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297
node_modules/lame/deps/mpg123/src/output/alsa.c
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297
node_modules/lame/deps/mpg123/src/output/alsa.c
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/*
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alsa: sound output with Advanced Linux Sound Architecture 1.x API
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copyright 2006-8 by the mpg123 project - free software under the terms of the LGPL 2.1
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see COPYING and AUTHORS files in distribution or http://mpg123.org
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initially written by Clemens Ladisch <clemens@ladisch.de>
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*/
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#include "mpg123app.h"
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#include "audio.h"
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#include "module.h"
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#include <errno.h>
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/* make ALSA 0.9.x compatible to the 1.0.x API */
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#define ALSA_PCM_NEW_HW_PARAMS_API
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#define ALSA_PCM_NEW_SW_PARAMS_API
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#include <alsa/asoundlib.h>
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#include "debug.h"
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/* My laptop has probs playing low-sampled files with only 0.5s buffer... this should be a user setting -- ThOr */
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#define BUFFER_LENGTH 0.5 /* in seconds */
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static const struct {
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snd_pcm_format_t alsa;
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int mpg123;
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} format_map[] = {
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{ SND_PCM_FORMAT_S16, MPG123_ENC_SIGNED_16 },
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{ SND_PCM_FORMAT_U16, MPG123_ENC_UNSIGNED_16 },
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{ SND_PCM_FORMAT_U8, MPG123_ENC_UNSIGNED_8 },
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{ SND_PCM_FORMAT_S8, MPG123_ENC_SIGNED_8 },
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{ SND_PCM_FORMAT_A_LAW, MPG123_ENC_ALAW_8 },
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{ SND_PCM_FORMAT_MU_LAW, MPG123_ENC_ULAW_8 },
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{ SND_PCM_FORMAT_S32, MPG123_ENC_SIGNED_32 },
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{ SND_PCM_FORMAT_U32, MPG123_ENC_UNSIGNED_32 },
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#ifdef WORDS_BIGENDIAN
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{ SND_PCM_FORMAT_S24_3BE, MPG123_ENC_SIGNED_24 },
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{ SND_PCM_FORMAT_U24_3BE, MPG123_ENC_UNSIGNED_24 },
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#else
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{ SND_PCM_FORMAT_S24_3LE, MPG123_ENC_SIGNED_24 },
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{ SND_PCM_FORMAT_U24_3LE, MPG123_ENC_UNSIGNED_24 },
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#endif
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{ SND_PCM_FORMAT_FLOAT, MPG123_ENC_FLOAT_32 },
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{ SND_PCM_FORMAT_FLOAT64, MPG123_ENC_FLOAT_64 }
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};
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#define NUM_FORMATS (sizeof format_map / sizeof format_map[0])
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static int rates_match(long int desired, unsigned int actual)
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{
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return actual * 100 > desired * (100 - AUDIO_RATE_TOLERANCE) &&
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actual * 100 < desired * (100 + AUDIO_RATE_TOLERANCE);
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}
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static int initialize_device(audio_output_t *ao)
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{
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snd_pcm_hw_params_t *hw=NULL;
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snd_pcm_sw_params_t *sw=NULL;
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snd_pcm_uframes_t buffer_size;
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snd_pcm_uframes_t period_size;
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snd_pcm_format_t format;
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snd_pcm_t *pcm=(snd_pcm_t*)ao->userptr;
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unsigned int rate;
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int i;
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snd_pcm_hw_params_alloca(&hw); /* Ignore GCC warning here... alsa-lib>=1.0.16 doesn't trigger that anymore, too. */
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if (snd_pcm_hw_params_any(pcm, hw) < 0) {
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if(!AOQUIET) error("initialize_device(): no configuration available");
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return -1;
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}
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if (snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
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if(!AOQUIET) error("initialize_device(): device does not support interleaved access");
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return -1;
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}
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format = SND_PCM_FORMAT_UNKNOWN;
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for (i = 0; i < NUM_FORMATS; ++i) {
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if (ao->format == format_map[i].mpg123) {
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format = format_map[i].alsa;
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break;
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}
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}
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if (format == SND_PCM_FORMAT_UNKNOWN) {
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if(!AOQUIET) error1("initialize_device(): invalid sample format %d", ao->format);
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errno = EINVAL;
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return -1;
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}
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if (snd_pcm_hw_params_set_format(pcm, hw, format) < 0) {
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if(!AOQUIET) error1("initialize_device(): cannot set format %s", snd_pcm_format_name(format));
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return -1;
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}
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if (snd_pcm_hw_params_set_channels(pcm, hw, ao->channels) < 0) {
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if(!AOQUIET) error1("initialize_device(): cannot set %d channels", ao->channels);
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return -1;
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}
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rate = ao->rate;
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if (snd_pcm_hw_params_set_rate_near(pcm, hw, &rate, NULL) < 0) {
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if(!AOQUIET) error1("initialize_device(): cannot set rate %u", rate);
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return -1;
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}
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if (!rates_match(ao->rate, rate)) {
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if(!AOQUIET) error2("initialize_device(): rate %ld not available, using %u", ao->rate, rate);
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/* return -1; */
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}
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buffer_size = rate * BUFFER_LENGTH;
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if (snd_pcm_hw_params_set_buffer_size_near(pcm, hw, &buffer_size) < 0) {
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if(!AOQUIET) error("initialize_device(): cannot set buffer size");
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return -1;
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}
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period_size = buffer_size / 4;
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if (snd_pcm_hw_params_set_period_size_near(pcm, hw, &period_size, NULL) < 0) {
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if(!AOQUIET) error("initialize_device(): cannot set period size");
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return -1;
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}
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if (snd_pcm_hw_params(pcm, hw) < 0) {
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if(!AOQUIET) error("initialize_device(): cannot set hw params");
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return -1;
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}
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snd_pcm_sw_params_alloca(&sw);
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if (snd_pcm_sw_params_current(pcm, sw) < 0) {
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if(!AOQUIET) error("initialize_device(): cannot get sw params");
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return -1;
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}
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/* start playing after the first write */
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if (snd_pcm_sw_params_set_start_threshold(pcm, sw, 1) < 0) {
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if(!AOQUIET) error("initialize_device(): cannot set start threshold");
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return -1;
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}
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/* wake up on every interrupt */
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if (snd_pcm_sw_params_set_avail_min(pcm, sw, 1) < 0) {
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if(!AOQUIET) error("initialize_device(): cannot set min available");
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return -1;
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}
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#if SND_LIB_VERSION < ((1<<16)|16)
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/* Always write as many frames as possible (deprecated since alsa-lib 1.0.16) */
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if (snd_pcm_sw_params_set_xfer_align(pcm, sw, 1) < 0) {
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if(!AOQUIET) error("initialize_device(): cannot set transfer alignment");
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return -1;
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}
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#endif
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if (snd_pcm_sw_params(pcm, sw) < 0) {
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if(!AOQUIET) error("initialize_device(): cannot set sw params");
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return -1;
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}
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return 0;
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}
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static void error_ignorer(const char *file, int line, const char *function, int err, const char *fmt,...)
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{
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/* I can make ALSA silent. */
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}
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static int open_alsa(audio_output_t *ao)
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{
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const char *pcm_name;
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snd_pcm_t *pcm=NULL;
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debug1("open_alsa with %p", ao->userptr);
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if(AOQUIET) snd_lib_error_set_handler(error_ignorer);
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pcm_name = ao->device ? ao->device : "default";
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if (snd_pcm_open(&pcm, pcm_name, SND_PCM_STREAM_PLAYBACK, 0) < 0) {
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if(!AOQUIET) error1("cannot open device %s", pcm_name);
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return -1;
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}
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ao->userptr = pcm;
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if (ao->format != -1) {
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/* we're going to play: initalize sample format */
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return initialize_device(ao);
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} else {
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/* query mode; sample format will be set for each query */
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return 0;
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}
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}
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static int get_formats_alsa(audio_output_t *ao)
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{
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snd_pcm_t *pcm=(snd_pcm_t*)ao->userptr;
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snd_pcm_hw_params_t *hw;
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unsigned int rate;
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int supported_formats, i;
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snd_pcm_hw_params_alloca(&hw);
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if (snd_pcm_hw_params_any(pcm, hw) < 0) {
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if(!AOQUIET) error("get_formats_alsa(): no configuration available");
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return -1;
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}
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if (snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED) < 0)
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return -1;
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if (snd_pcm_hw_params_set_channels(pcm, hw, ao->channels) < 0)
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return 0;
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rate = ao->rate;
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if (snd_pcm_hw_params_set_rate_near(pcm, hw, &rate, NULL) < 0)
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return -1;
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if (!rates_match(ao->rate, rate))
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return 0;
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supported_formats = 0;
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for (i = 0; i < NUM_FORMATS; ++i) {
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if (snd_pcm_hw_params_test_format(pcm, hw, format_map[i].alsa) == 0)
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supported_formats |= format_map[i].mpg123;
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}
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return supported_formats;
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}
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static int write_alsa(audio_output_t *ao, unsigned char *buf, int bytes)
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{
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snd_pcm_t *pcm=(snd_pcm_t*)ao->userptr;
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snd_pcm_uframes_t frames;
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snd_pcm_sframes_t written;
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frames = snd_pcm_bytes_to_frames(pcm, bytes);
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written = snd_pcm_writei(pcm, buf, frames);
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if (written == -EINTR) /* interrupted system call */
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written = 0;
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else if (written == -EPIPE) { /* underrun */
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if (snd_pcm_prepare(pcm) >= 0)
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written = snd_pcm_writei(pcm, buf, frames);
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}
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if (written >= 0)
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return snd_pcm_frames_to_bytes(pcm, written);
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else
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{
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if(snd_pcm_state(pcm) == SND_PCM_STATE_SUSPENDED)
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{
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/* Iamnothappyabouthisnothappyreallynot. */
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snd_pcm_resume(pcm);
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if(snd_pcm_state(pcm) == SND_PCM_STATE_SUSPENDED)
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{
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error("device still suspended after resume hackery... giving up");
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return -1;
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}
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}
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return 0;
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}
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}
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static void flush_alsa(audio_output_t *ao)
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{
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snd_pcm_t *pcm=(snd_pcm_t*)ao->userptr;
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/* is this the optimal solution? - we should figure out what we really whant from this function */
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debug("alsa drop");
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snd_pcm_drop(pcm);
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debug("alsa prepare");
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snd_pcm_prepare(pcm);
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debug("alsa flush done");
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}
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static int close_alsa(audio_output_t *ao)
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{
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snd_pcm_t *pcm=(snd_pcm_t*)ao->userptr;
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debug1("close_alsa with %p", ao->userptr);
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if(pcm != NULL) /* be really generous for being called without any device opening */
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{
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if (snd_pcm_state(pcm) == SND_PCM_STATE_RUNNING)
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snd_pcm_drain(pcm);
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ao->userptr = NULL; /* Should alsa do this or the module wrapper? */
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return snd_pcm_close(pcm);
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}
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else return 0;
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}
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static int init_alsa(audio_output_t* ao)
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{
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if (ao==NULL) return -1;
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/* Set callbacks */
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ao->open = open_alsa;
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ao->flush = flush_alsa;
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ao->write = write_alsa;
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ao->get_formats = get_formats_alsa;
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ao->close = close_alsa;
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/* Success */
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return 0;
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}
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/*
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Module information data structure
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*/
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mpg123_module_t mpg123_output_module_info = {
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/* api_version */ MPG123_MODULE_API_VERSION,
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/* name */ "alsa",
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/* description */ "Output audio using Advanced Linux Sound Architecture (ALSA).",
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/* revision */ "$Rev:$",
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/* handle */ NULL,
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/* init_output */ init_alsa,
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};
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