From 4e05a161e54fe05902b99eff521aa0759b102f05 Mon Sep 17 00:00:00 2001 From: "J. Alexander Treuman" Date: Mon, 28 May 2007 15:50:45 +0000 Subject: Making branch for 0.13.0 fixes. git-svn-id: https://svn.musicpd.org/mpd/branches/branch-0.13.0-fixes@6330 09075e82-0dd4-0310-85a5-a0d7c8717e4f --- trunk/src/audioOutputs/audioOutput_alsa.c | 427 ------------------- trunk/src/audioOutputs/audioOutput_ao.c | 246 ----------- trunk/src/audioOutputs/audioOutput_jack.c | 440 -------------------- trunk/src/audioOutputs/audioOutput_mvp.c | 284 ------------- trunk/src/audioOutputs/audioOutput_oss.c | 575 -------------------------- trunk/src/audioOutputs/audioOutput_osx.c | 374 ----------------- trunk/src/audioOutputs/audioOutput_pulse.c | 221 ---------- trunk/src/audioOutputs/audioOutput_shout.c | 636 ----------------------------- 8 files changed, 3203 deletions(-) delete mode 100644 trunk/src/audioOutputs/audioOutput_alsa.c delete mode 100644 trunk/src/audioOutputs/audioOutput_ao.c delete mode 100644 trunk/src/audioOutputs/audioOutput_jack.c delete mode 100644 trunk/src/audioOutputs/audioOutput_mvp.c delete mode 100644 trunk/src/audioOutputs/audioOutput_oss.c delete mode 100644 trunk/src/audioOutputs/audioOutput_osx.c delete mode 100644 trunk/src/audioOutputs/audioOutput_pulse.c delete mode 100644 trunk/src/audioOutputs/audioOutput_shout.c (limited to 'trunk/src/audioOutputs') diff --git a/trunk/src/audioOutputs/audioOutput_alsa.c b/trunk/src/audioOutputs/audioOutput_alsa.c deleted file mode 100644 index 3ade3df46..000000000 --- a/trunk/src/audioOutputs/audioOutput_alsa.c +++ /dev/null @@ -1,427 +0,0 @@ -/* the Music Player Daemon (MPD) - * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) - * 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 - -#ifdef HAVE_ALSA - -#define ALSA_PCM_NEW_HW_PARAMS_API -#define ALSA_PCM_NEW_SW_PARAMS_API - -#define MPD_ALSA_BUFFER_TIME_US 500000 -/* the default period time of xmms is 50 ms, so let's use that as well. - * a user can tweak this parameter via the "period_time" config parameter. - */ -#define MPD_ALSA_PERIOD_TIME_US 50000 -#define MPD_ALSA_RETRY_NR 5 - -#include "../conf.h" -#include "../log.h" - -#include - -#include - -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 *pcmHandle; - alsa_writei_t *writei; - unsigned int buffer_time; - unsigned int period_time; - int sampleSize; - int useMmap; - int canPause; - int canResume; -} AlsaData; - -static AlsaData *newAlsaData(void) -{ - AlsaData *ret = xmalloc(sizeof(AlsaData)); - - ret->device = NULL; - ret->pcmHandle = NULL; - ret->writei = snd_pcm_writei; - ret->useMmap = 0; - ret->buffer_time = MPD_ALSA_BUFFER_TIME_US; - ret->period_time = MPD_ALSA_PERIOD_TIME_US; - - return ret; -} - -static void freeAlsaData(AlsaData * ad) -{ - if (ad->device) - free(ad->device); - - free(ad); -} - -static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) -{ - AlsaData *ad = newAlsaData(); - - if (param) { - BlockParam *bp = getBlockParam(param, "device"); - ad->device = bp ? xstrdup(bp->value) : xstrdup("default"); - - if ((bp = getBlockParam(param, "use_mmap")) && - !strcasecmp(bp->value, "yes")) - ad->useMmap = 1; - if ((bp = getBlockParam(param, "buffer_time"))) - ad->buffer_time = atoi(bp->value); - if ((bp = getBlockParam(param, "period_time"))) - ad->period_time = atoi(bp->value); - } else - ad->device = xstrdup("default"); - audioOutput->data = ad; - - return 0; -} - -static void alsa_finishDriver(AudioOutput * audioOutput) -{ - AlsaData *ad = audioOutput->data; - - freeAlsaData(ad); -} - -static int alsa_testDefault(void) -{ - snd_pcm_t *handle; - - int ret = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, - SND_PCM_NONBLOCK); - snd_config_update_free_global(); - - if (ret) { - WARNING("Error opening default alsa device: %s\n", - snd_strerror(-ret)); - return -1; - } else - snd_pcm_close(handle); - - return 0; -} - -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; - unsigned int channels = audioFormat->channels; - snd_pcm_uframes_t alsa_buffer_size; - snd_pcm_uframes_t alsa_period_size; - int err; - const char *cmd = NULL; - int retry = MPD_ALSA_RETRY_NR; - unsigned int period_time, period_time_ro; - unsigned int buffer_time; - - 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_S24; - break; - case 32: - bitformat = SND_PCM_FORMAT_S32; - break; - default: - ERROR("ALSA device \"%s\" doesn't support %i bit audio\n", - ad->device, audioFormat->bits); - return -1; - } - - err = snd_pcm_open(&ad->pcmHandle, ad->device, - SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); - snd_config_update_free_global(); - if (err < 0) { - ad->pcmHandle = NULL; - goto error; - } - - cmd = "snd_pcm_nonblock"; - err = snd_pcm_nonblock(ad->pcmHandle, 0); - if (err < 0) - goto error; - - period_time_ro = period_time = ad->period_time; -configure_hw: - /* configure HW params */ - snd_pcm_hw_params_alloca(&hwparams); - - cmd = "snd_pcm_hw_params_any"; - err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams); - if (err < 0) - goto error; - - if (ad->useMmap) { - err = snd_pcm_hw_params_set_access(ad->pcmHandle, 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->useMmap = 0; - } else - ad->writei = snd_pcm_mmap_writei; - } - - if (!ad->useMmap) { - cmd = "snd_pcm_hw_params_set_access"; - err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, - SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) - goto error; - ad->writei = snd_pcm_writei; - } - - err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat); - if (err < 0) { - ERROR("ALSA device \"%s\" does not support %i bit audio: " - "%s\n", ad->device, audioFormat->bits, snd_strerror(-err)); - goto fail; - } - - err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams, - &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; - } - audioFormat->channels = channels; - - err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams, - &sampleRate, NULL); - if (err < 0 || sampleRate == 0) { - ERROR("ALSA device \"%s\" does not support %i Hz audio\n", - ad->device, (int)audioFormat->sampleRate); - goto fail; - } - audioFormat->sampleRate = sampleRate; - - buffer_time = ad->buffer_time; - cmd = "snd_pcm_hw_params_set_buffer_time_near"; - err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams, - &buffer_time, NULL); - if (err < 0) - goto error; - - period_time = period_time_ro; - cmd = "snd_pcm_hw_params_set_period_time_near"; - err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams, - &period_time, NULL); - if (err < 0) - goto error; - - cmd = "snd_pcm_hw_params"; - err = snd_pcm_hw_params(ad->pcmHandle, hwparams); - if (err == -EPIPE && --retry > 0) { - period_time_ro = period_time_ro >> 1; - goto configure_hw; - } else if (err < 0) - goto error; - if (retry != MPD_ALSA_RETRY_NR) - DEBUG("ALSA period_time set to %d\n", period_time); - - cmd = "snd_pcm_hw_params_get_buffer_size"; - err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size); - if (err < 0) - goto error; - - cmd = "snd_pcm_hw_params_get_period_size"; - err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, - NULL); - if (err < 0) - goto error; - - ad->canPause = snd_pcm_hw_params_can_pause(hwparams); - ad->canResume = snd_pcm_hw_params_can_resume(hwparams); - - /* configure SW params */ - snd_pcm_sw_params_alloca(&swparams); - - cmd = "snd_pcm_sw_params_current"; - err = snd_pcm_sw_params_current(ad->pcmHandle, swparams); - if (err < 0) - goto error; - - cmd = "snd_pcm_sw_params_set_start_threshold"; - err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams, - alsa_buffer_size - - alsa_period_size); - if (err < 0) - goto error; - - cmd = "snd_pcm_sw_params_set_avail_min"; - err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams, - alsa_period_size); - if (err < 0) - goto error; - - cmd = "snd_pcm_sw_params_set_xfer_align"; - err = snd_pcm_sw_params_set_xfer_align(ad->pcmHandle, swparams, 1); - if (err < 0) - goto error; - - cmd = "snd_pcm_sw_params"; - err = snd_pcm_sw_params(ad->pcmHandle, swparams); - if (err < 0) - goto error; - - ad->sampleSize = (audioFormat->bits / 8) * audioFormat->channels; - - audioOutput->open = 1; - - DEBUG("alsa device \"%s\" will be playing %i bit, %i channel audio at " - "%i Hz\n", ad->device, (int)audioFormat->bits, - channels, sampleRate); - - return 0; - -error: - if (cmd) { - ERROR("Error opening alsa device \"%s\" (%s): %s\n", - ad->device, cmd, snd_strerror(-err)); - } else { - ERROR("Error opening alsa device \"%s\": %s\n", ad->device, - snd_strerror(-err)); - } -fail: - if (ad->pcmHandle) - snd_pcm_close(ad->pcmHandle); - ad->pcmHandle = NULL; - audioOutput->open = 0; - return -1; -} - -static int alsa_errorRecovery(AlsaData * ad, int err) -{ - if (err == -EPIPE) { - DEBUG("Underrun on alsa device \"%s\"\n", ad->device); - } else if (err == -ESTRPIPE) { - DEBUG("alsa device \"%s\" was suspended\n", ad->device); - } - - switch (snd_pcm_state(ad->pcmHandle)) { - case SND_PCM_STATE_PAUSED: - err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0); - break; - case SND_PCM_STATE_SUSPENDED: - err = ad->canResume ? - snd_pcm_resume(ad->pcmHandle) : - snd_pcm_prepare(ad->pcmHandle); - break; - case SND_PCM_STATE_SETUP: - case SND_PCM_STATE_XRUN: - err = snd_pcm_prepare(ad->pcmHandle); - break; - case SND_PCM_STATE_DISCONNECTED: - /* so alsa_closeDevice won't try to drain: */ - snd_pcm_close(ad->pcmHandle); - ad->pcmHandle = NULL; - break; - default: - /* unknown state, do nothing */ - break; - } - - return err; -} - -static void alsa_dropBufferedAudio(AudioOutput * audioOutput) -{ - AlsaData *ad = audioOutput->data; - - alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle)); -} - -static void alsa_closeDevice(AudioOutput * audioOutput) -{ - AlsaData *ad = audioOutput->data; - - if (ad->pcmHandle) { - snd_pcm_drain(ad->pcmHandle); - snd_pcm_close(ad->pcmHandle); - ad->pcmHandle = NULL; - } - - audioOutput->open = 0; -} - -static int alsa_playAudio(AudioOutput * audioOutput, char *playChunk, int size) -{ - AlsaData *ad = audioOutput->data; - int ret; - - size /= ad->sampleSize; - - while (size > 0) { - ret = ad->writei(ad->pcmHandle, playChunk, size); - - if (ret == -EAGAIN || ret == -EINTR) - continue; - - if (ret < 0) { - if (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; - } - continue; - } - - playChunk += ret * ad->sampleSize; - size -= ret; - } - - return 0; -} - -AudioOutputPlugin alsaPlugin = { - "alsa", - alsa_testDefault, - alsa_initDriver, - alsa_finishDriver, - alsa_openDevice, - alsa_playAudio, - alsa_dropBufferedAudio, - alsa_closeDevice, - NULL, /* sendMetadataFunc */ -}; - -#else /* HAVE ALSA */ - -DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin) -#endif /* HAVE_ALSA */ diff --git a/trunk/src/audioOutputs/audioOutput_ao.c b/trunk/src/audioOutputs/audioOutput_ao.c deleted file mode 100644 index a7f437ef4..000000000 --- a/trunk/src/audioOutputs/audioOutput_ao.c +++ /dev/null @@ -1,246 +0,0 @@ -/* the Music Player Daemon (MPD) - * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) - * 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" - -#ifdef HAVE_AO - -#include "../conf.h" -#include "../log.h" - -#include - -#include - -static int driverInitCount; - -typedef struct _AoData { - int writeSize; - int driverId; - ao_option *options; - ao_device *device; -} AoData; - -static AoData *newAoData(void) -{ - AoData *ret = xmalloc(sizeof(AoData)); - ret->device = NULL; - ret->options = NULL; - - return ret; -} - -static void audioOutputAo_error(void) -{ - if (errno == AO_ENOTLIVE) { - ERROR("not a live ao device\n"); - } else if (errno == AO_EOPENDEVICE) { - ERROR("not able to open audio device\n"); - } else if (errno == AO_EBADOPTION) { - ERROR("bad driver option\n"); - } -} - -static int audioOutputAo_initDriver(AudioOutput * audioOutput, - ConfigParam * param) -{ - ao_info *ai; - char *dup; - char *stk1; - char *stk2; - char *n1; - char *key; - char *value; - char *test; - AoData *ad = newAoData(); - BlockParam *blockParam; - - audioOutput->data = ad; - - if ((blockParam = getBlockParam(param, "write_size"))) { - ad->writeSize = strtol(blockParam->value, &test, 10); - if (*test != '\0') { - FATAL("\"%s\" is not a valid write size at line %i\n", - blockParam->value, blockParam->line); - } - } else - ad->writeSize = 1024; - - if (driverInitCount == 0) { - ao_initialize(); - } - driverInitCount++; - - blockParam = getBlockParam(param, "driver"); - - if (!blockParam || 0 == strcmp(blockParam->value, "default")) { - ad->driverId = ao_default_driver_id(); - } else if ((ad->driverId = ao_driver_id(blockParam->value)) < 0) { - FATAL("\"%s\" is not a valid ao driver at line %i\n", - blockParam->value, blockParam->line); - } - - if ((ai = ao_driver_info(ad->driverId)) == NULL) { - FATAL("problems getting driver info for device defined at line %i\n" - "you may not have permission to the audio device\n", param->line); - } - - DEBUG("using ao driver \"%s\" for \"%s\"\n", ai->short_name, - audioOutput->name); - - blockParam = getBlockParam(param, "options"); - - if (blockParam) { - dup = xstrdup(blockParam->value); - } else - dup = xstrdup(""); - - if (strlen(dup)) { - stk1 = NULL; - n1 = strtok_r(dup, ";", &stk1); - while (n1) { - stk2 = NULL; - key = strtok_r(n1, "=", &stk2); - if (!key) - FATAL("problems parsing options \"%s\"\n", n1); - /*found = 0; - for(i=0;ioption_count;i++) { - if(strcmp(ai->options[i],key)==0) { - found = 1; - break; - } - } - if(!found) { - FATAL("\"%s\" is not an option for " - "\"%s\" ao driver\n",key, - ai->short_name); - } */ - value = strtok_r(NULL, "", &stk2); - if (!value) - FATAL("problems parsing options \"%s\"\n", n1); - ao_append_option(&ad->options, key, value); - n1 = strtok_r(NULL, ";", &stk1); - } - } - free(dup); - - return 0; -} - -static void freeAoData(AoData * ad) -{ - ao_free_options(ad->options); - free(ad); -} - -static void audioOutputAo_finishDriver(AudioOutput * audioOutput) -{ - AoData *ad = (AoData *) audioOutput->data; - freeAoData(ad); - - driverInitCount--; - - if (driverInitCount == 0) - ao_shutdown(); -} - -static void audioOutputAo_dropBufferedAudio(AudioOutput * audioOutput) -{ - /* not supported by libao */ -} - -static void audioOutputAo_closeDevice(AudioOutput * audioOutput) -{ - AoData *ad = (AoData *) audioOutput->data; - - if (ad->device) { - ao_close(ad->device); - ad->device = NULL; - } - - audioOutput->open = 0; -} - -static int audioOutputAo_openDevice(AudioOutput * audioOutput) -{ - ao_sample_format format; - AoData *ad = (AoData *) audioOutput->data; - - if (ad->device) { - audioOutputAo_closeDevice(audioOutput); - } - - format.bits = audioOutput->outAudioFormat.bits; - format.rate = audioOutput->outAudioFormat.sampleRate; - format.byte_format = AO_FMT_NATIVE; - format.channels = audioOutput->outAudioFormat.channels; - - ad->device = ao_open_live(ad->driverId, &format, ad->options); - - if (ad->device == NULL) - return -1; - - audioOutput->open = 1; - - return 0; -} - -static int audioOutputAo_play(AudioOutput * audioOutput, char *playChunk, - int size) -{ - int send; - AoData *ad = (AoData *) audioOutput->data; - - if (ad->device == NULL) - return -1; - - while (size > 0) { - send = ad->writeSize > size ? size : ad->writeSize; - - if (ao_play(ad->device, playChunk, send) == 0) { - audioOutputAo_error(); - ERROR("closing audio device due to write error\n"); - audioOutputAo_closeDevice(audioOutput); - return -1; - } - - playChunk += send; - size -= send; - } - - return 0; -} - -AudioOutputPlugin aoPlugin = { - "ao", - NULL, - audioOutputAo_initDriver, - audioOutputAo_finishDriver, - audioOutputAo_openDevice, - audioOutputAo_play, - audioOutputAo_dropBufferedAudio, - audioOutputAo_closeDevice, - NULL, /* sendMetadataFunc */ -}; - -#else - -#include - -DISABLED_AUDIO_OUTPUT_PLUGIN(aoPlugin) -#endif diff --git a/trunk/src/audioOutputs/audioOutput_jack.c b/trunk/src/audioOutputs/audioOutput_jack.c deleted file mode 100644 index 1fdfaf4bb..000000000 --- a/trunk/src/audioOutputs/audioOutput_jack.c +++ /dev/null @@ -1,440 +0,0 @@ -/* jack plug in for the Music Player Daemon (MPD) - * (c)2006 by anarch(anarchsss@gmail.com) - * - * 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" - -#ifdef HAVE_JACK - -#include -#include - -#include "../conf.h" -#include "../log.h" - -#include -#include - -#include -#include -#include - -pthread_mutex_t play_audio_lock = PTHREAD_MUTEX_INITIALIZER; -pthread_cond_t play_audio = PTHREAD_COND_INITIALIZER; - -/*#include "dmalloc.h"*/ - -#define MIN(a, b) ((a) < (b) ? (a) : (b)) -/*#define SAMPLE_SIZE sizeof(jack_default_audio_sample_t);*/ - - -static char *name = "mpd"; -static char *output_ports[2]; -static int ringbuf_sz = 32768; -size_t sample_size = sizeof(jack_default_audio_sample_t); - -typedef struct _JackData { - jack_port_t *ports[2]; - jack_client_t *client; - jack_ringbuffer_t *ringbuffer[2]; - int bps; - int shutdown; -} JackData; - -/*JackData *jd = NULL;*/ - -static JackData *newJackData(void) -{ - JackData *ret; - ret = xcalloc(sizeof(JackData), 1); - - return ret; -} - -static void freeJackData(AudioOutput *audioOutput) -{ - JackData *jd = audioOutput->data; - if (jd) { - if (jd->ringbuffer[0]) - jack_ringbuffer_free(jd->ringbuffer[0]); - if (jd->ringbuffer[1]) - jack_ringbuffer_free(jd->ringbuffer[1]); - free(jd); - audioOutput->data = NULL; - } -} - -static void jack_finishDriver(AudioOutput *audioOutput) -{ - JackData *jd = audioOutput->data; - int i; - - if ( jd && jd->client ) { - jack_deactivate(jd->client); - jack_client_close(jd->client); - } - DEBUG("disconnect_jack (pid=%d)\n", getpid ()); - - if ( strcmp(name, "mpd") ) { - free(name); - name = "mpd"; - } - - for ( i = ARRAY_SIZE(output_ports); --i >= 0; ) { - if (!output_ports[i]) - continue; - free(output_ports[i]); - output_ports[i] = NULL; - } - - freeJackData(audioOutput); -} - -static int srate(jack_nframes_t rate, void *data) -{ - JackData *jd = (JackData *) ((AudioOutput*) data)->data; - AudioFormat *audioFormat = &(((AudioOutput*) data)->outAudioFormat); - - audioFormat->sampleRate = (int)jack_get_sample_rate(jd->client); - - return 0; -} - -static int process(jack_nframes_t nframes, void *arg) -{ - size_t i; - JackData *jd = (JackData *) arg; - jack_default_audio_sample_t *out[2]; - size_t avail_data, avail_frames; - - if ( nframes <= 0 ) - return 0; - - out[0] = jack_port_get_buffer(jd->ports[0], nframes); - out[1] = jack_port_get_buffer(jd->ports[1], nframes); - - while ( nframes ) { - avail_data = jack_ringbuffer_read_space(jd->ringbuffer[1]); - - if ( avail_data > 0 ) { - avail_frames = avail_data / sample_size; - - if (avail_frames > nframes) { - avail_frames = nframes; - avail_data = nframes*sample_size; - } - - jack_ringbuffer_read(jd->ringbuffer[0], (char *)out[0], - avail_data); - jack_ringbuffer_read(jd->ringbuffer[1], (char *)out[1], - avail_data); - - nframes -= avail_frames; - out[0] += avail_data; - out[1] += avail_data; - } else { - for (i = 0; i < nframes; i++) - out[0][i] = out[1][i] = 0.0; - nframes = 0; - } - - if (pthread_mutex_trylock (&play_audio_lock) == 0) { - pthread_cond_signal (&play_audio); - pthread_mutex_unlock (&play_audio_lock); - } - } - - - /*DEBUG("process (pid=%d)\n", getpid());*/ - return 0; -} - -static void shutdown_callback(void *arg) -{ - JackData *jd = (JackData *) arg; - jd->shutdown = 1; -} - -static void set_audioformat(AudioOutput *audioOutput) -{ - JackData *jd = audioOutput->data; - AudioFormat *audioFormat = &audioOutput->outAudioFormat; - - audioFormat->sampleRate = (int) jack_get_sample_rate(jd->client); - DEBUG("samplerate = %d\n", audioFormat->sampleRate); - audioFormat->channels = 2; - audioFormat->bits = 16; - jd->bps = audioFormat->channels - * sizeof(jack_default_audio_sample_t) - * audioFormat->sampleRate; -} - -static void error_callback(const char *msg) -{ - ERROR("jack: %s\n", msg); -} - -static int jack_initDriver(AudioOutput *audioOutput, ConfigParam *param) -{ - BlockParam *bp; - char *endptr; - int val; - char *cp = NULL; - - DEBUG("jack_initDriver (pid=%d)\n", getpid()); - if ( ! param ) return 0; - - if ( (bp = getBlockParam(param, "ports")) ) { - DEBUG("output_ports=%s\n", bp->value); - - if (!(cp = strchr(bp->value, ','))) - FATAL("expected comma and a second value for '%s' " - "at line %d: %s\n", - bp->name, bp->line, bp->value); - - *cp = '\0'; - output_ports[0] = xstrdup(bp->value); - *cp++ = ','; - - if (!*cp) - FATAL("expected a second value for '%s' at line %d: " - "%s\n", bp->name, bp->line, bp->value); - - output_ports[1] = xstrdup(cp); - - if (strchr(cp,',')) - FATAL("Only %d values are supported for '%s' " - "at line %d\n", (int)ARRAY_SIZE(output_ports), - bp->name, bp->line); - } - - if ( (bp = getBlockParam(param, "ringbuffer_size")) ) { - errno = 0; - val = strtol(bp->value, &endptr, 10); - - if ( errno == 0 && endptr != bp->value) { - ringbuf_sz = val < 32768 ? 32768 : val; - DEBUG("ringbuffer_size=%d\n", ringbuf_sz); - } else { - FATAL("%s is not a number; ringbuf_size=%d\n", - bp->value, ringbuf_sz); - } - } - - if ( (bp = getBlockParam(param, "name")) - && (strcmp(bp->value, "mpd") != 0) ) { - name = xstrdup(bp->value); - DEBUG("name=%s\n", name); - } - - return 0; -} - -static int jack_testDefault(void) -{ - return 0; -} - -static int connect_jack(AudioOutput *audioOutput) -{ - JackData *jd = audioOutput->data; - char **jports; - char *port_name; - - if ( (jd->client = jack_client_new(name)) == NULL ) { - ERROR("jack server not running?\n"); - freeJackData(audioOutput); - return -1; - } - - jack_set_error_function(error_callback); - jack_set_process_callback(jd->client, process, (void *)jd); - jack_set_sample_rate_callback(jd->client, (JackProcessCallback)srate, - (void *)audioOutput); - jack_on_shutdown(jd->client, shutdown_callback, (void *)jd); - - if ( jack_activate(jd->client) ) { - ERROR("cannot activate client"); - freeJackData(audioOutput); - return -1; - } - - jd->ports[0] = jack_port_register(jd->client, "left", - JACK_DEFAULT_AUDIO_TYPE, - JackPortIsOutput, 0); - if ( !jd->ports[0] ) { - ERROR("Cannot register left output port.\n"); - freeJackData(audioOutput); - return -1; - } - - jd->ports[1] = jack_port_register(jd->client, "right", - JACK_DEFAULT_AUDIO_TYPE, - JackPortIsOutput, 0); - if ( !jd->ports[1] ) { - ERROR("Cannot register right output port.\n"); - freeJackData(audioOutput); - return -1; - } - - /* hay que buscar que hay */ - if ( !output_ports[1] - && (jports = (char **)jack_get_ports(jd->client, NULL, NULL, - JackPortIsPhysical| - JackPortIsInput)) ) { - output_ports[0] = jports[0]; - output_ports[1] = jports[1] ? jports[1] : jports[0]; - DEBUG("output_ports: %s %s\n", output_ports[0], output_ports[1]); - free(jports); - } - - if ( output_ports[1] ) { - jd->ringbuffer[0] = jack_ringbuffer_create(ringbuf_sz); - jd->ringbuffer[1] = jack_ringbuffer_create(ringbuf_sz); - memset(jd->ringbuffer[0]->buf, 0, jd->ringbuffer[0]->size); - memset(jd->ringbuffer[1]->buf, 0, jd->ringbuffer[1]->size); - - port_name = xmalloc(sizeof(char)*(7+strlen(name))); - - sprintf(port_name, "%s:left", name); - if ( (jack_connect(jd->client, port_name, - output_ports[0])) != 0 ) { - ERROR("%s is not a valid Jack Client / Port ", - output_ports[0]); - freeJackData(audioOutput); - free(port_name); - return -1; - } - sprintf(port_name, "%s:right", name); - if ( (jack_connect(jd->client, port_name, - output_ports[1])) != 0 ) { - ERROR("%s is not a valid Jack Client / Port ", - output_ports[1]); - freeJackData(audioOutput); - free(port_name); - return -1; - } - free(port_name); - } - - DEBUG("connect_jack (pid=%d)\n", getpid()); - return 1; -} - -static int jack_openDevice(AudioOutput *audioOutput) -{ - JackData *jd = audioOutput->data; - - if ( !jd ) { - DEBUG("connect!\n"); - jd = newJackData(); - audioOutput->data = jd; - - if (connect_jack(audioOutput) < 0) { - freeJackData(audioOutput); - audioOutput->open = 0; - return -1; - } - } - - set_audioformat(audioOutput); - audioOutput->open = 1; - - DEBUG("jack_openDevice (pid=%d)!\n", getpid ()); - return 0; -} - - -static void jack_closeDevice(AudioOutput * audioOutput) -{ - /*jack_finishDriver(audioOutput);*/ - audioOutput->open = 0; - DEBUG("jack_closeDevice (pid=%d)\n", getpid()); -} - -static void jack_dropBufferedAudio (AudioOutput * audioOutput) -{ -} - -static int jack_playAudio(AudioOutput * audioOutput, char *buff, int size) -{ - JackData *jd = audioOutput->data; - size_t space; - int i; - short *buffer = (short *) buff; - jack_default_audio_sample_t sample; - size_t samples = size/4; - - /*DEBUG("jack_playAudio: (pid=%d)!\n", getpid());*/ - - if ( jd->shutdown ) { - ERROR("Refusing to play, because there is no client thread.\n"); - freeJackData(audioOutput); - audioOutput->open = 0; - return 0; - } - - while ( samples && !jd->shutdown ) { - - if ( (space = jack_ringbuffer_write_space(jd->ringbuffer[0])) - >= samples*sample_size ) { - - /*space = MIN(space, samples*sample_size);*/ - /*space = samples*sample_size;*/ - - /*for(i=0; iringbuffer[0], (void*)&sample, - sample_size); - - sample = (jack_default_audio_sample_t) *(buffer++)/32768.0; - - jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample, - sample_size); - - /*samples--;*/ - } - samples=0; - - } else { - pthread_mutex_lock(&play_audio_lock); - pthread_cond_wait(&play_audio, &play_audio_lock); - pthread_mutex_unlock(&play_audio_lock); - } - - } - return 0; -} - -AudioOutputPlugin jackPlugin = { - "jack", - jack_testDefault, - jack_initDriver, - jack_finishDriver, - jack_openDevice, - jack_playAudio, - jack_dropBufferedAudio, - jack_closeDevice, - NULL, /* sendMetadataFunc */ -}; - -#else /* HAVE JACK */ - -DISABLED_AUDIO_OUTPUT_PLUGIN(jackPlugin) - -#endif /* HAVE_JACK */ diff --git a/trunk/src/audioOutputs/audioOutput_mvp.c b/trunk/src/audioOutputs/audioOutput_mvp.c deleted file mode 100644 index ea365c657..000000000 --- a/trunk/src/audioOutputs/audioOutput_mvp.c +++ /dev/null @@ -1,284 +0,0 @@ -/* the Music Player Daemon (MPD) - * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) - * This project's homepage is: http://www.musicpd.org - * - * Media MVP audio output based on code from MVPMC project: - * http://mvpmc.sourceforge.net/ - * - * 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 - -#ifdef HAVE_MVP - -#include "../conf.h" -#include "../log.h" - -#include - -#include -#include -#include -#include -#include -#include - -typedef struct { - unsigned long dsp_status; - unsigned long stream_decode_type; - unsigned long sample_rate; - unsigned long bit_rate; - unsigned long raw[64 / sizeof(unsigned long)]; -} aud_status_t; - -#define MVP_SET_AUD_STOP _IOW('a',1,int) -#define MVP_SET_AUD_PLAY _IOW('a',2,int) -#define MVP_SET_AUD_PAUSE _IOW('a',3,int) -#define MVP_SET_AUD_UNPAUSE _IOW('a',4,int) -#define MVP_SET_AUD_SRC _IOW('a',5,int) -#define MVP_SET_AUD_MUTE _IOW('a',6,int) -#define MVP_SET_AUD_BYPASS _IOW('a',8,int) -#define MVP_SET_AUD_CHANNEL _IOW('a',9,int) -#define MVP_GET_AUD_STATUS _IOR('a',10,aud_status_t) -#define MVP_SET_AUD_VOLUME _IOW('a',13,int) -#define MVP_GET_AUD_VOLUME _IOR('a',14,int) -#define MVP_SET_AUD_STREAMTYPE _IOW('a',15,int) -#define MVP_SET_AUD_FORMAT _IOW('a',16,int) -#define MVP_GET_AUD_SYNC _IOR('a',21,pts_sync_data_t*) -#define MVP_SET_AUD_STC _IOW('a',22,long long int *) -#define MVP_SET_AUD_SYNC _IOW('a',23,int) -#define MVP_SET_AUD_END_STREAM _IOW('a',25,int) -#define MVP_SET_AUD_RESET _IOW('a',26,int) -#define MVP_SET_AUD_DAC_CLK _IOW('a',27,int) -#define MVP_GET_AUD_REGS _IOW('a',28,aud_ctl_regs_t*) - -typedef struct _MvpData { - int fd; -} MvpData; - -static int pcmfrequencies[][3] = { - {9, 8000, 32000}, - {10, 11025, 44100}, - {11, 12000, 48000}, - {1, 16000, 32000}, - {2, 22050, 44100}, - {3, 24000, 48000}, - {5, 32000, 32000}, - {0, 44100, 44100}, - {7, 48000, 48000}, - {13, 64000, 32000}, - {14, 88200, 44100}, - {15, 96000, 48000} -}; - -static int numfrequencies = sizeof(pcmfrequencies) / 12; - -static int mvp_testDefault(void) -{ - int fd; - - fd = open("/dev/adec_pcm", O_WRONLY); - - if (fd) { - close(fd); - return 0; - } - - WARNING("Error opening PCM device \"/dev/adec_pcm\": %s\n", - strerror(errno)); - - return -1; -} - -static int mvp_initDriver(AudioOutput * audioOutput, ConfigParam * param) -{ - MvpData *md = xmalloc(sizeof(MvpData)); - md->fd = -1; - audioOutput->data = md; - - return 0; -} - -static void mvp_finishDriver(AudioOutput * audioOutput) -{ - MvpData *md = audioOutput->data; - free(md); -} - -static int mvp_setPcmParams(MvpData * md, unsigned long rate, int channels, - int big_endian, int bits) -{ - int iloop; - int mix[5]; - - if (channels == 1) - mix[0] = 1; - else if (channels == 2) - mix[0] = 0; - else - return -1; - - /* 0,1=24bit(24) , 2,3=16bit */ - if (bits == 16) - mix[1] = 2; - else if (bits == 24) - mix[1] = 0; - else - return -1; - - mix[3] = 0; /* stream type? */ - - if (big_endian == 1) - mix[4] = 1; - else if (big_endian == 0) - mix[4] = 0; - else - return -1; - - /* - * if there is an exact match for the frequency, use it. - */ - for (iloop = 0; iloop < numfrequencies; iloop++) { - if (rate == pcmfrequencies[iloop][1]) { - mix[2] = pcmfrequencies[iloop][0]; - break; - } - } - - if (iloop >= numfrequencies) { - ERROR("Can not find suitable output frequency for %ld\n", rate); - return -1; - } - - if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) { - ERROR("Can not set audio format\n"); - return -1; - } - - if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) { - ERROR("Can not set audio sync\n"); - return -1; - } - - if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) { - ERROR("Can not set audio play mode\n"); - return -1; - } - - return 0; -} - -static int mvp_openDevice(AudioOutput * audioOutput) -{ - long long int stc = 0; - MvpData *md = audioOutput->data; - AudioFormat *audioFormat = &audioOutput->outAudioFormat; - int mix[5] = { 0, 2, 7, 1, 0 }; - - if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) { - ERROR("Error opening /dev/adec_pcm: %s\n", strerror(errno)); - return -1; - } - if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) { - ERROR("Error setting audio source: %s\n", strerror(errno)); - return -1; - } - if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) { - ERROR("Error setting audio streamtype: %s\n", strerror(errno)); - return -1; - } - if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) { - ERROR("Error setting audio format: %s\n", strerror(errno)); - return -1; - } - ioctl(md->fd, MVP_SET_AUD_STC, &stc); - if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) { - ERROR("Error setting audio streamtype: %s\n", strerror(errno)); - return -1; - } -#ifdef WORDS_BIGENDIAN - mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 0, - audioFormat->bits); -#else - mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 1, - audioFormat->bits); -#endif - audioOutput->open = 1; - return 0; -} - -static void mvp_closeDevice(AudioOutput * audioOutput) -{ - MvpData *md = audioOutput->data; - if (md->fd >= 0) - close(md->fd); - md->fd = -1; - audioOutput->open = 0; -} - -static void mvp_dropBufferedAudio(AudioOutput * audioOutput) -{ - MvpData *md = audioOutput->data; - if (md->fd >= 0) { - ioctl(md->fd, MVP_SET_AUD_RESET, 0x11); - close(md->fd); - md->fd = -1; - audioOutput->open = 0; - } -} - -static int mvp_playAudio(AudioOutput * audioOutput, char *playChunk, int size) -{ - MvpData *md = audioOutput->data; - int ret; - - /* reopen the device since it was closed by dropBufferedAudio */ - if (md->fd < 0) - mvp_openDevice(audioOutput); - - while (size > 0) { - ret = write(md->fd, playChunk, size); - if (ret < 0) { - if (errno == EINTR) - continue; - ERROR("closing mvp PCM device due to write error: " - "%s\n", strerror(errno)); - mvp_closeDevice(audioOutput); - return -1; - } - playChunk += ret; - size -= ret; - } - return 0; -} - -AudioOutputPlugin mvpPlugin = { - "mvp", - mvp_testDefault, - mvp_initDriver, - mvp_finishDriver, - mvp_openDevice, - mvp_playAudio, - mvp_dropBufferedAudio, - mvp_closeDevice, - NULL, /* sendMetadataFunc */ -}; - -#else /* HAVE_MVP */ - -DISABLED_AUDIO_OUTPUT_PLUGIN(mvpPlugin) -#endif /* HAVE_MVP */ diff --git a/trunk/src/audioOutputs/audioOutput_oss.c b/trunk/src/audioOutputs/audioOutput_oss.c deleted file mode 100644 index 01293cbd1..000000000 --- a/trunk/src/audioOutputs/audioOutput_oss.c +++ /dev/null @@ -1,575 +0,0 @@ -/* the Music Player Daemon (MPD) - * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) - * This project's homepage is: http://www.musicpd.org - * - * OSS audio output (c) 2004, 2005, 2006, 2007 by Eric Wong - * and Warren Dukes - * - * 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 - -#ifdef HAVE_OSS - -#include "../conf.h" -#include "../log.h" - -#include - -#include -#include -#include -#include -#include -#include - -#if defined(__OpenBSD__) || defined(__NetBSD__) -# include -#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */ -# include -#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */ - -#ifdef WORDS_BIGENDIAN -# define AFMT_S16_MPD AFMT_S16_BE -#else -# define AFMT_S16_MPD AFMT_S16_LE -#endif /* WORDS_BIGENDIAN */ - -typedef struct _OssData { - int fd; - const char *device; - int channels; - int sampleRate; - int bitFormat; - int bits; - int *supported[3]; - int numSupported[3]; - int *unsupported[3]; - int numUnsupported[3]; -} OssData; - -#define OSS_SUPPORTED 1 -#define OSS_UNSUPPORTED 0 -#define OSS_UNKNOWN -1 - -#define OSS_RATE 0 -#define OSS_CHANNELS 1 -#define OSS_BITS 2 - -static int getIndexForParam(int param) -{ - int index = 0; - - switch (param) { - case SNDCTL_DSP_SPEED: - index = OSS_RATE; - break; - case SNDCTL_DSP_CHANNELS: - index = OSS_CHANNELS; - break; - case SNDCTL_DSP_SAMPLESIZE: - index = OSS_BITS; - break; - } - - return index; -} - -static int findSupportedParam(OssData * od, int param, int val) -{ - int i; - int index = getIndexForParam(param); - - for (i = 0; i < od->numSupported[index]; i++) { - if (od->supported[index][i] == val) - return 1; - } - - return 0; -} - -static int canConvert(int index, int val) -{ - switch (index) { - case OSS_BITS: - if (val != 16) - return 0; - break; - case OSS_CHANNELS: - if (val != 2) - return 0; - break; - } - - return 1; -} - -static int getSupportedParam(OssData * od, int param, int val) -{ - int i; - int index = getIndexForParam(param); - int ret = -1; - int least = val; - int diff; - - for (i = 0; i < od->numSupported[index]; i++) { - diff = od->supported[index][i] - val; - if (diff < 0) - diff = -diff; - if (diff < least) { - if (!canConvert(index, od->supported[index][i])) { - continue; - } - least = diff; - ret = od->supported[index][i]; - } - } - - return ret; -} - -static int findUnsupportedParam(OssData * od, int param, int val) -{ - int i; - int index = getIndexForParam(param); - - for (i = 0; i < od->numUnsupported[index]; i++) { - if (od->unsupported[index][i] == val) - return 1; - } - - return 0; -} - -static void addSupportedParam(OssData * od, int param, int val) -{ - int index = getIndexForParam(param); - - od->numSupported[index]++; - od->supported[index] = xrealloc(od->supported[index], - od->numSupported[index] * sizeof(int)); - od->supported[index][od->numSupported[index] - 1] = val; -} - -static void addUnsupportedParam(OssData * od, int param, int val) -{ - int index = getIndexForParam(param); - - od->numUnsupported[index]++; - od->unsupported[index] = xrealloc(od->unsupported[index], - od->numUnsupported[index] * - sizeof(int)); - od->unsupported[index][od->numUnsupported[index] - 1] = val; -} - -static void removeSupportedParam(OssData * od, int param, int val) -{ - int i = 0; - int j = 0; - int index = getIndexForParam(param); - - for (i = 0; i < od->numSupported[index] - 1; i++) { - if (od->supported[index][i] == val) - j = 1; - od->supported[index][i] = od->supported[index][i + j]; - } - - od->numSupported[index]--; - od->supported[index] = xrealloc(od->supported[index], - od->numSupported[index] * sizeof(int)); -} - -static void removeUnsupportedParam(OssData * od, int param, int val) -{ - int i = 0; - int j = 0; - int index = getIndexForParam(param); - - for (i = 0; i < od->numUnsupported[index] - 1; i++) { - if (od->unsupported[index][i] == val) - j = 1; - od->unsupported[index][i] = od->unsupported[index][i + j]; - } - - od->numUnsupported[index]--; - od->unsupported[index] = xrealloc(od->unsupported[index], - od->numUnsupported[index] * - sizeof(int)); -} - -static int isSupportedParam(OssData * od, int param, int val) -{ - if (findSupportedParam(od, param, val)) - return OSS_SUPPORTED; - if (findUnsupportedParam(od, param, val)) - return OSS_UNSUPPORTED; - return OSS_UNKNOWN; -} - -static void supportParam(OssData * od, int param, int val) -{ - int supported = isSupportedParam(od, param, val); - - if (supported == OSS_SUPPORTED) - return; - - if (supported == OSS_UNSUPPORTED) { - removeUnsupportedParam(od, param, val); - } - - addSupportedParam(od, param, val); -} - -static void unsupportParam(OssData * od, int param, int val) -{ - int supported = isSupportedParam(od, param, val); - - if (supported == OSS_UNSUPPORTED) - return; - - if (supported == OSS_SUPPORTED) { - removeSupportedParam(od, param, val); - } - - addUnsupportedParam(od, param, val); -} - -static OssData *newOssData(void) -{ - OssData *ret = xmalloc(sizeof(OssData)); - - ret->device = NULL; - ret->fd = -1; - - ret->supported[OSS_RATE] = NULL; - ret->supported[OSS_CHANNELS] = NULL; - ret->supported[OSS_BITS] = NULL; - ret->unsupported[OSS_RATE] = NULL; - ret->unsupported[OSS_CHANNELS] = NULL; - ret->unsupported[OSS_BITS] = NULL; - - ret->numSupported[OSS_RATE] = 0; - ret->numSupported[OSS_CHANNELS] = 0; - ret->numSupported[OSS_BITS] = 0; - ret->numUnsupported[OSS_RATE] = 0; - ret->numUnsupported[OSS_CHANNELS] = 0; - ret->numUnsupported[OSS_BITS] = 0; - - supportParam(ret, SNDCTL_DSP_SPEED, 48000); - supportParam(ret, SNDCTL_DSP_SPEED, 44100); - supportParam(ret, SNDCTL_DSP_CHANNELS, 2); - supportParam(ret, SNDCTL_DSP_SAMPLESIZE, 16); - - return ret; -} - -static void freeOssData(OssData * od) -{ - if (od->supported[OSS_RATE]) - free(od->supported[OSS_RATE]); - if (od->supported[OSS_CHANNELS]) - free(od->supported[OSS_CHANNELS]); - if (od->supported[OSS_BITS]) - free(od->supported[OSS_BITS]); - if (od->unsupported[OSS_RATE]) - free(od->unsupported[OSS_RATE]); - if (od->unsupported[OSS_CHANNELS]) - free(od->unsupported[OSS_CHANNELS]); - if (od->unsupported[OSS_BITS]) - free(od->unsupported[OSS_BITS]); - - free(od); -} - -#define OSS_STAT_NO_ERROR 0 -#define OSS_STAT_NOT_CHAR_DEV -1 -#define OSS_STAT_NO_PERMS -2 -#define OSS_STAT_DOESN_T_EXIST -3 -#define OSS_STAT_OTHER -4 - -static int oss_statDevice(const char *device, int *stErrno) -{ - struct stat st; - - if (0 == stat(device, &st)) { - if (!S_ISCHR(st.st_mode)) { - return OSS_STAT_NOT_CHAR_DEV; - } - } else { - *stErrno = errno; - - switch (errno) { - case ENOENT: - case ENOTDIR: - return OSS_STAT_DOESN_T_EXIST; - case EACCES: - return OSS_STAT_NO_PERMS; - default: - return OSS_STAT_OTHER; - } - } - - return 0; -} - -static const char *default_devices[] = { "/dev/sound/dsp", "/dev/dsp" }; - -static int oss_testDefault(void) -{ - int fd, i; - - for (i = ARRAY_SIZE(default_devices); --i >= 0; ) { - if ((fd = open(default_devices[i], O_WRONLY)) >= 0) { - xclose(fd); - return 0; - } - WARNING("Error opening OSS device \"%s\": %s\n", - default_devices[i], strerror(errno)); - } - - return -1; -} - -static int oss_open_default(AudioOutput *ao, ConfigParam *param, OssData *od) -{ - int i; - int err[ARRAY_SIZE(default_devices)]; - int ret[ARRAY_SIZE(default_devices)]; - - for (i = ARRAY_SIZE(default_devices); --i >= 0; ) { - ret[i] = oss_statDevice(default_devices[i], &err[i]); - if (ret[i] == 0) { - od->device = default_devices[i]; - return 0; - } - } - - if (param) - ERROR("error trying to open specified OSS device" - " at line %i\n", param->line); - else - ERROR("error trying to open default OSS device\n"); - - for (i = ARRAY_SIZE(default_devices); --i >= 0; ) { - const char *dev = default_devices[i]; - switch(ret[i]) { - case OSS_STAT_DOESN_T_EXIST: - ERROR("%s not found\n", dev); - break; - case OSS_STAT_NOT_CHAR_DEV: - ERROR("%s is not a character device\n", dev); - break; - case OSS_STAT_NO_PERMS: - ERROR("%s: permission denied\n", dev); - break; - default: - ERROR("Error accessing %s: %s", dev, strerror(err[i])); - } - } - exit(EXIT_FAILURE); - return 0; /* some compilers can be dumb... */ -} - -static int oss_initDriver(AudioOutput * audioOutput, ConfigParam * param) -{ - OssData *od = newOssData(); - audioOutput->data = od; - if (param) { - BlockParam *bp = getBlockParam(param, "device"); - if (bp) { - od->device = bp->value; - return 0; - } - } - return oss_open_default(audioOutput, param, od); -} - -static void oss_finishDriver(AudioOutput * audioOutput) -{ - OssData *od = audioOutput->data; - - freeOssData(od); -} - -static int setParam(OssData * od, int param, int *value) -{ - int val = *value; - int copy; - int supported = isSupportedParam(od, param, val); - - do { - if (supported == OSS_UNSUPPORTED) { - val = getSupportedParam(od, param, val); - if (copy < 0) - return -1; - } - copy = val; - if (ioctl(od->fd, param, ©)) { - unsupportParam(od, param, val); - supported = OSS_UNSUPPORTED; - } else { - if (supported == OSS_UNKNOWN) { - supportParam(od, param, val); - supported = OSS_SUPPORTED; - } - val = copy; - } - } while (supported == OSS_UNSUPPORTED); - - *value = val; - - return 0; -} - -static void oss_close(OssData * od) -{ - if (od->fd >= 0) - while (close(od->fd) && errno == EINTR) ; - od->fd = -1; -} - -static int oss_open(AudioOutput * audioOutput) -{ - int tmp; - OssData *od = audioOutput->data; - - if ((od->fd = open(od->device, O_WRONLY)) < 0) { - ERROR("Error opening OSS device \"%s\": %s\n", od->device, - strerror(errno)); - goto fail; - } - - if (setParam(od, SNDCTL_DSP_CHANNELS, &od->channels)) { - ERROR("OSS device \"%s\" does not support %i channels: %s\n", - od->device, od->channels, strerror(errno)); - goto fail; - } - - if (setParam(od, SNDCTL_DSP_SPEED, &od->sampleRate)) { - ERROR("OSS device \"%s\" does not support %i Hz audio: %s\n", - od->device, od->sampleRate, strerror(errno)); - goto fail; - } - - switch (od->bits) { - case 8: - tmp = AFMT_S8; - break; - case 16: - tmp = AFMT_S16_MPD; - } - - if (setParam(od, SNDCTL_DSP_SAMPLESIZE, &tmp)) { - ERROR("OSS device \"%s\" does not support %i bit audio: %s\n", - od->device, tmp, strerror(errno)); - goto fail; - } - - audioOutput->open = 1; - - return 0; - -fail: - oss_close(od); - audioOutput->open = 0; - return -1; -} - -static int oss_openDevice(AudioOutput * audioOutput) -{ - int ret = -1; - OssData *od = audioOutput->data; - AudioFormat *audioFormat = &audioOutput->outAudioFormat; - - od->channels = audioFormat->channels; - od->sampleRate = audioFormat->sampleRate; - od->bits = audioFormat->bits; - - if ((ret = oss_open(audioOutput)) < 0) - return ret; - - audioFormat->channels = od->channels; - audioFormat->sampleRate = od->sampleRate; - audioFormat->bits = od->bits; - - DEBUG("oss device \"%s\" will be playing %i bit %i channel audio at " - "%i Hz\n", od->device, od->bits, od->channels, od->sampleRate); - - return ret; -} - -static void oss_closeDevice(AudioOutput * audioOutput) -{ - OssData *od = audioOutput->data; - - oss_close(od); - - audioOutput->open = 0; -} - -static void oss_dropBufferedAudio(AudioOutput * audioOutput) -{ - OssData *od = audioOutput->data; - - if (od->fd >= 0) { - ioctl(od->fd, SNDCTL_DSP_RESET, 0); - oss_close(od); - } -} - -static int oss_playAudio(AudioOutput * audioOutput, char *playChunk, int size) -{ - OssData *od = audioOutput->data; - int ret; - - /* reopen the device since it was closed by dropBufferedAudio */ - if (od->fd < 0 && oss_open(audioOutput) < 0) - return -1; - - while (size > 0) { - ret = write(od->fd, playChunk, size); - if (ret < 0) { - if (errno == EINTR) - continue; - ERROR("closing oss device \"%s\" due to write error: " - "%s\n", od->device, strerror(errno)); - oss_closeDevice(audioOutput); - return -1; - } - playChunk += ret; - size -= ret; - } - - return 0; -} - -AudioOutputPlugin ossPlugin = { - "oss", - oss_testDefault, - oss_initDriver, - oss_finishDriver, - oss_openDevice, - oss_playAudio, - oss_dropBufferedAudio, - oss_closeDevice, - NULL, /* sendMetadataFunc */ -}; - -#else /* HAVE OSS */ - -DISABLED_AUDIO_OUTPUT_PLUGIN(ossPlugin) -#endif /* HAVE_OSS */ diff --git a/trunk/src/audioOutputs/audioOutput_osx.c b/trunk/src/audioOutputs/audioOutput_osx.c deleted file mode 100644 index 1caebade5..000000000 --- a/trunk/src/audioOutputs/audioOutput_osx.c +++ /dev/null @@ -1,374 +0,0 @@ -/* the Music Player Daemon (MPD) - * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) - * 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" - -#ifdef HAVE_OSX - -#include -#include -#include - -#include "../log.h" - -typedef struct _OsxData { - AudioUnit au; - pthread_mutex_t mutex; - pthread_cond_t condition; - char *buffer; - int bufferSize; - int pos; - int len; - int started; -} OsxData; - -static OsxData *newOsxData() -{ - OsxData *ret = xmalloc(sizeof(OsxData)); - - pthread_mutex_init(&ret->mutex, NULL); - pthread_cond_init(&ret->condition, NULL); - - ret->pos = 0; - ret->len = 0; - ret->started = 0; - ret->buffer = NULL; - ret->bufferSize = 0; - - return ret; -} - -static int osx_testDefault() -{ - /*AudioUnit au; - ComponentDescription desc; - Component comp; - - desc.componentType = kAudioUnitType_Output; - desc.componentSubType = kAudioUnitSubType_Output; - desc.componentManufacturer = kAudioUnitManufacturer_Apple; - desc.componentFlags = 0; - desc.componentFlagsMask = 0; - - comp = FindNextComponent(NULL, &desc); - if(!comp) { - ERROR("Unable to open default OS X defice\n"); - return -1; - } - - if(OpenAComponent(comp, &au) != noErr) { - ERROR("Unable to open default OS X defice\n"); - return -1; - } - - CloseComponent(au); */ - - return 0; -} - -static int osx_initDriver(AudioOutput * audioOutput, ConfigParam * param) -{ - OsxData *od = newOsxData(); - - audioOutput->data = od; - - return 0; -} - -static void freeOsxData(OsxData * od) -{ - if (od->buffer) - free(od->buffer); - pthread_mutex_destroy(&od->mutex); - pthread_cond_destroy(&od->condition); - free(od); -} - -static void osx_finishDriver(AudioOutput * audioOutput) -{ - OsxData *od = (OsxData *) audioOutput->data; - freeOsxData(od); -} - -static void osx_dropBufferedAudio(AudioOutput * audioOutput) -{ - OsxData *od = (OsxData *) audioOutput->data; - - pthread_mutex_lock(&od->mutex); - od->len = 0; - pthread_mutex_unlock(&od->mutex); -} - -static void osx_closeDevice(AudioOutput * audioOutput) -{ - OsxData *od = (OsxData *) audioOutput->data; - - pthread_mutex_lock(&od->mutex); - while (od->len) { - pthread_cond_wait(&od->condition, &od->mutex); - } - pthread_mutex_unlock(&od->mutex); - - if (od->started) { - AudioOutputUnitStop(od->au); - od->started = 0; - } - - CloseComponent(od->au); - AudioUnitUninitialize(od->au); - - audioOutput->open = 0; -} - -static OSStatus osx_render(void *vdata, - AudioUnitRenderActionFlags * ioActionFlags, - const AudioTimeStamp * inTimeStamp, - UInt32 inBusNumber, UInt32 inNumberFrames, - AudioBufferList * bufferList) -{ - OsxData *od = (OsxData *) vdata; - AudioBuffer *buffer = &bufferList->mBuffers[0]; - int bufferSize = buffer->mDataByteSize; - int bytesToCopy; - int curpos = 0; - - /*DEBUG("osx_render: enter : %i\n", (int)bufferList->mNumberBuffers); - DEBUG("osx_render: ioActionFlags: %p\n", ioActionFlags); - if(ioActionFlags) { - if(*ioActionFlags & kAudioUnitRenderAction_PreRender) { - DEBUG("prerender\n"); - } - if(*ioActionFlags & kAudioUnitRenderAction_PostRender) { - DEBUG("post render\n"); - } - if(*ioActionFlags & kAudioUnitRenderAction_OutputIsSilence) { - DEBUG("post render\n"); - } - if(*ioActionFlags & kAudioOfflineUnitRenderAction_Preflight) { - DEBUG("prefilight\n"); - } - if(*ioActionFlags & kAudioOfflineUnitRenderAction_Render) { - DEBUG("render\n"); - } - if(*ioActionFlags & kAudioOfflineUnitRenderAction_Complete) { - DEBUG("complete\n"); - } - } */ - - /* while(bufferSize) { - DEBUG("osx_render: lock\n"); */ - pthread_mutex_lock(&od->mutex); - /* - DEBUG("%i:%i\n", bufferSize, od->len); - while(od->go && od->len < bufferSize && - od->len < od->bufferSize) - { - DEBUG("osx_render: wait\n"); - pthread_cond_wait(&od->condition, &od->mutex); - } - */ - - bytesToCopy = od->len < bufferSize ? od->len : bufferSize; - bufferSize = bytesToCopy; - od->len -= bytesToCopy; - - if (od->pos + bytesToCopy > od->bufferSize) { - int bytes = od->bufferSize - od->pos; - memcpy(buffer->mData + curpos, od->buffer + od->pos, bytes); - od->pos = 0; - curpos += bytes; - bytesToCopy -= bytes; - } - - memcpy(buffer->mData + curpos, od->buffer + od->pos, bytesToCopy); - od->pos += bytesToCopy; - curpos += bytesToCopy; - - if (od->pos >= od->bufferSize) - od->pos = 0; - /* DEBUG("osx_render: unlock\n"); */ - pthread_mutex_unlock(&od->mutex); - pthread_cond_signal(&od->condition); - /* } */ - - buffer->mDataByteSize = bufferSize; - - if (!bufferSize) { - my_usleep(1000); - } - - /* DEBUG("osx_render: leave\n"); */ - return 0; -} - -static int osx_openDevice(AudioOutput * audioOutput) -{ - OsxData *od = (OsxData *) audioOutput->data; - ComponentDescription desc; - Component comp; - AURenderCallbackStruct callback; - AudioFormat *audioFormat = &audioOutput->outAudioFormat; - AudioStreamBasicDescription streamDesc; - - desc.componentType = kAudioUnitType_Output; - desc.componentSubType = kAudioUnitSubType_DefaultOutput; - desc.componentManufacturer = kAudioUnitManufacturer_Apple; - desc.componentFlags = 0; - desc.componentFlagsMask = 0; - - comp = FindNextComponent(NULL, &desc); - if (comp == 0) { - ERROR("Error finding OS X component\n"); - return -1; - } - - if (OpenAComponent(comp, &od->au) != noErr) { - ERROR("Unable to open OS X component\n"); - return -1; - } - - if (AudioUnitInitialize(od->au) != 0) { - CloseComponent(od->au); - ERROR("Unable to initialize OS X audio unit\n"); - return -1; - } - - callback.inputProc = osx_render; - callback.inputProcRefCon = od; - - if (AudioUnitSetProperty(od->au, kAudioUnitProperty_SetRenderCallback, - kAudioUnitScope_Input, 0, - &callback, sizeof(callback)) != 0) { - AudioUnitUninitialize(od->au); - CloseComponent(od->au); - ERROR("unable to set callback for OS X audio unit\n"); - return -1; - } - - streamDesc.mSampleRate = audioFormat->sampleRate; - streamDesc.mFormatID = kAudioFormatLinearPCM; - streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger; -#ifdef WORDS_BIGENDIAN - streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; -#endif - - streamDesc.mBytesPerPacket = - audioFormat->channels * audioFormat->bits / 8; - streamDesc.mFramesPerPacket = 1; - streamDesc.mBytesPerFrame = streamDesc.mBytesPerPacket; - streamDesc.mChannelsPerFrame = audioFormat->channels; - streamDesc.mBitsPerChannel = audioFormat->bits; - - if (AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat, - kAudioUnitScope_Input, 0, - &streamDesc, sizeof(streamDesc)) != 0) { - AudioUnitUninitialize(od->au); - CloseComponent(od->au); - ERROR("Unable to set format on OS X device\n"); - return -1; - } - - /* create a buffer of 1s */ - od->bufferSize = (audioFormat->sampleRate) * - (audioFormat->bits >> 3) * (audioFormat->channels); - od->buffer = xrealloc(od->buffer, od->bufferSize); - - od->pos = 0; - od->len = 0; - - audioOutput->open = 1; - - return 0; -} - -static int osx_play(AudioOutput * audioOutput, char *playChunk, int size) -{ - OsxData *od = (OsxData *) audioOutput->data; - int bytesToCopy; - int curpos; - - /* DEBUG("osx_play: enter\n"); */ - - if (!od->started) { - int err; - od->started = 1; - err = AudioOutputUnitStart(od->au); - if (err) { - ERROR("unable to start audio output: %i\n", err); - return -1; - } - } - - pthread_mutex_lock(&od->mutex); - - while (size) { - /* DEBUG("osx_play: lock\n"); */ - curpos = od->pos + od->len; - if (curpos >= od->bufferSize) - curpos -= od->bufferSize; - - bytesToCopy = od->bufferSize < size ? od->bufferSize : size; - - while (od->len > od->bufferSize - bytesToCopy) { - /* DEBUG("osx_play: wait\n"); */ - pthread_cond_wait(&od->condition, &od->mutex); - } - - bytesToCopy = od->bufferSize - od->len; - bytesToCopy = bytesToCopy < size ? bytesToCopy : size; - size -= bytesToCopy; - od->len += bytesToCopy; - - if (curpos + bytesToCopy > od->bufferSize) { - int bytes = od->bufferSize - curpos; - memcpy(od->buffer + curpos, playChunk, bytes); - curpos = 0; - playChunk += bytes; - bytesToCopy -= bytes; - } - - memcpy(od->buffer + curpos, playChunk, bytesToCopy); - curpos += bytesToCopy; - playChunk += bytesToCopy; - - } - /* DEBUG("osx_play: unlock\n"); */ - pthread_mutex_unlock(&od->mutex); - - /* DEBUG("osx_play: leave\n"); */ - return 0; -} - -AudioOutputPlugin osxPlugin = { - "osx", - osx_testDefault, - osx_initDriver, - osx_finishDriver, - osx_openDevice, - osx_play, - osx_dropBufferedAudio, - osx_closeDevice, - NULL, /* sendMetadataFunc */ -}; - -#else - -#include - -DISABLED_AUDIO_OUTPUT_PLUGIN(osxPlugin) -#endif diff --git a/trunk/src/audioOutputs/audioOutput_pulse.c b/trunk/src/audioOutputs/audioOutput_pulse.c deleted file mode 100644 index 8948e0263..000000000 --- a/trunk/src/audioOutputs/audioOutput_pulse.c +++ /dev/null @@ -1,221 +0,0 @@ -/* the Music Player Daemon (MPD) - * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) - * 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 - -#ifdef HAVE_PULSE - -#include "../conf.h" -#include "../log.h" - -#include -#include - -#include -#include - -#define MPD_PULSE_NAME "mpd" -#define CONN_ATTEMPT_INTERVAL 60 - -typedef struct _PulseData { - pa_simple *s; - char *server; - char *sink; - int connAttempts; - time_t lastAttempt; -} PulseData; - -static PulseData *newPulseData(void) -{ - PulseData *ret; - - ret = xmalloc(sizeof(PulseData)); - - ret->s = NULL; - ret->server = NULL; - ret->sink = NULL; - ret->connAttempts = 0; - ret->lastAttempt = 0; - - return ret; -} - -static void freePulseData(PulseData * pd) -{ - if (pd->server) - free(pd->server); - if (pd->sink) - free(pd->sink); - free(pd); -} - -static int pulse_initDriver(AudioOutput * audioOutput, ConfigParam * param) -{ - BlockParam *server = NULL; - BlockParam *sink = NULL; - PulseData *pd; - - if (param) { - server = getBlockParam(param, "server"); - sink = getBlockParam(param, "sink"); - } - - pd = newPulseData(); - pd->server = server ? xstrdup(server->value) : NULL; - pd->sink = sink ? xstrdup(sink->value) : NULL; - audioOutput->data = pd; - - return 0; -} - -static void pulse_finishDriver(AudioOutput * audioOutput) -{ - freePulseData((PulseData *) audioOutput->data); -} - -static int pulse_testDefault(void) -{ - pa_simple *s; - pa_sample_spec ss; - int error; - - ss.format = PA_SAMPLE_S16NE; - ss.rate = 44100; - ss.channels = 2; - - s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL, - MPD_PULSE_NAME, &ss, NULL, NULL, &error); - if (!s) { - WARNING("Cannot connect to default PulseAudio server: %s\n", - pa_strerror(error)); - return -1; - } - - pa_simple_free(s); - - return 0; -} - -static int pulse_openDevice(AudioOutput * audioOutput) -{ - PulseData *pd; - AudioFormat *audioFormat; - pa_sample_spec ss; - time_t t; - int error; - - t = time(NULL); - pd = audioOutput->data; - audioFormat = &audioOutput->outAudioFormat; - - if (pd->connAttempts != 0 && - (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL) - return -1; - - pd->connAttempts++; - pd->lastAttempt = t; - - if (audioFormat->bits != 16) { - ERROR("PulseAudio doesn't support %i bit audio\n", - audioFormat->bits); - return -1; - } - - ss.format = PA_SAMPLE_S16NE; - ss.rate = audioFormat->sampleRate; - ss.channels = audioFormat->channels; - - pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, - pd->sink, audioOutput->name, &ss, NULL, NULL, - &error); - if (!pd->s) { - ERROR("Cannot connect to server in PulseAudio output " - "\"%s\" (attempt %i): %s\n", audioOutput->name, - pd->connAttempts, pa_strerror(error)); - return -1; - } - - pd->connAttempts = 0; - audioOutput->open = 1; - - DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i " - "channel audio at %i Hz\n", audioOutput->name, audioFormat->bits, - audioFormat->channels, audioFormat->sampleRate); - - return 0; -} - -static void pulse_dropBufferedAudio(AudioOutput * audioOutput) -{ - PulseData *pd; - int error; - - pd = audioOutput->data; - if (pa_simple_flush(pd->s, &error) < 0) - WARNING("Flush failed in PulseAudio output \"%s\": %s\n", - audioOutput->name, pa_strerror(error)); -} - -static void pulse_closeDevice(AudioOutput * audioOutput) -{ - PulseData *pd; - - pd = audioOutput->data; - if (pd->s) { - pa_simple_drain(pd->s, NULL); - pa_simple_free(pd->s); - } - - audioOutput->open = 0; -} - -static int pulse_playAudio(AudioOutput * audioOutput, char *playChunk, int size) -{ - PulseData *pd; - int error; - - pd = audioOutput->data; - - if (pa_simple_write(pd->s, playChunk, size, &error) < 0) { - ERROR("PulseAudio output \"%s\" disconnecting due to write " - "error: %s\n", audioOutput->name, pa_strerror(error)); - pulse_closeDevice(audioOutput); - return -1; - } - - return 0; -} - -AudioOutputPlugin pulsePlugin = { - "pulse", - pulse_testDefault, - pulse_initDriver, - pulse_finishDriver, - pulse_openDevice, - pulse_playAudio, - pulse_dropBufferedAudio, - pulse_closeDevice, - NULL, /* sendMetadataFunc */ -}; - -#else /* HAVE_PULSE */ - -DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin) -#endif /* HAVE_PULSE */ diff --git a/trunk/src/audioOutputs/audioOutput_shout.c b/trunk/src/audioOutputs/audioOutput_shout.c deleted file mode 100644 index 7d93f8f85..000000000 --- a/trunk/src/audioOutputs/audioOutput_shout.c +++ /dev/null @@ -1,636 +0,0 @@ -/* the Music Player Daemon (MPD) - * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) - * 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 - -#ifdef HAVE_SHOUT - -#include "../conf.h" -#include "../log.h" -#include "../pcm_utils.h" - -#include -#include - -#include -#include - -#define CONN_ATTEMPT_INTERVAL 60 - -static int shoutInitCount; - -/* lots of this code blatantly stolent from bossogg/bossao2 */ - -typedef struct _ShoutData { - shout_t *shoutConn; - int shoutError; - - ogg_stream_state os; - ogg_page og; - ogg_packet op; - ogg_packet header_main; - ogg_packet header_comments; - ogg_packet header_codebooks; - - vorbis_dsp_state vd; - vorbis_block vb; - vorbis_info vi; - vorbis_comment vc; - - float quality; - int bitrate; - - int opened; - - MpdTag *tag; - int tagToSend; - - int connAttempts; - time_t lastAttempt; - int last_err; - - /* just a pointer to audioOutput->outAudioFormat */ - AudioFormat *audioFormat; -} ShoutData; - -static ShoutData *newShoutData(void) -{ - ShoutData *ret = xmalloc(sizeof(ShoutData)); - - ret->shoutConn = shout_new(); - ret->opened = 0; - ret->tag = NULL; - ret->tagToSend = 0; - ret->bitrate = -1; - ret->quality = -2.0; - ret->connAttempts = 0; - ret->lastAttempt = 0; - ret->audioFormat = NULL; - ret->last_err = SHOUTERR_UNCONNECTED; - - return ret; -} - -static void freeShoutData(ShoutData * sd) -{ - if (sd->shoutConn) - shout_free(sd->shoutConn); - if (sd->tag) - freeMpdTag(sd->tag); - - free(sd); -} - -#define checkBlockParam(name) { \ - blockParam = getBlockParam(param, name); \ - if (!blockParam) { \ - FATAL("no \"%s\" defined for shout device defined at line " \ - "%i\n", name, param->line); \ - } \ -} - -static int myShout_initDriver(AudioOutput * audioOutput, ConfigParam * param) -{ - ShoutData *sd; - char *test; - int port; - char *host; - char *mount; - char *passwd; - char *user; - char *name; - BlockParam *blockParam; - unsigned int public = 0; - - sd = newShoutData(); - - if (shoutInitCount == 0) - shout_init(); - - shoutInitCount++; - - checkBlockParam("host"); - host = blockParam->value; - - checkBlockParam("mount"); - mount = blockParam->value; - - checkBlockParam("port"); - - port = strtol(blockParam->value, &test, 10); - - if (*test != '\0' || port <= 0) { - FATAL("shout port \"%s\" is not a positive integer, line %i\n", - blockParam->value, blockParam->line); - } - - checkBlockParam("password"); - passwd = blockParam->value; - - checkBlockParam("name"); - name = blockParam->value; - - blockParam = getBlockParam(param, "public"); - if (blockParam) { - if (0 == strcmp(blockParam->value, "yes")) { - public = 1; - } else if (0 == strcmp(blockParam->value, "no")) { - public = 0; - } else { - FATAL("public \"%s\" is not \"yes\" or \"no\" at line " - "%i\n", param->value, param->line); - } - } - - blockParam = getBlockParam(param, "user"); - if (blockParam) - user = blockParam->value; - else - user = "source"; - - blockParam = getBlockParam(param, "quality"); - - if (blockParam) { - int line = blockParam->line; - - sd->quality = strtod(blockParam->value, &test); - - if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) { - FATAL("shout quality \"%s\" is not a number in the " - "range -1 to 10, line %i\n", blockParam->value, - blockParam->line); - } - - blockParam = getBlockParam(param, "bitrate"); - - if (blockParam) { - FATAL("quality (line %i) and bitrate (line %i) are " - "both defined for shout output\n", line, - blockParam->line); - } - } else { - blockParam = getBlockParam(param, "bitrate"); - - if (!blockParam) { - FATAL("neither bitrate nor quality defined for shout " - "output at line %i\n", param->line); - } - - sd->bitrate = strtol(blockParam->value, &test, 10); - - if (*test != '\0' || sd->bitrate <= 0) { - FATAL("bitrate at line %i should be a positive integer " - "\n", blockParam->line); - } - } - - checkBlockParam("format"); - sd->audioFormat = &audioOutput->outAudioFormat; - - if (shout_set_host(sd->shoutConn, host) != SHOUTERR_SUCCESS || - shout_set_port(sd->shoutConn, port) != SHOUTERR_SUCCESS || - shout_set_password(sd->shoutConn, passwd) != SHOUTERR_SUCCESS || - shout_set_mount(sd->shoutConn, mount) != SHOUTERR_SUCCESS || - shout_set_name(sd->shoutConn, name) != SHOUTERR_SUCCESS || - shout_set_user(sd->shoutConn, user) != SHOUTERR_SUCCESS || - shout_set_public(sd->shoutConn, public) != SHOUTERR_SUCCESS || - shout_set_nonblocking(sd->shoutConn, 1) != SHOUTERR_SUCCESS || - shout_set_format(sd->shoutConn, SHOUT_FORMAT_VORBIS) - != SHOUTERR_SUCCESS || - shout_set_protocol(sd->shoutConn, SHOUT_PROTOCOL_HTTP) - != SHOUTERR_SUCCESS || - shout_set_agent(sd->shoutConn, "MPD") != SHOUTERR_SUCCESS) { - FATAL("error configuring shout defined at line %i: %s\n", - param->line, shout_get_error(sd->shoutConn)); - } - - /* optional paramters */ - blockParam = getBlockParam(param, "genre"); - if (blockParam && shout_set_genre(sd->shoutConn, blockParam->value)) { - FATAL("error configuring shout defined at line %i: %s\n", - param->line, shout_get_error(sd->shoutConn)); - } - - blockParam = getBlockParam(param, "description"); - if (blockParam && shout_set_description(sd->shoutConn, - blockParam->value)) { - FATAL("error configuring shout defined at line %i: %s\n", - param->line, shout_get_error(sd->shoutConn)); - } - - { - char temp[11]; - memset(temp, 0, sizeof(temp)); - - snprintf(temp, sizeof(temp), "%d", sd->audioFormat->channels); - shout_set_audio_info(sd->shoutConn, SHOUT_AI_CHANNELS, temp); - - snprintf(temp, sizeof(temp), "%d", sd->audioFormat->sampleRate); - - shout_set_audio_info(sd->shoutConn, SHOUT_AI_SAMPLERATE, temp); - - if (sd->quality >= -1.0) { - snprintf(temp, sizeof(temp), "%2.2f", sd->quality); - shout_set_audio_info(sd->shoutConn, SHOUT_AI_QUALITY, - temp); - } else { - snprintf(temp, sizeof(temp), "%d", sd->bitrate); - shout_set_audio_info(sd->shoutConn, SHOUT_AI_BITRATE, - temp); - } - } - - audioOutput->data = sd; - - return 0; -} - -static int myShout_handleError(ShoutData * sd, int err) -{ - switch (err) { - case SHOUTERR_SUCCESS: - break; - case SHOUTERR_UNCONNECTED: - case SHOUTERR_SOCKET: - ERROR("Lost shout connection to %s:%i : %s\n", - shout_get_host(sd->shoutConn), - shout_get_port(sd->shoutConn), - shout_get_error(sd->shoutConn)); - sd->shoutError = 1; - return -1; - default: - ERROR("shout: connection to %s:%i error : %s\n", - shout_get_host(sd->shoutConn), - shout_get_port(sd->shoutConn), - shout_get_error(sd->shoutConn)); - sd->shoutError = 1; - return -1; - } - - return 0; -} - -static int write_page(ShoutData * sd) -{ - int err = 0; - - /*DEBUG("shout_delay: %i\n", shout_delay(sd->shoutConn)); */ - shout_sync(sd->shoutConn); - err = shout_send(sd->shoutConn, sd->og.header, sd->og.header_len); - if (myShout_handleError(sd, err) < 0) - return -1; - err = shout_send(sd->shoutConn, sd->og.body, sd->og.body_len); - if (myShout_handleError(sd, err) < 0) - return -1; - - return 0; -} - -static void finishEncoder(ShoutData * sd) -{ - vorbis_analysis_wrote(&sd->vd, 0); - - while (vorbis_analysis_blockout(&sd->vd, &sd->vb) == 1) { - vorbis_analysis(&sd->vb, NULL); - vorbis_bitrate_addblock(&sd->vb); - while (vorbis_bitrate_flushpacket(&sd->vd, &sd->op)) { - ogg_stream_packetin(&sd->os, &sd->op); - } - } -} - -static int flushEncoder(ShoutData * sd) -{ - return (ogg_stream_pageout(&sd->os, &sd->og) > 0); -} - -static void clearEncoder(ShoutData * sd) -{ - finishEncoder(sd); - while (1 == flushEncoder(sd)) { - if (!sd->shoutError) - write_page(sd); - } - - vorbis_comment_clear(&sd->vc); - ogg_stream_clear(&sd->os); - vorbis_block_clear(&sd->vb); - vorbis_dsp_clear(&sd->vd); - vorbis_info_clear(&sd->vi); -} - -static void myShout_closeShoutConn(ShoutData * sd) -{ - if (sd->opened) { - clearEncoder(sd); - - if (shout_close(sd->shoutConn) != SHOUTERR_SUCCESS) { - ERROR("problem closing connection to shout server: " - "%s\n", shout_get_error(sd->shoutConn)); - } - } - - sd->last_err = SHOUTERR_UNCONNECTED; - sd->opened = 0; -} - -static void myShout_finishDriver(AudioOutput * audioOutput) -{ - ShoutData *sd = (ShoutData *) audioOutput->data; - - myShout_closeShoutConn(sd); - - freeShoutData(sd); - - shoutInitCount--; - - if (shoutInitCount == 0) - shout_shutdown(); -} - -static void myShout_dropBufferedAudio(AudioOutput * audioOutput) -{ - /* needs to be implemented */ -} - -static void myShout_closeDevice(AudioOutput * audioOutput) -{ - ShoutData *sd = (ShoutData *) audioOutput->data; - - myShout_closeShoutConn(sd); - - audioOutput->open = 0; -} - -#define addTag(name, value) { \ - if(value) vorbis_comment_add_tag(&(sd->vc), name, value); \ -} - -static void copyTagToVorbisComment(ShoutData * sd) -{ - if (sd->tag) { - int i; - - for (i = 0; i < sd->tag->numOfItems; i++) { - switch (sd->tag->items[i].type) { - case TAG_ITEM_ARTIST: - addTag("ARTIST", sd->tag->items[i].value); - break; - case TAG_ITEM_ALBUM: - addTag("ALBUM", sd->tag->items[i].value); - break; - case TAG_ITEM_TITLE: - addTag("TITLE", sd->tag->items[i].value); - break; - } - } - } -} - -static int initEncoder(ShoutData * sd) -{ - vorbis_info_init(&(sd->vi)); - - if (sd->quality >= -1.0) { - if (0 != vorbis_encode_init_vbr(&(sd->vi), - sd->audioFormat->channels, - sd->audioFormat->sampleRate, - sd->quality * 0.1)) { - ERROR("problem setting up vorbis encoder for shout\n"); - vorbis_info_clear(&(sd->vi)); - return -1; - } - } else { - if (0 != vorbis_encode_init(&(sd->vi), - sd->audioFormat->channels, - sd->audioFormat->sampleRate, -1.0, - sd->bitrate * 1000, -1.0)) { - ERROR("problem setting up vorbis encoder for shout\n"); - vorbis_info_clear(&(sd->vi)); - return -1; - } - } - - vorbis_analysis_init(&(sd->vd), &(sd->vi)); - vorbis_block_init(&(sd->vd), &(sd->vb)); - - ogg_stream_init(&(sd->os), rand()); - - vorbis_comment_init(&(sd->vc)); - - return 0; -} - -static int myShout_openShoutConn(AudioOutput * audioOutput) -{ - ShoutData *sd = (ShoutData *) audioOutput->data; - time_t t = time(NULL); - - if (sd->connAttempts != 0 && - (t - sd->lastAttempt) < CONN_ATTEMPT_INTERVAL) { - return -1; - } - - sd->connAttempts++; - - if (sd->last_err == SHOUTERR_UNCONNECTED) - sd->last_err = shout_open(sd->shoutConn); - switch (sd->last_err) { - case SHOUTERR_SUCCESS: - case SHOUTERR_CONNECTED: - break; - case SHOUTERR_BUSY: - sd->last_err = shout_get_connected(sd->shoutConn); - if (sd->last_err == SHOUTERR_CONNECTED) - break; - return -1; - default: - sd->lastAttempt = t; - ERROR("problem opening connection to shout server %s:%i " - "(attempt %i): %s\n", - shout_get_host(sd->shoutConn), - shout_get_port(sd->shoutConn), - sd->connAttempts, shout_get_error(sd->shoutConn)); - return -1; - } - - if (initEncoder(sd) < 0) { - shout_close(sd->shoutConn); - return -1; - } - - sd->shoutError = 0; - - copyTagToVorbisComment(sd); - - vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main), - &(sd->header_comments), - &(sd->header_codebooks)); - - ogg_stream_packetin(&(sd->os), &(sd->header_main)); - ogg_stream_packetin(&(sd->os), &(sd->header_comments)); - ogg_stream_packetin(&(sd->os), &(sd->header_codebooks)); - - sd->opened = 1; - sd->tagToSend = 0; - - while (ogg_stream_flush(&(sd->os), &(sd->og))) { - if (write_page(sd) < 0) { - myShout_closeShoutConn(sd); - return -1; - } - } - - sd->connAttempts = 0; - - return 0; -} - -static int myShout_openDevice(AudioOutput * audioOutput) -{ - ShoutData *sd = (ShoutData *) audioOutput->data; - - audioOutput->open = 1; - - if (sd->opened) - return 0; - - if (myShout_openShoutConn(audioOutput) < 0) { - audioOutput->open = 0; - return -1; - } - - return 0; -} - -static void myShout_sendMetadata(ShoutData * sd) -{ - if (!sd->opened || !sd->tag) - return; - - clearEncoder(sd); - if (initEncoder(sd) < 0) - return; - - copyTagToVorbisComment(sd); - - vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main), - &(sd->header_comments), - &(sd->header_codebooks)); - - ogg_stream_packetin(&(sd->os), &(sd->header_main)); - ogg_stream_packetin(&(sd->os), &(sd->header_comments)); - ogg_stream_packetin(&(sd->os), &(sd->header_codebooks)); - - while (ogg_stream_flush(&(sd->os), &(sd->og))) { - if (write_page(sd) < 0) { - myShout_closeShoutConn(sd); - return; - } - } - - /*if(sd->tag) freeMpdTag(sd->tag); - sd->tag = NULL; */ - sd->tagToSend = 0; -} - -static int myShout_play(AudioOutput * audioOutput, char *playChunk, int size) -{ - int i, j; - ShoutData *sd = (ShoutData *) audioOutput->data; - float **vorbbuf; - int samples; - int bytes = sd->audioFormat->bits / 8; - - if (sd->opened && sd->tagToSend) - myShout_sendMetadata(sd); - - if (!sd->opened) { - if (myShout_openShoutConn(audioOutput) < 0) { - return -1; - } - } - - samples = size / (bytes * sd->audioFormat->channels); - - /* this is for only 16-bit audio */ - - vorbbuf = vorbis_analysis_buffer(&(sd->vd), samples); - - for (i = 0; i < samples; i++) { - for (j = 0; j < sd->audioFormat->channels; j++) { - vorbbuf[j][i] = (*((mpd_sint16 *) playChunk)) / 32768.0; - playChunk += bytes; - } - } - - vorbis_analysis_wrote(&(sd->vd), samples); - - while (1 == vorbis_analysis_blockout(&(sd->vd), &(sd->vb))) { - vorbis_analysis(&(sd->vb), NULL); - vorbis_bitrate_addblock(&(sd->vb)); - - while (vorbis_bitrate_flushpacket(&(sd->vd), &(sd->op))) { - ogg_stream_packetin(&(sd->os), &(sd->op)); - } - } - - while (ogg_stream_pageout(&(sd->os), &(sd->og)) != 0) { - if (write_page(sd) < 0) { - myShout_closeShoutConn(sd); - return -1; - } - } - - return 0; -} - -static void myShout_setTag(AudioOutput * audioOutput, MpdTag * tag) -{ - ShoutData *sd = (ShoutData *) audioOutput->data; - - if (sd->tag) - freeMpdTag(sd->tag); - sd->tag = NULL; - sd->tagToSend = 0; - - if (!tag) - return; - - sd->tag = mpdTagDup(tag); - sd->tagToSend = 1; -} - -AudioOutputPlugin shoutPlugin = { - "shout", - NULL, - myShout_initDriver, - myShout_finishDriver, - myShout_openDevice, - myShout_play, - myShout_dropBufferedAudio, - myShout_closeDevice, - myShout_setTag, -}; - -#else - -DISABLED_AUDIO_OUTPUT_PLUGIN(shoutPlugin) -#endif -- cgit v1.2.3