implemented alsa audioOutput plugin, now it needs testing
git-svn-id: https://svn.musicpd.org/mpd/trunk@3008 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes
parent
82ff4c321c
commit
d1a4bb382f
4
TODO
4
TODO
@ -22,6 +22,10 @@
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*) add a method for clearling the audio device's buffer. This isn't possible
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with libao api.
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*) add support so that audioOutput plugins can modify the output audio format.
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(This way, alsa's _near functions can be used to adjust for output
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devices on the fly: for channels, bits, and rate)
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*) add support for playing aac streams (gee, thanks icecast)
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*) implement apev2 and id3v1 tag reader from xmms-musepack plugin
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@ -2,6 +2,7 @@ bin_PROGRAMS = mpd
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SUBDIRS = $(ID3_SUBDIR) $(MAD_SUBDIR) $(MP4FF_SUBDIR)
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mpd_audioOutputs = \
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audioOutputs/audioOutput_alsa.c \
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audioOutputs/audioOutput_ao.c \
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audioOutputs/audioOutput_oss.c \
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audioOutputs/audioOutput_shout.c
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@ -56,9 +56,10 @@ int cmpAudioFormat(AudioFormat * f1, AudioFormat * f2) {
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return memcmp(f1, f2, sizeof(AudioFormat));
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}
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extern AudioOutputPlugin alsaPlugin;
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extern AudioOutputPlugin aoPlugin;
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extern AudioOutputPlugin shoutPlugin;
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extern AudioOutputPlugin ossPlugin;
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extern AudioOutputPlugin shoutPlugin;
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/* make sure initPlayerData is called before this function!! */
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void initAudioDriver() {
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@ -66,9 +67,10 @@ void initAudioDriver() {
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int i;
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initAudioOutputPlugins();
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loadAudioOutputPlugin(&alsaPlugin);
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loadAudioOutputPlugin(&aoPlugin);
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loadAudioOutputPlugin(&shoutPlugin);
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loadAudioOutputPlugin(&ossPlugin);
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loadAudioOutputPlugin(&shoutPlugin);
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pdAudioDevicesEnabled = (getPlayerData())->audioDeviceEnabled;
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293
src/audioOutputs/audioOutput_alsa.c
Normal file
293
src/audioOutputs/audioOutput_alsa.c
Normal file
@ -0,0 +1,293 @@
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/* the Music Player Daemon (MPD)
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* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
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* This project's homepage is: http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "../audioOutput.h"
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#include <stdlib.h>
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#ifdef HAVE_ALSA
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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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#define MPD_ALSA_BUFFER_TIME 500000
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#define MPD_ALSA_PERIOD_TIME 50000
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#include "../conf.h"
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#include "../log.h"
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#include "../sig_handlers.h"
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#include <string.h>
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#include <assert.h>
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#include <signal.h>
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#include <alsa/asoundlib.h>
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typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer,
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snd_pcm_uframes_t size);
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typedef struct _AlsaData {
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char * device;
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snd_pcm_t * pcm_handle;
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int mmap;
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alsa_writei_t * writei;
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} AlsaData;
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static AlsaData * newAlsaData() {
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AlsaData * ret = malloc(sizeof(AlsaData));
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ret->device = NULL;
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ret->pcm_handle = NULL;
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ret->writei = snd_pcm_writei;
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ret->mmap = 0;
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return ret;
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}
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static void freeAlsaData(AlsaData * ad) {
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if(ad->device) free(ad->device);
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free(ad);
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}
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static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) {
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BlockParam * bp = getBlockParam(param, "device");
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AlsaData * ad = newAlsaData();
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audioOutput->data = ad;
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ad->device = bp ? strdup(bp->value) : strdup("default");
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return 0;
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}
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static void alsa_finishDriver(AudioOutput * audioOutput) {
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AlsaData * ad = audioOutput->data;
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freeAlsaData(ad);
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}
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static int alsa_openDevice(AudioOutput * audioOutput)
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{
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AlsaData * ad = audioOutput->data;
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AudioFormat * audioFormat = &audioOutput->outAudioFormat;
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snd_pcm_format_t bitformat;
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snd_pcm_hw_params_t * hwparams;
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snd_pcm_sw_params_t * swparams;
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unsigned int sampleRate = audioFormat->sampleRate;
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snd_pcm_uframes_t alsa_buffer_size;
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snd_pcm_uframes_t alsa_period_size;
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unsigned int alsa_buffer_time = MPD_ALSA_BUFFER_TIME;
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unsigned int alsa_period_time = MPD_ALSA_PERIOD_TIME;
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int err;
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switch(audioFormat->bits) {
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case 8:
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bitformat = SND_PCM_FORMAT_S8;
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break;
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case 16:
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bitformat = SND_PCM_FORMAT_S16;
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break;
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case 24:
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bitformat = SND_PCM_FORMAT_S16;
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break;
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case 32:
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bitformat = SND_PCM_FORMAT_S16;
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break;
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default:
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ERROR("Alsa device \"%s\" doesn't support %i bit audio\n",
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ad->device, audioFormat->bits);
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return -1;
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}
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err = snd_pcm_open(&ad->pcm_handle, ad->device,
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SND_PCM_STREAM_PLAYBACK, 0);
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if(err < 0) {
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ad->pcm_handle = NULL;
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goto error;
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}
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err = snd_pcm_nonblock(ad->pcm_handle, 0);
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if(err < 0) goto error;
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// configure HW params
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snd_pcm_hw_params_alloca(&hwparams);
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err = snd_pcm_hw_params_any(ad->pcm_handle, hwparams);
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if(err < 0) goto error;
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if(ad->mmap) {
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err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams,
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SND_PCM_ACCESS_MMAP_INTERLEAVED);
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if(err < 0) {
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ERROR("Cannot set mmap'ed mode on alsa device \"%s\": "
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" %s\n", ad->device,
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snd_strerror(-err));
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ERROR("Falling back to direct write mode\n");
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ad->mmap = 0;
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}
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else ad->writei = snd_pcm_mmap_writei;
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}
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if(!ad->mmap) {
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err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams,
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SND_PCM_ACCESS_RW_INTERLEAVED);
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if(err < 0) goto error;
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ad->writei = snd_pcm_mmap_writei;
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}
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err = snd_pcm_hw_params_set_format(ad->pcm_handle, hwparams, bitformat);
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if(err < 0) {
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ERROR("Alsa device \"%s\" does not support %i bit audio: "
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"%s\n", ad->device, (int)bitformat,
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snd_strerror(-err));
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goto fail;
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}
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err = snd_pcm_hw_params_set_channels(ad->pcm_handle, hwparams,
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(unsigned int)audioFormat->channels);
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if(err < 0) {
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ERROR("Alsa device \"%s\" does not support %i channels: "
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"%s\n", ad->device, (int)audioFormat->channels,
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snd_strerror(-err));
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goto fail;
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}
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err = snd_pcm_hw_params_set_rate_near(ad->pcm_handle, hwparams,
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&sampleRate, 0);
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if(err < 0 || sampleRate == 0) {
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ERROR("Alsa device \"%s\" does not support %i Hz audio\n",
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ad->device, (int)audioFormat->sampleRate);
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goto fail;
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}
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err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm_handle, hwparams,
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&alsa_buffer_time, 0);
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if(err < 0) goto error;
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err = snd_pcm_hw_params_set_period_time_near(ad->pcm_handle, hwparams,
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&alsa_period_time, 0);
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if(err < 0) goto error;
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err = snd_pcm_hw_params(ad->pcm_handle, hwparams);
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if(err < 0) goto error;
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err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
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if(err < 0) goto error;
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err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0);
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if(err < 0) goto error;
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// configure SW params
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snd_pcm_sw_params_alloca(&swparams);
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snd_pcm_sw_params_current(ad->pcm_handle, swparams);
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err = snd_pcm_sw_params_set_start_threshold(ad->pcm_handle, swparams,
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alsa_buffer_size - alsa_period_size);
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if(err < 0) goto error;
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err = snd_pcm_sw_params(ad->pcm_handle, swparams);
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if(err < 0) goto error;
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audioOutput->open = 1;
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return 0;
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error:
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ERROR("Error opening alsa device \"%s\": %s\n", ad->device,
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snd_strerror(-err));
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fail:
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if(ad->pcm_handle) snd_pcm_close(ad->pcm_handle);
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audioOutput->open = 0;
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return -1;
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}
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static void alsa_closeDevice(AudioOutput * audioOutput) {
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AlsaData * ad = audioOutput->data;
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if(ad->pcm_handle) {
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snd_pcm_drain(ad->pcm_handle);
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ad->pcm_handle = NULL;
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}
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audioOutput->open = 0;
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}
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inline static int alsa_errorRecovery(AlsaData * ad, int err) {
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if(err == -EPIPE) {
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DEBUG("Underrun on alsa device \"%s\"\n", ad->device);
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err = snd_pcm_prepare(ad->pcm_handle);
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if(err < 0) return -1;
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return 0;
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}
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return err;
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}
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static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk,
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int size)
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{
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AlsaData * ad = audioOutput->data;
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int ret;
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while (size > 0) {
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ret = ad->writei(ad->pcm_handle, playChunk, size);
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if(ret == -EAGAIN) continue;
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if(ret < 0 && alsa_errorRecovery(ad, ret) < 0) {
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ERROR("closing alsa device \"%s\" due to write error:"
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" %s\n", ad->device,
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snd_strerror(-errno));
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alsa_closeDevice(audioOutput);
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return -1;
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}
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playChunk += ret;
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size -= ret;
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}
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return 0;
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}
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AudioOutputPlugin alsaPlugin =
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{
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"alsa",
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alsa_initDriver,
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alsa_finishDriver,
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alsa_openDevice,
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alsa_playAudio,
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alsa_closeDevice,
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NULL /* sendMetadataFunc */
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};
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#else /* HAVE ALSA */
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AudioOutputPlugin alsaPlugin =
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{
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NULL,
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NULL,
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NULL,
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NULL,
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NULL,
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NULL,
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NULL /* sendMetadataFunc */
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};
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#endif /* HAVE_ALSA */
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