diff options
Diffstat (limited to '')
| -rw-r--r-- | TODO | 4 | ||||
| -rw-r--r-- | src/Makefile.am | 1 | ||||
| -rw-r--r-- | src/audio.c | 6 | ||||
| -rw-r--r-- | src/audioOutputs/audioOutput_alsa.c | 293 |
4 files changed, 302 insertions, 2 deletions
@@ -22,6 +22,10 @@ *) add a method for clearling the audio device's buffer. This isn't possible with libao api. +*) add support so that audioOutput plugins can modify the output audio format. + (This way, alsa's _near functions can be used to adjust for output + devices on the fly: for channels, bits, and rate) + *) add support for playing aac streams (gee, thanks icecast) *) implement apev2 and id3v1 tag reader from xmms-musepack plugin diff --git a/src/Makefile.am b/src/Makefile.am index 595259c26..ee52e9605 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -2,6 +2,7 @@ bin_PROGRAMS = mpd SUBDIRS = $(ID3_SUBDIR) $(MAD_SUBDIR) $(MP4FF_SUBDIR) mpd_audioOutputs = \ + audioOutputs/audioOutput_alsa.c \ audioOutputs/audioOutput_ao.c \ audioOutputs/audioOutput_oss.c \ audioOutputs/audioOutput_shout.c diff --git a/src/audio.c b/src/audio.c index 02974228a..cf2b54c20 100644 --- a/src/audio.c +++ b/src/audio.c @@ -56,9 +56,10 @@ int cmpAudioFormat(AudioFormat * f1, AudioFormat * f2) { return memcmp(f1, f2, sizeof(AudioFormat)); } +extern AudioOutputPlugin alsaPlugin; extern AudioOutputPlugin aoPlugin; -extern AudioOutputPlugin shoutPlugin; extern AudioOutputPlugin ossPlugin; +extern AudioOutputPlugin shoutPlugin; /* make sure initPlayerData is called before this function!! */ void initAudioDriver() { @@ -66,9 +67,10 @@ void initAudioDriver() { int i; initAudioOutputPlugins(); + loadAudioOutputPlugin(&alsaPlugin); loadAudioOutputPlugin(&aoPlugin); - loadAudioOutputPlugin(&shoutPlugin); loadAudioOutputPlugin(&ossPlugin); + loadAudioOutputPlugin(&shoutPlugin); pdAudioDevicesEnabled = (getPlayerData())->audioDeviceEnabled; diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c new file mode 100644 index 000000000..63217e9fe --- /dev/null +++ b/src/audioOutputs/audioOutput_alsa.c @@ -0,0 +1,293 @@ +/* the Music Player Daemon (MPD) + * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu) + * This project's homepage is: http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include "../audioOutput.h" + +#include <stdlib.h> + +#ifdef HAVE_ALSA + +#define ALSA_PCM_NEW_HW_PARAMS_API +#define ALSA_PCM_NEW_SW_PARAMS_API + +#define MPD_ALSA_BUFFER_TIME 500000 +#define MPD_ALSA_PERIOD_TIME 50000 + +#include "../conf.h" +#include "../log.h" +#include "../sig_handlers.h" + +#include <string.h> +#include <assert.h> +#include <signal.h> + +#include <alsa/asoundlib.h> + +typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer, + snd_pcm_uframes_t size); + +typedef struct _AlsaData { + char * device; + snd_pcm_t * pcm_handle; + int mmap; + alsa_writei_t * writei; +} AlsaData; + +static AlsaData * newAlsaData() { + AlsaData * ret = malloc(sizeof(AlsaData)); + + ret->device = NULL; + ret->pcm_handle = NULL; + ret->writei = snd_pcm_writei; + ret->mmap = 0; + + return ret; +} + +static void freeAlsaData(AlsaData * ad) { + if(ad->device) free(ad->device); + + free(ad); +} + +static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) { + BlockParam * bp = getBlockParam(param, "device"); + AlsaData * ad = newAlsaData(); + + audioOutput->data = ad; + + ad->device = bp ? strdup(bp->value) : strdup("default"); + + return 0; +} + +static void alsa_finishDriver(AudioOutput * audioOutput) { + AlsaData * ad = audioOutput->data; + + freeAlsaData(ad); +} + +static int alsa_openDevice(AudioOutput * audioOutput) +{ + AlsaData * ad = audioOutput->data; + AudioFormat * audioFormat = &audioOutput->outAudioFormat; + snd_pcm_format_t bitformat; + snd_pcm_hw_params_t * hwparams; + snd_pcm_sw_params_t * swparams; + unsigned int sampleRate = audioFormat->sampleRate; + snd_pcm_uframes_t alsa_buffer_size; + snd_pcm_uframes_t alsa_period_size; + unsigned int alsa_buffer_time = MPD_ALSA_BUFFER_TIME; + unsigned int alsa_period_time = MPD_ALSA_PERIOD_TIME; + int err; + + switch(audioFormat->bits) { + case 8: + bitformat = SND_PCM_FORMAT_S8; + break; + case 16: + bitformat = SND_PCM_FORMAT_S16; + break; + case 24: + bitformat = SND_PCM_FORMAT_S16; + break; + case 32: + bitformat = SND_PCM_FORMAT_S16; + break; + default: + ERROR("Alsa device \"%s\" doesn't support %i bit audio\n", + ad->device, audioFormat->bits); + return -1; + } + + err = snd_pcm_open(&ad->pcm_handle, ad->device, + SND_PCM_STREAM_PLAYBACK, 0); + if(err < 0) { + ad->pcm_handle = NULL; + goto error; + } + + err = snd_pcm_nonblock(ad->pcm_handle, 0); + if(err < 0) goto error; + + // configure HW params + snd_pcm_hw_params_alloca(&hwparams); + + err = snd_pcm_hw_params_any(ad->pcm_handle, hwparams); + if(err < 0) goto error; + + if(ad->mmap) { + err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams, + SND_PCM_ACCESS_MMAP_INTERLEAVED); + if(err < 0) { + ERROR("Cannot set mmap'ed mode on alsa device \"%s\": " + " %s\n", ad->device, + snd_strerror(-err)); + ERROR("Falling back to direct write mode\n"); + ad->mmap = 0; + } + else ad->writei = snd_pcm_mmap_writei; + } + + if(!ad->mmap) { + err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams, + SND_PCM_ACCESS_RW_INTERLEAVED); + if(err < 0) goto error; + ad->writei = snd_pcm_mmap_writei; + } + + err = snd_pcm_hw_params_set_format(ad->pcm_handle, hwparams, bitformat); + if(err < 0) { + ERROR("Alsa device \"%s\" does not support %i bit audio: " + "%s\n", ad->device, (int)bitformat, + snd_strerror(-err)); + goto fail; + } + + err = snd_pcm_hw_params_set_channels(ad->pcm_handle, hwparams, + (unsigned int)audioFormat->channels); + if(err < 0) { + ERROR("Alsa device \"%s\" does not support %i channels: " + "%s\n", ad->device, (int)audioFormat->channels, + snd_strerror(-err)); + goto fail; + } + + err = snd_pcm_hw_params_set_rate_near(ad->pcm_handle, hwparams, + &sampleRate, 0); + if(err < 0 || sampleRate == 0) { + ERROR("Alsa device \"%s\" does not support %i Hz audio\n", + ad->device, (int)audioFormat->sampleRate); + goto fail; + } + + err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm_handle, hwparams, + &alsa_buffer_time, 0); + if(err < 0) goto error; + + err = snd_pcm_hw_params_set_period_time_near(ad->pcm_handle, hwparams, + &alsa_period_time, 0); + if(err < 0) goto error; + + err = snd_pcm_hw_params(ad->pcm_handle, hwparams); + if(err < 0) goto error; + + err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size); + if(err < 0) goto error; + + err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0); + if(err < 0) goto error; + + // configure SW params + snd_pcm_sw_params_alloca(&swparams); + snd_pcm_sw_params_current(ad->pcm_handle, swparams); + + err = snd_pcm_sw_params_set_start_threshold(ad->pcm_handle, swparams, + alsa_buffer_size - alsa_period_size); + if(err < 0) goto error; + + err = snd_pcm_sw_params(ad->pcm_handle, swparams); + if(err < 0) goto error; + + audioOutput->open = 1; + + return 0; + +error: + ERROR("Error opening alsa device \"%s\": %s\n", ad->device, + snd_strerror(-err)); +fail: + if(ad->pcm_handle) snd_pcm_close(ad->pcm_handle); + audioOutput->open = 0; + return -1; +} + +static void alsa_closeDevice(AudioOutput * audioOutput) { + AlsaData * ad = audioOutput->data; + + if(ad->pcm_handle) { + snd_pcm_drain(ad->pcm_handle); + ad->pcm_handle = NULL; + } + + audioOutput->open = 0; +} + +inline static int alsa_errorRecovery(AlsaData * ad, int err) { + if(err == -EPIPE) { + DEBUG("Underrun on alsa device \"%s\"\n", ad->device); + err = snd_pcm_prepare(ad->pcm_handle); + if(err < 0) return -1; + return 0; + } + + return err; +} + +static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk, + int size) +{ + AlsaData * ad = audioOutput->data; + int ret; + + while (size > 0) { + ret = ad->writei(ad->pcm_handle, playChunk, size); + + if(ret == -EAGAIN) continue; + + if(ret < 0 && alsa_errorRecovery(ad, ret) < 0) { + ERROR("closing alsa device \"%s\" due to write error:" + " %s\n", ad->device, + snd_strerror(-errno)); + alsa_closeDevice(audioOutput); + return -1; + } + playChunk += ret; + size -= ret; + } + + return 0; +} + +AudioOutputPlugin alsaPlugin = +{ + "alsa", + alsa_initDriver, + alsa_finishDriver, + alsa_openDevice, + alsa_playAudio, + alsa_closeDevice, + NULL /* sendMetadataFunc */ +}; + +#else /* HAVE ALSA */ + +AudioOutputPlugin alsaPlugin = +{ + NULL, + NULL, + NULL, + NULL, + NULL, + NULL, + NULL /* sendMetadataFunc */ +}; + +#endif /* HAVE_ALSA */ + + |