mode\n" + "-h\tPrints this help list\n\n" + "Example:\n" + "\tcplay -c 1 -d 2 test.mp3\n" + "\tcplay -f 5 test.mp3\n"); + + exit(EXIT_FAILURE); +} + +void play_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag); + +struct mp3_header { + uint16_t sync; + uint8_t format1; + uint8_t format2; +}; + +int parse_mp3_header(struct mp3_header *header, unsigned int *num_channels, + unsigned int *sample_rate, unsigned int *bit_rate) +{ + int ver_idx, mp3_version, layer, bit_rate_idx, sample_rate_idx, channel_idx; + + /* check sync bits */ + if ((header->sync & MP3_SYNC) != MP3_SYNC) { + fprintf(stderr, "Error: Can't find sync word\n"); + return -1; + } + ver_idx = (header->sync >> 11) & 0x03; + mp3_version = ver_idx == 0 ? MPEG25 : ((ver_idx & 0x1) ? MPEG1 : MPEG2); + layer = 4 - ((header->sync >> 9) & 0x03); + bit_rate_idx = ((header->format1 >> 4) & 0x0f); + sample_rate_idx = ((header->format1 >> 2) & 0x03); + channel_idx = ((header->format2 >> 6) & 0x03); + + if (sample_rate_idx == 3 || layer == 4 || bit_rate_idx == 15) { + fprintf(stderr, "Error: Can't find valid header\n"); + return -1; + } + *num_channels = (channel_idx == MONO ? 1 : 2); + *sample_rate = mp3_sample_rates[mp3_version][sample_rate_idx]; + *bit_rate = (mp3_bit_rates[mp3_version][layer - 1][bit_rate_idx]) * 1000; + if (verbose) + printf("%s: exit\n", __func__); + return 0; +} + +int check_codec_format_supported(unsigned int card, unsigned int device, struct snd_codec *codec) +{ + if (snd_compr_is_codec_supported(card, device, COMPRESS_IN, codec) == false) { + fprintf(stderr, "Error: This codec or format is not supported by DSP\n"); + return -1; + } + return 0; +} + +static int print_time(struct snd_compr *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (snd_compr_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + return -1; + } else + fprintf(stderr, "DSP played %jd.%jd\n", (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + return 0; +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0; + + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvb: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; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + play_samples(file, card, device, buffer_size, frag); + + fprintf(stderr, "Finish Playing.... Close Normally\n"); + exit(EXIT_SUCCESS); +} + +void play_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag) +{ + struct snd_compr_config config; + struct snd_codec codec; + struct snd_compr *compress; + struct mp3_header header; + FILE *file; + char *buffer; + int size, num_read, wrote; + unsigned int channels, rate, bits; + + if (verbose) + printf("%s: entry\n", __func__); + file = fopen(name, "rb"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + fread(&header, sizeof(header), 1, file); + + if (parse_mp3_header(&header, &channels, &rate, &bits) == -1) { + fclose(file); + exit(EXIT_FAILURE); + } + + codec.id = SND_AUDIOCODEC_MP3; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = rate; + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + fclose(file); + exit(EXIT_FAILURE); + } + codec.bit_rate = bits; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = 0; + 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 = snd_compr_open(card, device, COMPRESS_IN, &config); + if (!compress || !snd_compr_is_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", snd_compr_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; + } + + /* we will write frag fragment_size and then start */ + num_read = fread(buffer, 1, size * config.fragments, file); + if (num_read > 0) { + if (verbose) + printf("%s: Doing first buffer write of %d\n", __func__, num_read); + wrote = snd_compr_write(compress, buffer, num_read); + if (wrote < 0) { + fprintf(stderr, "Error %d playing sample\n", wrote); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + goto BUF_EXIT; + } + if (wrote != num_read) { + /* TODO: Buufer pointer needs to be set here */ + fprintf(stderr, "We wrote %d, DSP accepted %d\n", num_read, wrote); + } + } + printf("Playing 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, Bit Rate %d\n", + SND_AUDIOCODEC_MP3, channels, rate, bits); + + snd_compr_start(compress); + if (verbose) + printf("%s: You should hear audio NOW!!!\n", __func__); + + do { + num_read = fread(buffer, 1, size, file); + if (num_read > 0) { + wrote = snd_compr_write(compress, buffer, num_read); + if (wrote < 0) { + fprintf(stderr, "Error playing sample\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + goto BUF_EXIT; + } + if (wrote != num_read) { + /* TODO: Buffer pointer needs to be set here */ + fprintf(stderr, "We wrote %d, DSP accepted %d\n", num_read, wrote); + } + if (verbose) { + print_time(compress); + printf("%s: wrote %d\n", __func__, wrote); + } + } + } while (num_read > 0); + + if (verbose) + printf("%s: exit success\n", __func__); + /* issue drain if it supports */ + snd_compr_drain(compress); + free(buffer); + fclose(file); + snd_compr_close(compress); + return; +BUF_EXIT: + free(buffer); +COMP_EXIT: + snd_compr_close(compress); +FILE_EXIT: + fclose(file); + if (verbose) + printf("%s: exit failure\n", __func__); + exit(EXIT_FAILURE); +} + diff --git a/cplay/crec.c b/cplay/crec.c new file mode 100644 index 000000000000..6a3b4a260ceb --- /dev/null +++ b/cplay/crec.c @@ -0,0 +1,449 @@ +/* + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013-2014, 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> +#include <sys/types.h> +#include <sys/stat.h> +#include <alsa/asoundlib.h> +#include <sound/compress_params.h> + +static int verbose; +static int file; +static FILE *finfo; +static bool streamed; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SND_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: crecord [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" + "If filename is not given the output is\n" + "written to stdout\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 snd_compr *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (snd_compr_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + return -1; + } else { + fprintf(finfo, "DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +static int finish_record() +{ + struct wave_header header; + int ret; + size_t nread, written; + + if (!file) + return -ENOENT; + + /* can't rewind if streaming to stdout */ + if (streamed) + return 0; + + /* Get amount of data written to file */ + ret = lseek(file, 0, SEEK_END); + if (ret < 0) + return -errno; + + written = ret; + if (written < sizeof(header)) + return -ENOENT; + written -= sizeof(header); + + /* Sync file header from file */ + ret = lseek(file, 0, SEEK_SET); + if (ret < 0) + return -errno; + + nread = read(file, &header, sizeof(header)); + if (nread != sizeof(header)) + return -errno; + + /* Update file header */ + ret = lseek(file, 0, SEEK_SET); + if (ret < 0) + return -errno; + + size_wave_header(&header, written); + + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) + return -errno; + + 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 snd_compr_config config; + struct snd_codec codec; + struct snd_compr *compress; + struct wave_header header; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SND_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) + fprintf(finfo, "%s: entry, reading %u bytes\n", __func__, length); + if (!name) { + file = STDOUT_FILENO; + } else { + file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); + if (file == -1) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + } + + /* Write a header, will update with size once record is complete */ + if (!streamed) { + init_wave_header(&header, channels, rate, samplebits); + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { + fprintf(stderr, "Error writing output file header: %s\n", + strerror(errno)); + goto file_exit; + } + } + + memset(&codec, 0, sizeof(codec)); + memset(&config, 0, sizeof(config)); + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = rate; + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } + config.codec = &codec; + + compress = snd_compr_open(card, device, COMPRESS_OUT, &config); + if (!compress || !snd_compr_is_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + goto file_exit; + }; + + if (verbose) + fprintf(finfo, "%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; + } + + fprintf(finfo, "Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + fprintf(finfo, "Codec %u Format %u Channels %u, %u Hz\n", + codec.id, codec.format, codec.ch_out, rate); + + snd_compr_start(compress); + + if (verbose) + fprintf(finfo, "%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = snd_compr_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_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 = write(file, buffer, read); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %s\n", + strerror(errno)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + fprintf(finfo, "%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = snd_compr_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", snd_compr_get_error(compress)); + } + + ret = finish_record(); + if (ret < 0) { + fprintf(stderr, "Failed to finish header: %s\n", strerror(ret)); + goto buf_exit; + } + + if (verbose) + fprintf(finfo, "%s: exit success\n", __func__); + + free(buffer); + close(file); + file = 0; + + snd_compr_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + snd_compr_close(compress); +file_exit: + close(file); + + if (verbose) + fprintf(finfo, "%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +static void sig_handler(int signum __attribute__ ((unused))) +{ + finish_record(); + + if (file) + close(file); + + _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 (signal(SIGINT, sig_handler) == SIG_ERR) { + fprintf(stderr, "Error registering signal handler\n"); + exit(EXIT_FAILURE); + } + + if (argc < 1) + 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 = SND_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SND_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) { + file = NULL; + finfo = fopen("/dev/null", "w"); + streamed = true; + } else { + file = argv[optind]; + finfo = stdout; + streamed = false; + } + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + fprintf(finfo, "Finish capturing... Close Normally\n"); + + exit(EXIT_SUCCESS); +} + diff --git a/cplay/tinymp3.h b/cplay/tinymp3.h new file mode 100644 index 000000000000..13afaec2757d --- /dev/null +++ b/cplay/tinymp3.h @@ -0,0 +1,72 @@ +/* + * mp3 header and parsing + * Copyright (c) 2011-2012, Intel Corporation. + * + * + * 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. + */ + + +#ifndef __TINYMP3_H +#define __TINYMP3_H + +#if defined(__cplusplus) +extern "C" { +#endif + + +#define MP3_SYNC 0xe0ff + +const int mp3_sample_rates[3][3] = { + {44100, 48000, 32000}, /* MPEG-1 */ + {22050, 24000, 16000}, /* MPEG-2 */ + {11025, 12000, 8000}, /* MPEG-2.5 */ +}; + +const int mp3_bit_rates[3][3][15] = { + { + /* MPEG-1 */ + { 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448}, /* Layer 1 */ + { 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384}, /* Layer 2 */ + { 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320}, /* Layer 3 */ + }, + { + /* MPEG-2 */ + { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256}, /* Layer 1 */ + { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, /* Layer 2 */ + { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, /* Layer 3 */ + }, + { + /* MPEG-2.5 */ + { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256}, /* Layer 1 */ + { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, /* Layer 2 */ + { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, /* Layer 3 */ + }, +}; + +enum mpeg_version { + MPEG1 = 0, + MPEG2 = 1, + MPEG25 = 2 +}; + +enum mp3_stereo_mode { + STEREO = 0x00, + JOINT = 0x01, + DUAL = 0x02, + MONO = 0x03 +}; + +#endif

[ALSA-UTILS] Add support for cplay and crecord

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