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-rw-r--r--src/audioOutputs/audioOutput_alsa.c444
-rw-r--r--src/audioOutputs/audioOutput_ao.c253
-rw-r--r--src/audioOutputs/audioOutput_fifo.c290
-rw-r--r--src/audioOutputs/audioOutput_jack.c486
-rw-r--r--src/audioOutputs/audioOutput_mvp.c280
-rw-r--r--src/audioOutputs/audioOutput_null.c85
-rw-r--r--src/audioOutputs/audioOutput_oss.c571
-rw-r--r--src/audioOutputs/audioOutput_osx.c368
-rw-r--r--src/audioOutputs/audioOutput_pulse.c218
-rw-r--r--src/audioOutputs/audioOutput_shout.c596
-rw-r--r--src/audioOutputs/audioOutput_shout.h93
-rw-r--r--src/audioOutputs/audioOutput_shout_mp3.c189
-rw-r--r--src/audioOutputs/audioOutput_shout_ogg.c306
13 files changed, 0 insertions, 4179 deletions
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
deleted file mode 100644
index 1845f1b76..000000000
--- a/src/audioOutputs/audioOutput_alsa.c
+++ /dev/null
@@ -1,444 +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 "../output_api.h"
-
-#ifdef HAVE_ALSA
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-static const char default_device[] = "default";
-
-#define MPD_ALSA_RETRY_NR 5
-
-#include "../utils.h"
-#include "../log.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 {
- const char *device;
-
- /** the mode flags passed to snd_pcm_open */
- int mode;
-
- snd_pcm_t *pcmHandle;
- alsa_writei_t *writei;
- unsigned int buffer_time;
- unsigned int period_time;
- int sampleSize;
- int useMmap;
-} AlsaData;
-
-static AlsaData *newAlsaData(void)
-{
- AlsaData *ret = xmalloc(sizeof(AlsaData));
-
- ret->device = default_device;
- ret->mode = 0;
- ret->pcmHandle = NULL;
- ret->writei = snd_pcm_writei;
- ret->useMmap = 0;
- ret->buffer_time = 0;
- ret->period_time = 0;
-
- return ret;
-}
-
-static void freeAlsaData(AlsaData * ad)
-{
- if (ad->device && ad->device != default_device)
- xfree(ad->device);
- free(ad);
-}
-
-static void
-alsa_configure(AlsaData *ad, ConfigParam *param)
-{
- BlockParam *bp;
-
- if ((bp = getBlockParam(param, "device")))
- ad->device = xstrdup(bp->value);
- ad->useMmap = getBoolBlockParam(param, "use_mmap", 1);
- if (ad->useMmap == CONF_BOOL_UNSET)
- ad->useMmap = 0;
- if ((bp = getBlockParam(param, "buffer_time")))
- ad->buffer_time = atoi(bp->value);
- if ((bp = getBlockParam(param, "period_time")))
- ad->period_time = atoi(bp->value);
-
-#ifdef SND_PCM_NO_AUTO_RESAMPLE
- if (!getBoolBlockParam(param, "auto_resample", true))
- ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_CHANNELS
- if (!getBoolBlockParam(param, "auto_channels", true))
- ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_FORMAT
- if (!getBoolBlockParam(param, "auto_format", true))
- ad->mode |= SND_PCM_NO_AUTO_FORMAT;
-#endif
-}
-
-static void *alsa_initDriver(mpd_unused struct audio_output *ao,
- mpd_unused const struct audio_format *audio_format,
- ConfigParam * param)
-{
- /* no need for pthread_once thread-safety when reading config */
- static int free_global_registered;
- AlsaData *ad = newAlsaData();
-
- if (!free_global_registered) {
- atexit((void(*)(void))snd_config_update_free_global);
- free_global_registered = 1;
- }
-
- if (param)
- alsa_configure(ad, param);
-
- return ad;
-}
-
-static void alsa_finishDriver(void *data)
-{
- AlsaData *ad = data;
-
- freeAlsaData(ad);
-}
-
-static int alsa_testDefault(void)
-{
- snd_pcm_t *handle;
-
- int ret = snd_pcm_open(&handle, default_device,
- SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
- if (ret) {
- WARNING("Error opening default ALSA device: %s\n",
- snd_strerror(-ret));
- return -1;
- } else
- snd_pcm_close(handle);
-
- return 0;
-}
-
-static snd_pcm_format_t get_bitformat(const struct audio_format *af)
-{
- switch (af->bits) {
- case 8: return SND_PCM_FORMAT_S8;
- case 16: return SND_PCM_FORMAT_S16;
- case 24: return SND_PCM_FORMAT_S24;
- case 32: return SND_PCM_FORMAT_S32;
- }
- return SND_PCM_FORMAT_UNKNOWN;
-}
-
-static int alsa_openDevice(void *data, struct audio_format *audioFormat)
-{
- AlsaData *ad = data;
- snd_pcm_format_t bitformat;
- snd_pcm_hw_params_t *hwparams;
- snd_pcm_sw_params_t *swparams;
- unsigned int sample_rate = audioFormat->sample_rate;
- 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;
-
- if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN)
- ERROR("ALSA device \"%s\" doesn't support %u bit audio\n",
- ad->device, audioFormat->bits);
-
- err = snd_pcm_open(&ad->pcmHandle, ad->device,
- SND_PCM_STREAM_PLAYBACK, ad->mode);
- if (err < 0) {
- ad->pcmHandle = NULL;
- 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 == -EINVAL && audioFormat->bits != 16) {
- /* fall back to 16 bit, let pcm_utils.c do the conversion */
- err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams,
- SND_PCM_FORMAT_S16);
- if (err == 0) {
- DEBUG("ALSA device \"%s\": converting %u bit to 16 bit\n",
- ad->device, audioFormat->bits);
- audioFormat->bits = 16;
- }
- }
-
- if (err < 0) {
- ERROR("ALSA device \"%s\" does not support %u 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 = (int8_t)channels;
-
- err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
- &sample_rate, NULL);
- if (err < 0 || sample_rate == 0) {
- ERROR("ALSA device \"%s\" does not support %u Hz audio\n",
- ad->device, audioFormat->sample_rate);
- goto fail;
- }
- audioFormat->sample_rate = sample_rate;
-
- if (ad->buffer_time > 0) {
- 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;
- }
-
- if (period_time_ro > 0) {
- 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 > 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;
-
- /* 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";
- err = snd_pcm_sw_params(ad->pcmHandle, swparams);
- if (err < 0)
- goto error;
-
- ad->sampleSize = audio_format_frame_size(audioFormat);
-
- DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at "
- "%u Hz\n", ad->device, audioFormat->bits,
- channels, sample_rate);
-
- 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;
- 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 = snd_pcm_resume(ad->pcmHandle);
- if (err == -EAGAIN)
- return 0;
- /* fall-through to snd_pcm_prepare: */
- 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;
- /* this is no error, so just keep running */
- case SND_PCM_STATE_RUNNING:
- err = 0;
- break;
- default:
- /* unknown state, do nothing */
- break;
- }
-
- return err;
-}
-
-static void alsa_dropBufferedAudio(void *data)
-{
- AlsaData *ad = data;
-
- alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
-}
-
-static void alsa_closeDevice(void *data)
-{
- AlsaData *ad = data;
-
- if (ad->pcmHandle) {
- if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) {
- snd_pcm_drain(ad->pcmHandle);
- }
- snd_pcm_close(ad->pcmHandle);
- ad->pcmHandle = NULL;
- }
-}
-
-static int alsa_playAudio(void *data, const char *playChunk, size_t size)
-{
- AlsaData *ad = 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(ad);
- return -1;
- }
- continue;
- }
-
- playChunk += ret * ad->sampleSize;
- size -= ret;
- }
-
- return 0;
-}
-
-const struct audio_output_plugin alsaPlugin = {
- .name = "alsa",
- .test_default_device = alsa_testDefault,
- .init = alsa_initDriver,
- .finish = alsa_finishDriver,
- .open = alsa_openDevice,
- .play = alsa_playAudio,
- .cancel = alsa_dropBufferedAudio,
- .close = alsa_closeDevice,
-};
-
-#else /* HAVE ALSA */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
-#endif /* HAVE_ALSA */
diff --git a/src/audioOutputs/audioOutput_ao.c b/src/audioOutputs/audioOutput_ao.c
deleted file mode 100644
index e731f972a..000000000
--- a/src/audioOutputs/audioOutput_ao.c
+++ /dev/null
@@ -1,253 +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 "../output_api.h"
-
-#ifdef HAVE_AO
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <ao/ao.h>
-
-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 void *audioOutputAo_initDriver(struct audio_output *ao,
- mpd_unused const struct audio_format *audio_format,
- ConfigParam * param)
-{
- ao_info *ai;
- char *duplicated;
- char *stk1;
- char *stk2;
- char *n1;
- char *key;
- char *value;
- char *test;
- AoData *ad = newAoData();
- BlockParam *blockParam;
-
- 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,
- audio_output_get_name(ao));
-
- blockParam = getBlockParam(param, "options");
-
- if (blockParam) {
- duplicated = xstrdup(blockParam->value);
- } else
- duplicated = xstrdup("");
-
- if (strlen(duplicated)) {
- stk1 = NULL;
- n1 = strtok_r(duplicated, ";", &stk1);
- while (n1) {
- stk2 = NULL;
- key = strtok_r(n1, "=", &stk2);
- if (!key)
- FATAL("problems parsing options \"%s\"\n", n1);
- /*found = 0;
- for(i=0;i<ai->option_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(duplicated);
-
- return ad;
-}
-
-static void freeAoData(AoData * ad)
-{
- ao_free_options(ad->options);
- free(ad);
-}
-
-static void audioOutputAo_finishDriver(void *data)
-{
- AoData *ad = (AoData *)data;
- freeAoData(ad);
-
- driverInitCount--;
-
- if (driverInitCount == 0)
- ao_shutdown();
-}
-
-static void audioOutputAo_dropBufferedAudio(mpd_unused void *data)
-{
- /* not supported by libao */
-}
-
-static void audioOutputAo_closeDevice(void *data)
-{
- AoData *ad = (AoData *)data;
-
- if (ad->device) {
- ao_close(ad->device);
- ad->device = NULL;
- }
-}
-
-static int audioOutputAo_openDevice(void *data,
- struct audio_format *audio_format)
-{
- ao_sample_format format;
- AoData *ad = (AoData *)data;
-
- if (ad->device) {
- audioOutputAo_closeDevice(ad);
- }
-
- format.bits = audio_format->bits;
- format.rate = audio_format->sample_rate;
- format.byte_format = AO_FMT_NATIVE;
- format.channels = audio_format->channels;
-
- ad->device = ao_open_live(ad->driverId, &format, ad->options);
-
- if (ad->device == NULL)
- return -1;
-
- return 0;
-}
-
-/**
- * For whatever reason, libao wants a non-const pointer. Let's hope
- * it does not write to the buffer, and use the union deconst hack to
- * work around this API misdesign.
- */
-static int ao_play_deconst(ao_device *device, const void *output_samples,
- uint_32 num_bytes)
-{
- union {
- const void *in;
- void *out;
- } u;
-
- u.in = output_samples;
- return ao_play(device, u.out, num_bytes);
-}
-
-static int audioOutputAo_play(void *data, const char *playChunk, size_t size)
-{
- AoData *ad = (AoData *)data;
- size_t chunk_size;
-
- if (ad->device == NULL)
- return -1;
-
- while (size > 0) {
- chunk_size = (size_t)ad->writeSize > size
- ? size : (size_t)ad->writeSize;
-
- if (ao_play_deconst(ad->device, playChunk, chunk_size) == 0) {
- audioOutputAo_error();
- ERROR("closing audio device due to write error\n");
- audioOutputAo_closeDevice(ad);
- return -1;
- }
-
- playChunk += chunk_size;
- size -= chunk_size;
- }
-
- return 0;
-}
-
-const struct audio_output_plugin aoPlugin = {
- .name = "ao",
- .init = audioOutputAo_initDriver,
- .finish = audioOutputAo_finishDriver,
- .open = audioOutputAo_openDevice,
- .play = audioOutputAo_play,
- .cancel = audioOutputAo_dropBufferedAudio,
- .close = audioOutputAo_closeDevice,
-};
-
-#else
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(aoPlugin)
-#endif
diff --git a/src/audioOutputs/audioOutput_fifo.c b/src/audioOutputs/audioOutput_fifo.c
deleted file mode 100644
index d7eb91ff6..000000000
--- a/src/audioOutputs/audioOutput_fifo.c
+++ /dev/null
@@ -1,290 +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 "../output_api.h"
-
-#ifdef HAVE_FIFO
-
-#include "../log.h"
-#include "../utils.h"
-#include "../timer.h"
-
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-
-#define FIFO_BUFFER_SIZE 65536 /* pipe capacity on Linux >= 2.6.11 */
-
-typedef struct _FifoData {
- char *path;
- int input;
- int output;
- int created;
- Timer *timer;
-} FifoData;
-
-static FifoData *newFifoData(void)
-{
- FifoData *ret;
-
- ret = xmalloc(sizeof(FifoData));
-
- ret->path = NULL;
- ret->input = -1;
- ret->output = -1;
- ret->created = 0;
- ret->timer = NULL;
-
- return ret;
-}
-
-static void freeFifoData(FifoData *fd)
-{
- if (fd->path)
- free(fd->path);
-
- if (fd->timer)
- timer_free(fd->timer);
-
- free(fd);
-}
-
-static void removeFifo(FifoData *fd)
-{
- DEBUG("Removing FIFO \"%s\"\n", fd->path);
-
- if (unlink(fd->path) < 0) {
- ERROR("Could not remove FIFO \"%s\": %s\n",
- fd->path, strerror(errno));
- return;
- }
-
- fd->created = 0;
-}
-
-static void closeFifo(FifoData *fd)
-{
- struct stat st;
-
- if (fd->input >= 0) {
- close(fd->input);
- fd->input = -1;
- }
-
- if (fd->output >= 0) {
- close(fd->output);
- fd->output = -1;
- }
-
- if (fd->created && (stat(fd->path, &st) == 0))
- removeFifo(fd);
-}
-
-static int makeFifo(FifoData *fd)
-{
- if (mkfifo(fd->path, 0666) < 0) {
- ERROR("Couldn't create FIFO \"%s\": %s\n",
- fd->path, strerror(errno));
- return -1;
- }
-
- fd->created = 1;
-
- return 0;
-}
-
-static int checkFifo(FifoData *fd)
-{
- struct stat st;
-
- if (stat(fd->path, &st) < 0) {
- if (errno == ENOENT) {
- /* Path doesn't exist */
- return makeFifo(fd);
- }
-
- ERROR("Failed to stat FIFO \"%s\": %s\n",
- fd->path, strerror(errno));
- return -1;
- }
-
- if (!S_ISFIFO(st.st_mode)) {
- ERROR("\"%s\" already exists, but is not a FIFO\n", fd->path);
- return -1;
- }
-
- return 0;
-}
-
-static int openFifo(FifoData *fd)
-{
- if (checkFifo(fd) < 0)
- return -1;
-
- fd->input = open(fd->path, O_RDONLY|O_NONBLOCK);
- if (fd->input < 0) {
- ERROR("Could not open FIFO \"%s\" for reading: %s\n",
- fd->path, strerror(errno));
- closeFifo(fd);
- return -1;
- }
-
- fd->output = open(fd->path, O_WRONLY|O_NONBLOCK);
- if (fd->output < 0) {
- ERROR("Could not open FIFO \"%s\" for writing: %s\n",
- fd->path, strerror(errno));
- closeFifo(fd);
- return -1;
- }
-
- return 0;
-}
-
-static void *fifo_initDriver(mpd_unused struct audio_output *ao,
- mpd_unused const struct audio_format *audio_format,
- ConfigParam *param)
-{
- FifoData *fd;
- BlockParam *blockParam;
- char *path;
-
- blockParam = getBlockParam(param, "path");
- if (!blockParam) {
- FATAL("No \"path\" parameter specified for fifo output "
- "defined at line %i\n", param->line);
- }
-
- path = parsePath(blockParam->value);
- if (!path) {
- FATAL("Could not parse \"path\" parameter for fifo output "
- "at line %i\n", blockParam->line);
- }
-
- fd = newFifoData();
- fd->path = path;
-
- if (openFifo(fd) < 0) {
- freeFifoData(fd);
- return NULL;
- }
-
- return fd;
-}
-
-static void fifo_finishDriver(void *data)
-{
- FifoData *fd = (FifoData *)data;
-
- closeFifo(fd);
- freeFifoData(fd);
-}
-
-static int fifo_openDevice(void *data,
- struct audio_format *audio_format)
-{
- FifoData *fd = (FifoData *)data;
-
- if (fd->timer)
- timer_free(fd->timer);
-
- fd->timer = timer_new(audio_format);
-
- return 0;
-}
-
-static void fifo_closeDevice(void *data)
-{
- FifoData *fd = (FifoData *)data;
-
- if (fd->timer) {
- timer_free(fd->timer);
- fd->timer = NULL;
- }
-}
-
-static void fifo_dropBufferedAudio(void *data)
-{
- FifoData *fd = (FifoData *)data;
- char buf[FIFO_BUFFER_SIZE];
- int bytes = 1;
-
- timer_reset(fd->timer);
-
- while (bytes > 0 && errno != EINTR)
- bytes = read(fd->input, buf, FIFO_BUFFER_SIZE);
-
- if (bytes < 0 && errno != EAGAIN) {
- WARNING("Flush of FIFO \"%s\" failed: %s\n",
- fd->path, strerror(errno));
- }
-}
-
-static int fifo_playAudio(void *data,
- const char *playChunk, size_t size)
-{
- FifoData *fd = (FifoData *)data;
- size_t offset = 0;
- ssize_t bytes;
-
- if (!fd->timer->started)
- timer_start(fd->timer);
- else
- timer_sync(fd->timer);
-
- timer_add(fd->timer, size);
-
- while (size) {
- bytes = write(fd->output, playChunk + offset, size);
- if (bytes < 0) {
- switch (errno) {
- case EAGAIN:
- /* The pipe is full, so empty it */
- fifo_dropBufferedAudio(fd);
- continue;
- case EINTR:
- continue;
- }
-
- ERROR("Closing FIFO output \"%s\" due to write error: "
- "%s\n", fd->path, strerror(errno));
- fifo_closeDevice(fd);
- return -1;
- }
-
- size -= bytes;
- offset += bytes;
- }
-
- return 0;
-}
-
-const struct audio_output_plugin fifoPlugin = {
- .name = "fifo",
- .init = fifo_initDriver,
- .finish = fifo_finishDriver,
- .open = fifo_openDevice,
- .play = fifo_playAudio,
- .cancel = fifo_dropBufferedAudio,
- .close = fifo_closeDevice,
-};
-
-#else /* HAVE_FIFO */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(fifoPlugin)
-
-#endif /* !HAVE_FIFO */
diff --git a/src/audioOutputs/audioOutput_jack.c b/src/audioOutputs/audioOutput_jack.c
deleted file mode 100644
index 6f2bcd3a1..000000000
--- a/src/audioOutputs/audioOutput_jack.c
+++ /dev/null
@@ -1,486 +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 "../output_api.h"
-
-#ifdef HAVE_JACK
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <assert.h>
-
-#include <jack/jack.h>
-#include <jack/types.h>
-#include <jack/ringbuffer.h>
-
-static const size_t sample_size = sizeof(jack_default_audio_sample_t);
-
-struct jack_data {
- struct audio_output *ao;
-
- /* configuration */
- const char *name;
- const char *output_ports[2];
- int ringbuffer_size;
-
- /* for srate() only */
- struct audio_format *audio_format;
-
- /* jack library stuff */
- jack_port_t *ports[2];
- jack_client_t *client;
- jack_ringbuffer_t *ringbuffer[2];
- int bps;
- int shutdown;
-};
-
-static struct jack_data *
-mpd_jack_new(void)
-{
- struct jack_data *ret;
-
- ret = xcalloc(sizeof(*ret), 1);
-
- ret->name = "mpd";
- ret->ringbuffer_size = 32768;
-
- return ret;
-}
-
-static void
-mpd_jack_client_free(struct jack_data *jd)
-{
- assert(jd != NULL);
-
- if (jd->client != NULL) {
- jack_deactivate(jd->client);
- jack_client_close(jd->client);
- jd->client = NULL;
- }
-
- if (jd->ringbuffer[0] != NULL) {
- jack_ringbuffer_free(jd->ringbuffer[0]);
- jd->ringbuffer[0] = NULL;
- }
-
- if (jd->ringbuffer[1] != NULL) {
- jack_ringbuffer_free(jd->ringbuffer[1]);
- jd->ringbuffer[1] = NULL;
- }
-}
-
-static void
-mpd_jack_free(struct jack_data *jd)
-{
- int i;
-
- assert(jd != NULL);
-
- mpd_jack_client_free(jd);
-
- if (strcmp(jd->name, "mpd") != 0)
- xfree(jd->name);
-
- for ( i = ARRAY_SIZE(jd->output_ports); --i >= 0; ) {
- if (!jd->output_ports[i])
- continue;
- xfree(jd->output_ports[i]);
- }
-
- free(jd);
-}
-
-static void
-mpd_jack_finish(void *data)
-{
- struct jack_data *jd = data;
- mpd_jack_free(jd);
-}
-
-static int
-mpd_jack_srate(mpd_unused jack_nframes_t rate, void *data)
-{
- struct jack_data *jd = (struct jack_data *)data;
- struct audio_format *audioFormat = jd->audio_format;
-
- audioFormat->sample_rate = (int)jack_get_sample_rate(jd->client);
-
- return 0;
-}
-
-static int
-mpd_jack_process(jack_nframes_t nframes, void *arg)
-{
- struct jack_data *jd = (struct jack_data *) arg;
- jack_default_audio_sample_t *out;
- size_t available;
-
- if (nframes <= 0)
- return 0;
-
- for (unsigned i = 0; i < 2; ++i) {
- available = jack_ringbuffer_read_space(jd->ringbuffer[i]);
- assert(available % sample_size == 0);
- available /= sample_size;
- if (available > nframes)
- available = nframes;
-
- out = jack_port_get_buffer(jd->ports[i], nframes);
- jack_ringbuffer_read(jd->ringbuffer[i],
- (char *)out, available * sample_size);
-
- while (available < nframes)
- /* ringbuffer underrun, fill with silence */
- out[available++] = 0.0;
- }
-
- return 0;
-}
-
-static void
-mpd_jack_shutdown(void *arg)
-{
- struct jack_data *jd = (struct jack_data *) arg;
- jd->shutdown = 1;
-}
-
-static void
-set_audioformat(struct jack_data *jd, struct audio_format *audio_format)
-{
- audio_format->sample_rate = jack_get_sample_rate(jd->client);
- DEBUG("samplerate = %u\n", audio_format->sample_rate);
- audio_format->channels = 2;
-
- if (audio_format->bits != 16 && audio_format->bits != 24)
- audio_format->bits = 24;
-
- jd->bps = audio_format->channels
- * sizeof(jack_default_audio_sample_t)
- * audio_format->sample_rate;
-}
-
-static void
-mpd_jack_error(const char *msg)
-{
- ERROR("jack: %s\n", msg);
-}
-
-static void *
-mpd_jack_init(struct audio_output *ao,
- mpd_unused const struct audio_format *audio_format,
- ConfigParam *param)
-{
- struct jack_data *jd;
- BlockParam *bp;
- char *endptr;
- int val;
- char *cp = NULL;
-
- jd = mpd_jack_new();
- jd->ao = ao;
-
- DEBUG("mpd_jack_init (pid=%d)\n", getpid());
- if (param == NULL)
- return jd;
-
- 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';
- jd->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);
-
- jd->output_ports[1] = xstrdup(cp);
-
- if (strchr(cp,','))
- FATAL("Only %d values are supported for '%s' "
- "at line %d\n",
- (int)ARRAY_SIZE(jd->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) {
- jd->ringbuffer_size = val < 32768 ? 32768 : val;
- DEBUG("ringbuffer_size=%d\n", jd->ringbuffer_size);
- } else {
- FATAL("%s is not a number; ringbuf_size=%d\n",
- bp->value, jd->ringbuffer_size);
- }
- }
-
- if ( (bp = getBlockParam(param, "name"))
- && (strcmp(bp->value, "mpd") != 0) ) {
- jd->name = xstrdup(bp->value);
- DEBUG("name=%s\n", jd->name);
- }
-
- return jd;
-}
-
-static int
-mpd_jack_test_default_device(void)
-{
- return 0;
-}
-
-static int
-mpd_jack_connect(struct jack_data *jd, struct audio_format *audio_format)
-{
- const char **jports;
- char *port_name;
-
- jd->audio_format = audio_format;
-
- if ( (jd->client = jack_client_new(jd->name)) == NULL ) {
- ERROR("jack server not running?\n");
- return -1;
- }
-
- jack_set_error_function(mpd_jack_error);
- jack_set_process_callback(jd->client, mpd_jack_process, jd);
- jack_set_sample_rate_callback(jd->client, mpd_jack_srate, jd);
- jack_on_shutdown(jd->client, mpd_jack_shutdown, jd);
-
- if ( jack_activate(jd->client) ) {
- ERROR("cannot activate client\n");
- 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");
- 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");
- return -1;
- }
-
- /* hay que buscar que hay */
- if (!jd->output_ports[1] &&
- (jports = jack_get_ports(jd->client, NULL, NULL,
- JackPortIsPhysical | JackPortIsInput))) {
- jd->output_ports[0] = jports[0];
- jd->output_ports[1] = jports[1] ? jports[1] : jports[0];
- DEBUG("output_ports: %s %s\n",
- jd->output_ports[0], jd->output_ports[1]);
- free(jports);
- }
-
- if ( jd->output_ports[1] ) {
- jd->ringbuffer[0] = jack_ringbuffer_create(jd->ringbuffer_size);
- jd->ringbuffer[1] = jack_ringbuffer_create(jd->ringbuffer_size);
- 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(jd->name)));
-
- sprintf(port_name, "%s:left", jd->name);
- if ( (jack_connect(jd->client, port_name,
- jd->output_ports[0])) != 0 ) {
- ERROR("%s is not a valid Jack Client / Port\n",
- jd->output_ports[0]);
- free(port_name);
- return -1;
- }
- sprintf(port_name, "%s:right", jd->name);
- if ( (jack_connect(jd->client, port_name,
- jd->output_ports[1])) != 0 ) {
- ERROR("%s is not a valid Jack Client / Port\n",
- jd->output_ports[1]);
- free(port_name);
- return -1;
- }
- free(port_name);
- }
-
- return 1;
-}
-
-static int
-mpd_jack_open(void *data, struct audio_format *audio_format)
-{
- struct jack_data *jd = data;
-
- assert(jd != NULL);
-
- if (jd->client == NULL && mpd_jack_connect(jd, audio_format) < 0) {
- mpd_jack_client_free(jd);
- return -1;
- }
-
- set_audioformat(jd, audio_format);
-
- return 0;
-}
-
-static void
-mpd_jack_close(mpd_unused void *data)
-{
- /*mpd_jack_finish(audioOutput);*/
-}
-
-static void
-mpd_jack_cancel (mpd_unused void *data)
-{
-}
-
-static inline jack_default_audio_sample_t
-sample_16_to_jack(int16_t sample)
-{
- return sample / (jack_default_audio_sample_t)(1 << (16 - 1));
-}
-
-static void
-mpd_jack_write_samples_16(struct jack_data *jd, const int16_t *src,
- unsigned num_samples)
-{
- jack_default_audio_sample_t sample;
-
- while (num_samples-- > 0) {
- sample = sample_16_to_jack(*src++);
- jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
- sizeof(sample));
-
- sample = sample_16_to_jack(*src++);
- jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
- sizeof(sample));
- }
-}
-
-static inline jack_default_audio_sample_t
-sample_24_to_jack(int32_t sample)
-{
- return sample / (jack_default_audio_sample_t)(1 << (24 - 1));
-}
-
-static void
-mpd_jack_write_samples_24(struct jack_data *jd, const int32_t *src,
- unsigned num_samples)
-{
- jack_default_audio_sample_t sample;
-
- while (num_samples-- > 0) {
- sample = sample_24_to_jack(*src++);
- jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
- sizeof(sample));
-
- sample = sample_24_to_jack(*src++);
- jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
- sizeof(sample));
- }
-}
-
-static void
-mpd_jack_write_samples(struct jack_data *jd, const void *src,
- unsigned num_samples)
-{
- switch (jd->audio_format->bits) {
- case 16:
- mpd_jack_write_samples_16(jd, (const int16_t*)src,
- num_samples);
- break;
-
- case 24:
- mpd_jack_write_samples_24(jd, (const int32_t*)src,
- num_samples);
- break;
-
- default:
- assert(false);
- }
-}
-
-static int
-mpd_jack_play(void *data, const char *buff, size_t size)
-{
- struct jack_data *jd = data;
- const size_t frame_size = audio_format_frame_size(jd->audio_format);
- size_t space, space1;
-
- if (jd->shutdown) {
- ERROR("Refusing to play, because there is no client thread.\n");
- mpd_jack_client_free(jd);
- audio_output_closed(jd->ao);
- return 0;
- }
-
- assert(size % frame_size == 0);
- size /= frame_size;
- while (size > 0 && !jd->shutdown) {
- space = jack_ringbuffer_write_space(jd->ringbuffer[0]);
- space1 = jack_ringbuffer_write_space(jd->ringbuffer[1]);
- if (space > space1)
- /* send data symmetrically */
- space = space1;
-
- space /= sample_size;
- if (space > 0) {
- if (space > size)
- space = size;
-
- mpd_jack_write_samples(jd, buff, space);
-
- buff += space * frame_size;
- size -= space;
- } else {
- /* XXX do something more intelligent to
- synchronize */
- my_usleep(10000);
- }
-
- }
-
- return 0;
-}
-
-const struct audio_output_plugin jackPlugin = {
- .name = "jack",
- .test_default_device = mpd_jack_test_default_device,
- .init = mpd_jack_init,
- .finish = mpd_jack_finish,
- .open = mpd_jack_open,
- .play = mpd_jack_play,
- .cancel = mpd_jack_cancel,
- .close = mpd_jack_close,
-};
-
-#else /* HAVE JACK */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(jackPlugin)
-
-#endif /* HAVE_JACK */
diff --git a/src/audioOutputs/audioOutput_mvp.c b/src/audioOutputs/audioOutput_mvp.c
deleted file mode 100644
index 70dd25f9d..000000000
--- a/src/audioOutputs/audioOutput_mvp.c
+++ /dev/null
@@ -1,280 +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 "../output_api.h"
-
-#ifdef HAVE_MVP
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-
-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 {
- struct audio_output *audio_output;
- struct audio_format audio_format;
- int fd;
-} MvpData;
-
-static unsigned 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 const unsigned numfrequencies =
- sizeof(pcmfrequencies) / sizeof(pcmfrequencies[0]);
-
-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 void *mvp_initDriver(mpd_unused struct audio_output *audio_output,
- mpd_unused const struct audio_format *audio_format,
- mpd_unused ConfigParam *param)
-{
- MvpData *md = xmalloc(sizeof(MvpData));
- md->audio_output = audio_output;
- md->fd = -1;
-
- return md;
-}
-
-static void mvp_finishDriver(void *data)
-{
- MvpData *md = data;
- free(md);
-}
-
-static int mvp_setPcmParams(MvpData * md, unsigned long rate, int channels,
- int big_endian, unsigned bits)
-{
- unsigned iloop;
- unsigned 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(void *data, struct audio_format *audioFormat)
-{
- MvpData *md = data;
- long long int stc = 0;
- 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->sample_rate, audioFormat->channels,
- 0, audioFormat->bits);
-#else
- mvp_setPcmParams(md, audioFormat->sample_rate, audioFormat->channels,
- 1, audioFormat->bits);
-#endif
- md->audio_format = *audioFormat;
- return 0;
-}
-
-static void mvp_closeDevice(void *data)
-{
- MvpData *md = data;
- if (md->fd >= 0)
- close(md->fd);
- md->fd = -1;
-}
-
-static void mvp_dropBufferedAudio(void *data)
-{
- MvpData *md = data;
- if (md->fd >= 0) {
- ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
- close(md->fd);
- md->fd = -1;
- audio_output_closed(md->audio_output);
- }
-}
-
-static int mvp_playAudio(void *data, const char *playChunk, size_t size)
-{
- MvpData *md = data;
- ssize_t ret;
-
- /* reopen the device since it was closed by dropBufferedAudio */
- if (md->fd < 0)
- mvp_openDevice(md, &md->audio_format);
-
- 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(md);
- return -1;
- }
- playChunk += ret;
- size -= ret;
- }
- return 0;
-}
-
-const struct audio_output_plugin mvpPlugin = {
- .name = "mvp",
- .test_default_device = mvp_testDefault,
- .init = mvp_initDriver,
- .finish = mvp_finishDriver,
- .open = mvp_openDevice,
- .play = mvp_playAudio,
- .cancel = mvp_dropBufferedAudio,
- .close = mvp_closeDevice,
-};
-
-#else /* HAVE_MVP */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(mvpPlugin)
-#endif /* HAVE_MVP */
diff --git a/src/audioOutputs/audioOutput_null.c b/src/audioOutputs/audioOutput_null.c
deleted file mode 100644
index ff3a9833c..000000000
--- a/src/audioOutputs/audioOutput_null.c
+++ /dev/null
@@ -1,85 +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 "../output_api.h"
-#include "../timer.h"
-#include "../utils.h"
-
-struct null_data {
- Timer *timer;
-};
-
-static void *null_initDriver(mpd_unused struct audio_output *audioOutput,
- mpd_unused const struct audio_format *audio_format,
- mpd_unused ConfigParam *param)
-{
- struct null_data *nd = xmalloc(sizeof(*nd));
- nd->timer = NULL;
- return nd;
-}
-
-static int null_openDevice(void *data,
- struct audio_format *audio_format)
-{
- struct null_data *nd = data;
-
- nd->timer = timer_new(audio_format);
- return 0;
-}
-
-static void null_closeDevice(void *data)
-{
- struct null_data *nd = data;
-
- if (nd->timer != NULL) {
- timer_free(nd->timer);
- nd->timer = NULL;
- }
-}
-
-static int null_playAudio(void *data,
- mpd_unused const char *playChunk, size_t size)
-{
- struct null_data *nd = data;
- Timer *timer = nd->timer;
-
- if (!timer->started)
- timer_start(timer);
- else
- timer_sync(timer);
-
- timer_add(timer, size);
-
- return 0;
-}
-
-static void null_dropBufferedAudio(void *data)
-{
- struct null_data *nd = data;
-
- timer_reset(nd->timer);
-}
-
-const struct audio_output_plugin nullPlugin = {
- .name = "null",
- .init = null_initDriver,
- .open = null_openDevice,
- .play = null_playAudio,
- .cancel = null_dropBufferedAudio,
- .close = null_closeDevice,
-};
diff --git a/src/audioOutputs/audioOutput_oss.c b/src/audioOutputs/audioOutput_oss.c
deleted file mode 100644
index 67ea11fe4..000000000
--- a/src/audioOutputs/audioOutput_oss.c
+++ /dev/null
@@ -1,571 +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 <eric@petta-tech.com>
- * and Warren Dukes <warren.dukes@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 "../output_api.h"
-
-#ifdef HAVE_OSS
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-
-#if defined(__OpenBSD__) || defined(__NetBSD__)
-# include <soundcard.h>
-#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
-# include <sys/soundcard.h>
-#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;
- struct audio_format audio_format;
- int bitFormat;
- int *supported[3];
- int numSupported[3];
- int *unsupported[3];
- int numUnsupported[3];
-} OssData;
-
-enum oss_support {
- OSS_SUPPORTED = 1,
- OSS_UNSUPPORTED = 0,
- OSS_UNKNOWN = -1,
-};
-
-enum oss_param {
- OSS_RATE = 0,
- OSS_CHANNELS = 1,
- OSS_BITS = 2,
-};
-
-static enum oss_param
-getIndexForParam(unsigned param)
-{
- enum oss_param idx = OSS_RATE;
-
- switch (param) {
- case SNDCTL_DSP_SPEED:
- idx = OSS_RATE;
- break;
- case SNDCTL_DSP_CHANNELS:
- idx = OSS_CHANNELS;
- break;
- case SNDCTL_DSP_SAMPLESIZE:
- idx = OSS_BITS;
- break;
- }
-
- return idx;
-}
-
-static int findSupportedParam(OssData * od, unsigned param, int val)
-{
- int i;
- enum oss_param idx = getIndexForParam(param);
-
- for (i = 0; i < od->numSupported[idx]; i++) {
- if (od->supported[idx][i] == val)
- return 1;
- }
-
- return 0;
-}
-
-static int canConvert(int idx, int val)
-{
- switch (idx) {
- 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, unsigned param, int val)
-{
- int i;
- enum oss_param idx = getIndexForParam(param);
- int ret = -1;
- int least = val;
- int diff;
-
- for (i = 0; i < od->numSupported[idx]; i++) {
- diff = od->supported[idx][i] - val;
- if (diff < 0)
- diff = -diff;
- if (diff < least) {
- if (!canConvert(idx, od->supported[idx][i])) {
- continue;
- }
- least = diff;
- ret = od->supported[idx][i];
- }
- }
-
- return ret;
-}
-
-static int findUnsupportedParam(OssData * od, unsigned param, int val)
-{
- int i;
- enum oss_param idx = getIndexForParam(param);
-
- for (i = 0; i < od->numUnsupported[idx]; i++) {
- if (od->unsupported[idx][i] == val)
- return 1;
- }
-
- return 0;
-}
-
-static void addSupportedParam(OssData * od, unsigned param, int val)
-{
- enum oss_param idx = getIndexForParam(param);
-
- od->numSupported[idx]++;
- od->supported[idx] = xrealloc(od->supported[idx],
- od->numSupported[idx] * sizeof(int));
- od->supported[idx][od->numSupported[idx] - 1] = val;
-}
-
-static void addUnsupportedParam(OssData * od, unsigned param, int val)
-{
- enum oss_param idx = getIndexForParam(param);
-
- od->numUnsupported[idx]++;
- od->unsupported[idx] = xrealloc(od->unsupported[idx],
- od->numUnsupported[idx] *
- sizeof(int));
- od->unsupported[idx][od->numUnsupported[idx] - 1] = val;
-}
-
-static void removeSupportedParam(OssData * od, unsigned param, int val)
-{
- int i;
- int j = 0;
- enum oss_param idx = getIndexForParam(param);
-
- for (i = 0; i < od->numSupported[idx] - 1; i++) {
- if (od->supported[idx][i] == val)
- j = 1;
- od->supported[idx][i] = od->supported[idx][i + j];
- }
-
- od->numSupported[idx]--;
- od->supported[idx] = xrealloc(od->supported[idx],
- od->numSupported[idx] * sizeof(int));
-}
-
-static void removeUnsupportedParam(OssData * od, unsigned param, int val)
-{
- int i;
- int j = 0;
- enum oss_param idx = getIndexForParam(param);
-
- for (i = 0; i < od->numUnsupported[idx] - 1; i++) {
- if (od->unsupported[idx][i] == val)
- j = 1;
- od->unsupported[idx][i] = od->unsupported[idx][i + j];
- }
-
- od->numUnsupported[idx]--;
- od->unsupported[idx] = xrealloc(od->unsupported[idx],
- od->numUnsupported[idx] *
- sizeof(int));
-}
-
-static enum oss_support
-isSupportedParam(OssData * od, unsigned 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, unsigned param, int val)
-{
- enum oss_support 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, unsigned param, int val)
-{
- enum oss_support 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 void *oss_open_default(ConfigParam *param)
-{
- 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) {
- OssData *od = newOssData();
- od->device = default_devices[i];
- return od;
- }
- }
-
- 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\n", dev, strerror(err[i]));
- }
- }
- exit(EXIT_FAILURE);
- return NULL; /* some compilers can be dumb... */
-}
-
-static void *oss_initDriver(mpd_unused struct audio_output *audioOutput,
- mpd_unused const struct audio_format *audio_format,
- ConfigParam * param)
-{
- if (param) {
- BlockParam *bp = getBlockParam(param, "device");
- if (bp) {
- OssData *od = newOssData();
- od->device = bp->value;
- return od;
- }
- }
- return oss_open_default(param);
-}
-
-static void oss_finishDriver(void *data)
-{
- OssData *od = data;
-
- freeOssData(od);
-}
-
-static int setParam(OssData * od, unsigned param, int *value)
-{
- int val = *value;
- int copy;
- enum oss_support 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, &copy)) {
- 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(OssData *od)
-{
- int tmp;
-
- if ((od->fd = open(od->device, O_WRONLY)) < 0) {
- ERROR("Error opening OSS device \"%s\": %s\n", od->device,
- strerror(errno));
- goto fail;
- }
-
- tmp = od->audio_format.channels;
- if (setParam(od, SNDCTL_DSP_CHANNELS, &tmp)) {
- ERROR("OSS device \"%s\" does not support %u channels: %s\n",
- od->device, od->audio_format.channels, strerror(errno));
- goto fail;
- }
- od->audio_format.channels = tmp;
-
- tmp = od->audio_format.sample_rate;
- if (setParam(od, SNDCTL_DSP_SPEED, &tmp)) {
- ERROR("OSS device \"%s\" does not support %u Hz audio: %s\n",
- od->device, od->audio_format.sample_rate,
- strerror(errno));
- goto fail;
- }
- od->audio_format.sample_rate = tmp;
-
- switch (od->audio_format.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 %u bit audio: %s\n",
- od->device, tmp, strerror(errno));
- goto fail;
- }
-
- return 0;
-
-fail:
- oss_close(od);
- return -1;
-}
-
-static int oss_openDevice(void *data,
- struct audio_format *audioFormat)
-{
- int ret;
- OssData *od = data;
-
- od->audio_format = *audioFormat;
-
- if ((ret = oss_open(od)) < 0)
- return ret;
-
- *audioFormat = od->audio_format;
-
- DEBUG("oss device \"%s\" will be playing %u bit %u channel audio at "
- "%u Hz\n", od->device,
- od->audio_format.bits, od->audio_format.channels,
- od->audio_format.sample_rate);
-
- return ret;
-}
-
-static void oss_closeDevice(void *data)
-{
- OssData *od = data;
-
- oss_close(od);
-}
-
-static void oss_dropBufferedAudio(void *data)
-{
- OssData *od = data;
-
- if (od->fd >= 0) {
- ioctl(od->fd, SNDCTL_DSP_RESET, 0);
- oss_close(od);
- }
-}
-
-static int oss_playAudio(void *data,
- const char *playChunk, size_t size)
-{
- OssData *od = data;
- ssize_t ret;
-
- /* reopen the device since it was closed by dropBufferedAudio */
- if (od->fd < 0 && oss_open(od) < 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(od);
- return -1;
- }
- playChunk += ret;
- size -= ret;
- }
-
- return 0;
-}
-
-const struct audio_output_plugin ossPlugin = {
- .name = "oss",
- .test_default_device = oss_testDefault,
- .init = oss_initDriver,
- .finish = oss_finishDriver,
- .open = oss_openDevice,
- .play = oss_playAudio,
- .cancel = oss_dropBufferedAudio,
- .close = oss_closeDevice,
-};
-
-#else /* HAVE OSS */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(ossPlugin)
-#endif /* HAVE_OSS */
diff --git a/src/audioOutputs/audioOutput_osx.c b/src/audioOutputs/audioOutput_osx.c
deleted file mode 100644
index 65060cc8c..000000000
--- a/src/audioOutputs/audioOutput_osx.c
+++ /dev/null
@@ -1,368 +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 "../output_api.h"
-
-#ifdef HAVE_OSX
-
-#include <AudioUnit/AudioUnit.h>
-
-#include "../utils.h"
-#include "../log.h"
-
-typedef struct _OsxData {
- AudioUnit au;
- pthread_mutex_t mutex;
- pthread_cond_t condition;
- char *buffer;
- size_t bufferSize;
- size_t pos;
- size_t 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(struct audio_output *audioOutput,
- mpd_unused const struct audio_format *audio_format,
- 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(struct audio_output *audioOutput)
-{
- OsxData *od = (OsxData *) audioOutput->data;
- freeOsxData(od);
-}
-
-static void osx_dropBufferedAudio(struct audio_output *audioOutput)
-{
- OsxData *od = (OsxData *) audioOutput->data;
-
- pthread_mutex_lock(&od->mutex);
- od->len = 0;
- pthread_mutex_unlock(&od->mutex);
-}
-
-static void osx_closeDevice(struct audio_output *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);
-}
-
-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];
- size_t bufferSize = buffer->mDataByteSize;
- size_t 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) {
- size_t 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(struct audio_output *audioOutput,
- struct audio_format *audioFormat)
-{
- OsxData *od = (OsxData *) audioOutput->data;
- ComponentDescription desc;
- Component comp;
- AURenderCallbackStruct callback;
- 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->sample_rate;
- streamDesc.mFormatID = kAudioFormatLinearPCM;
- streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
-#ifdef WORDS_BIGENDIAN
- streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
-#endif
-
- streamDesc.mBytesPerPacket = audio_format_frame_size(audioFormat);
- 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->sample_rate) *
- audio_format_frame_size(audioFormat);
- od->buffer = xrealloc(od->buffer, od->bufferSize);
-
- od->pos = 0;
- od->len = 0;
-
- return 0;
-}
-
-static int osx_play(struct audio_output *audioOutput,
- const char *playChunk, size_t size)
-{
- OsxData *od = (OsxData *) audioOutput->data;
- size_t bytesToCopy;
- size_t 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) {
- size_t 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;
-}
-
-const struct audio_output_plugin osxPlugin = {
- .name = "osx",
- .test_default_device = osx_testDefault,
- .init = osx_initDriver,
- .finish = osx_finishDriver,
- .open = osx_openDevice,
- .play = osx_play,
- .cancel = osx_dropBufferedAudio,
- .close = osx_closeDevice,
-};
-
-#else
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(osxPlugin)
-#endif
diff --git a/src/audioOutputs/audioOutput_pulse.c b/src/audioOutputs/audioOutput_pulse.c
deleted file mode 100644
index b1ce3d049..000000000
--- a/src/audioOutputs/audioOutput_pulse.c
+++ /dev/null
@@ -1,218 +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 "../output_api.h"
-
-#ifdef HAVE_PULSE
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <pulse/simple.h>
-#include <pulse/error.h>
-
-#define MPD_PULSE_NAME "mpd"
-#define CONN_ATTEMPT_INTERVAL 60
-
-typedef struct _PulseData {
- struct audio_output *ao;
-
- 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 void *pulse_initDriver(struct audio_output *ao,
- mpd_unused const struct audio_format *audio_format,
- ConfigParam * param)
-{
- BlockParam *server = NULL;
- BlockParam *sink = NULL;
- PulseData *pd;
-
- if (param) {
- server = getBlockParam(param, "server");
- sink = getBlockParam(param, "sink");
- }
-
- pd = newPulseData();
- pd->ao = ao;
- pd->server = server ? xstrdup(server->value) : NULL;
- pd->sink = sink ? xstrdup(sink->value) : NULL;
-
- return pd;
-}
-
-static void pulse_finishDriver(void *data)
-{
- PulseData *pd = data;
-
- freePulseData(pd);
-}
-
-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(void *data,
- struct audio_format *audioFormat)
-{
- PulseData *pd = data;
- pa_sample_spec ss;
- time_t t;
- int error;
-
- t = time(NULL);
-
- if (pd->connAttempts != 0 &&
- (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL)
- return -1;
-
- pd->connAttempts++;
- pd->lastAttempt = t;
-
- /* MPD doesn't support the other pulseaudio sample formats, so
- we just force MPD to send us everything as 16 bit */
- audioFormat->bits = 16;
-
- ss.format = PA_SAMPLE_S16NE;
- ss.rate = audioFormat->sample_rate;
- ss.channels = audioFormat->channels;
-
- pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
- pd->sink, audio_output_get_name(pd->ao),
- &ss, NULL, NULL,
- &error);
- if (!pd->s) {
- ERROR("Cannot connect to server in PulseAudio output "
- "\"%s\" (attempt %i): %s\n",
- audio_output_get_name(pd->ao),
- pd->connAttempts, pa_strerror(error));
- return -1;
- }
-
- pd->connAttempts = 0;
-
- DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i "
- "channel audio at %i Hz\n",
- audio_output_get_name(pd->ao),
- audioFormat->bits,
- audioFormat->channels, audioFormat->sample_rate);
-
- return 0;
-}
-
-static void pulse_dropBufferedAudio(void *data)
-{
- PulseData *pd = data;
- int error;
-
- if (pa_simple_flush(pd->s, &error) < 0)
- WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
- audio_output_get_name(pd->ao),
- pa_strerror(error));
-}
-
-static void pulse_closeDevice(void *data)
-{
- PulseData *pd = data;
-
- if (pd->s) {
- pa_simple_drain(pd->s, NULL);
- pa_simple_free(pd->s);
- }
-}
-
-static int pulse_playAudio(void *data,
- const char *playChunk, size_t size)
-{
- PulseData *pd = data;
- int error;
-
- if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
- ERROR("PulseAudio output \"%s\" disconnecting due to write "
- "error: %s\n",
- audio_output_get_name(pd->ao),
- pa_strerror(error));
- pulse_closeDevice(pd);
- return -1;
- }
-
- return 0;
-}
-
-const struct audio_output_plugin pulsePlugin = {
- .name = "pulse",
- .test_default_device = pulse_testDefault,
- .init = pulse_initDriver,
- .finish = pulse_finishDriver,
- .open = pulse_openDevice,
- .play = pulse_playAudio,
- .cancel = pulse_dropBufferedAudio,
- .close = pulse_closeDevice,
-};
-
-#else /* HAVE_PULSE */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin)
-#endif /* HAVE_PULSE */
diff --git a/src/audioOutputs/audioOutput_shout.c b/src/audioOutputs/audioOutput_shout.c
deleted file mode 100644
index ae31b07ea..000000000
--- a/src/audioOutputs/audioOutput_shout.c
+++ /dev/null
@@ -1,596 +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_shout.h"
-
-#ifdef HAVE_SHOUT
-
-#include "../utils.h"
-
-#include <assert.h>
-
-#define CONN_ATTEMPT_INTERVAL 60
-#define DEFAULT_CONN_TIMEOUT 2
-
-static int shout_init_count;
-
-static const struct shout_encoder_plugin *const shout_encoder_plugins[] = {
-#ifdef HAVE_SHOUT_MP3
- &shout_mp3_encoder,
-#endif
-#ifdef HAVE_SHOUT_OGG
- &shout_ogg_encoder,
-#endif
- NULL
-};
-
-static const struct shout_encoder_plugin *
-shout_encoder_plugin_get(const char *name)
-{
- unsigned i;
-
- for (i = 0; shout_encoder_plugins[i] != NULL; ++i)
- if (strcmp(shout_encoder_plugins[i]->name, name) == 0)
- return shout_encoder_plugins[i];
-
- return NULL;
-}
-
-static struct shout_data *new_shout_data(void)
-{
- struct shout_data *ret = xmalloc(sizeof(*ret));
-
- ret->shout_conn = shout_new();
- ret->shout_meta = shout_metadata_new();
- ret->opened = 0;
- ret->tag = NULL;
- ret->tag_to_send = 0;
- ret->bitrate = -1;
- ret->quality = -2.0;
- ret->timeout = DEFAULT_CONN_TIMEOUT;
- ret->conn_attempts = 0;
- ret->last_attempt = 0;
- ret->timer = NULL;
- ret->buf.len = 0;
-
- return ret;
-}
-
-static void free_shout_data(struct shout_data *sd)
-{
- if (sd->shout_meta)
- shout_metadata_free(sd->shout_meta);
- if (sd->shout_conn)
- shout_free(sd->shout_conn);
- if (sd->tag)
- tag_free(sd->tag);
- if (sd->timer)
- timer_free(sd->timer);
-
- free(sd);
-}
-
-#define check_block_param(name) { \
- block_param = getBlockParam(param, name); \
- if (!block_param) { \
- FATAL("no \"%s\" defined for shout device defined at line " \
- "%i\n", name, param->line); \
- } \
- }
-
-static void *my_shout_init_driver(struct audio_output *audio_output,
- const struct audio_format *audio_format,
- ConfigParam *param)
-{
- struct shout_data *sd;
- char *test;
- int port;
- char *host;
- char *mount;
- char *passwd;
- const char *encoding;
- unsigned protocol;
- const char *user;
- char *name;
- BlockParam *block_param;
- int public;
-
- sd = new_shout_data();
- sd->audio_output = audio_output;
-
- if (shout_init_count == 0)
- shout_init();
-
- shout_init_count++;
-
- check_block_param("host");
- host = block_param->value;
-
- check_block_param("mount");
- mount = block_param->value;
-
- check_block_param("port");
-
- port = strtol(block_param->value, &test, 10);
-
- if (*test != '\0' || port <= 0) {
- FATAL("shout port \"%s\" is not a positive integer, line %i\n",
- block_param->value, block_param->line);
- }
-
- check_block_param("password");
- passwd = block_param->value;
-
- check_block_param("name");
- name = block_param->value;
-
- public = getBoolBlockParam(param, "public", 1);
- if (public == CONF_BOOL_UNSET)
- public = 0;
-
- block_param = getBlockParam(param, "user");
- if (block_param)
- user = block_param->value;
- else
- user = "source";
-
- block_param = getBlockParam(param, "quality");
-
- if (block_param) {
- int line = block_param->line;
-
- sd->quality = strtod(block_param->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", block_param->value,
- block_param->line);
- }
-
- block_param = getBlockParam(param, "bitrate");
-
- if (block_param) {
- FATAL("quality (line %i) and bitrate (line %i) are "
- "both defined for shout output\n", line,
- block_param->line);
- }
- } else {
- block_param = getBlockParam(param, "bitrate");
-
- if (!block_param) {
- FATAL("neither bitrate nor quality defined for shout "
- "output at line %i\n", param->line);
- }
-
- sd->bitrate = strtol(block_param->value, &test, 10);
-
- if (*test != '\0' || sd->bitrate <= 0) {
- FATAL("bitrate at line %i should be a positive integer "
- "\n", block_param->line);
- }
- }
-
- check_block_param("format");
-
- assert(audio_format != NULL);
- sd->audio_format = *audio_format;
-
- block_param = getBlockParam(param, "encoding");
- if (block_param) {
- if (0 == strcmp(block_param->value, "mp3"))
- encoding = block_param->value;
- else if (0 == strcmp(block_param->value, "ogg"))
- encoding = block_param->value;
- else
- FATAL("shout encoding \"%s\" is not \"ogg\" or "
- "\"mp3\", line %i\n", block_param->value,
- block_param->line);
- } else {
- encoding = "ogg";
- }
-
- sd->encoder = shout_encoder_plugin_get(encoding);
- if (sd->encoder == NULL)
- FATAL("couldn't find shout encoder plugin for \"%s\" "
- "at line %i\n", encoding, block_param->line);
-
- check_block_param("protocol");
-
- block_param = getBlockParam(param, "protocol");
- if (block_param) {
- if (0 == strcmp(block_param->value, "shoutcast") &&
- 0 != strcmp(encoding, "mp3"))
- FATAL("you cannot stream \"%s\" to shoutcast, use mp3\n",
- encoding);
- else if (0 == strcmp(block_param->value, "shoutcast"))
- protocol = SHOUT_PROTOCOL_ICY;
- else if (0 == strcmp(block_param->value, "icecast1"))
- protocol = SHOUT_PROTOCOL_XAUDIOCAST;
- else if (0 == strcmp(block_param->value, "icecast2"))
- protocol = SHOUT_PROTOCOL_HTTP;
- else
- FATAL("shout protocol \"%s\" is not \"shoutcast\" or "
- "\"icecast1\"or "
- "\"icecast2\", line %i\n", block_param->value,
- block_param->line);
- } else {
- protocol = SHOUT_PROTOCOL_HTTP;
- }
-
- if (shout_set_host(sd->shout_conn, host) != SHOUTERR_SUCCESS ||
- shout_set_port(sd->shout_conn, port) != SHOUTERR_SUCCESS ||
- shout_set_password(sd->shout_conn, passwd) != SHOUTERR_SUCCESS ||
- shout_set_mount(sd->shout_conn, mount) != SHOUTERR_SUCCESS ||
- shout_set_name(sd->shout_conn, name) != SHOUTERR_SUCCESS ||
- shout_set_user(sd->shout_conn, user) != SHOUTERR_SUCCESS ||
- shout_set_public(sd->shout_conn, public) != SHOUTERR_SUCCESS ||
- shout_set_nonblocking(sd->shout_conn, 1) != SHOUTERR_SUCCESS ||
- shout_set_format(sd->shout_conn, sd->encoder->shout_format)
- != SHOUTERR_SUCCESS ||
- shout_set_protocol(sd->shout_conn, protocol) != SHOUTERR_SUCCESS ||
- shout_set_agent(sd->shout_conn, "MPD") != SHOUTERR_SUCCESS) {
- FATAL("error configuring shout defined at line %i: %s\n",
- param->line, shout_get_error(sd->shout_conn));
- }
-
- /* optional paramters */
- block_param = getBlockParam(param, "timeout");
- if (block_param) {
- sd->timeout = (int)strtol(block_param->value, &test, 10);
- if (*test != '\0' || sd->timeout <= 0) {
- FATAL("shout timeout is not a positive integer, "
- "line %i\n", block_param->line);
- }
- }
-
- block_param = getBlockParam(param, "genre");
- if (block_param && shout_set_genre(sd->shout_conn, block_param->value)) {
- FATAL("error configuring shout defined at line %i: %s\n",
- param->line, shout_get_error(sd->shout_conn));
- }
-
- block_param = getBlockParam(param, "description");
- if (block_param && shout_set_description(sd->shout_conn,
- block_param->value)) {
- FATAL("error configuring shout defined at line %i: %s\n",
- param->line, shout_get_error(sd->shout_conn));
- }
-
- {
- char temp[11];
- memset(temp, 0, sizeof(temp));
-
- snprintf(temp, sizeof(temp), "%u", sd->audio_format.channels);
- shout_set_audio_info(sd->shout_conn, SHOUT_AI_CHANNELS, temp);
-
- snprintf(temp, sizeof(temp), "%u", sd->audio_format.sample_rate);
-
- shout_set_audio_info(sd->shout_conn, SHOUT_AI_SAMPLERATE, temp);
-
- if (sd->quality >= -1.0) {
- snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
- shout_set_audio_info(sd->shout_conn, SHOUT_AI_QUALITY,
- temp);
- } else {
- snprintf(temp, sizeof(temp), "%d", sd->bitrate);
- shout_set_audio_info(sd->shout_conn, SHOUT_AI_BITRATE,
- temp);
- }
- }
-
- if (sd->encoder->init_func(sd) != 0)
- FATAL("shout: encoder plugin '%s' failed to initialize\n",
- sd->encoder->name);
-
- return sd;
-}
-
-static int handle_shout_error(struct shout_data *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->shout_conn),
- shout_get_port(sd->shout_conn),
- shout_get_error(sd->shout_conn));
- sd->shout_error = 1;
- return -1;
- default:
- ERROR("shout: connection to %s:%i error: %s\n",
- shout_get_host(sd->shout_conn),
- shout_get_port(sd->shout_conn),
- shout_get_error(sd->shout_conn));
- sd->shout_error = 1;
- return -1;
- }
-
- return 0;
-}
-
-static int write_page(struct shout_data *sd)
-{
- int err;
-
- if (sd->buf.len == 0)
- return 0;
-
- shout_sync(sd->shout_conn);
- err = shout_send(sd->shout_conn, sd->buf.data, sd->buf.len);
- if (handle_shout_error(sd, err) < 0)
- return -1;
- sd->buf.len = 0;
-
- return 0;
-}
-
-static void close_shout_conn(struct shout_data * sd)
-{
- if (sd->opened) {
- if (sd->encoder->clear_encoder_func(sd))
- write_page(sd);
- }
-
- if (shout_get_connected(sd->shout_conn) != SHOUTERR_UNCONNECTED &&
- shout_close(sd->shout_conn) != SHOUTERR_SUCCESS) {
- ERROR("problem closing connection to shout server: %s\n",
- shout_get_error(sd->shout_conn));
- }
-
- sd->opened = 0;
-}
-
-static void my_shout_finish_driver(void *data)
-{
- struct shout_data *sd = (struct shout_data *)data;
-
- close_shout_conn(sd);
-
- sd->encoder->finish_func(sd);
- free_shout_data(sd);
-
- shout_init_count--;
-
- if (shout_init_count == 0)
- shout_shutdown();
-}
-
-static void my_shout_drop_buffered_audio(void *data)
-{
- struct shout_data *sd = (struct shout_data *)data;
- timer_reset(sd->timer);
-
- /* needs to be implemented for shout */
-}
-
-static void my_shout_close_device(void *data)
-{
- struct shout_data *sd = (struct shout_data *)data;
-
- close_shout_conn(sd);
-
- if (sd->timer) {
- timer_free(sd->timer);
- sd->timer = NULL;
- }
-}
-
-static int shout_connect(struct shout_data *sd)
-{
- time_t t = time(NULL);
- int state = shout_get_connected(sd->shout_conn);
-
- /* already connected */
- if (state == SHOUTERR_CONNECTED)
- return 0;
-
- /* waiting to connect */
- if (state == SHOUTERR_BUSY && sd->conn_attempts != 0) {
- /* timeout waiting to connect */
- if ((t - sd->last_attempt) > sd->timeout) {
- ERROR("timeout connecting to shout server %s:%i "
- "(attempt %i)\n",
- shout_get_host(sd->shout_conn),
- shout_get_port(sd->shout_conn),
- sd->conn_attempts);
- return -1;
- }
-
- return 1;
- }
-
- /* we're in some funky state, so just reset it to unconnected */
- if (state != SHOUTERR_UNCONNECTED)
- shout_close(sd->shout_conn);
-
- /* throttle new connection attempts */
- if (sd->conn_attempts != 0 &&
- (t - sd->last_attempt) <= CONN_ATTEMPT_INTERVAL) {
- return -1;
- }
-
- /* initiate a new connection */
-
- sd->conn_attempts++;
- sd->last_attempt = t;
-
- state = shout_open(sd->shout_conn);
- switch (state) {
- case SHOUTERR_SUCCESS:
- case SHOUTERR_CONNECTED:
- return 0;
- case SHOUTERR_BUSY:
- return 1;
- default:
- ERROR("problem opening connection to shout server %s:%i "
- "(attempt %i): %s\n",
- shout_get_host(sd->shout_conn),
- shout_get_port(sd->shout_conn),
- sd->conn_attempts, shout_get_error(sd->shout_conn));
- return -1;
- }
-}
-
-static int open_shout_conn(void *data)
-{
- struct shout_data *sd = (struct shout_data *)data;
- int status;
-
- status = shout_connect(sd);
- if (status != 0)
- return status;
-
- if (sd->encoder->init_encoder_func(sd) < 0) {
- shout_close(sd->shout_conn);
- return -1;
- }
-
- write_page(sd);
-
- sd->shout_error = 0;
- sd->opened = 1;
- sd->tag_to_send = 1;
- sd->conn_attempts = 0;
-
- return 0;
-}
-
-static int my_shout_open_device(void *data,
- struct audio_format *audio_format)
-{
- struct shout_data *sd = (struct shout_data *)data;
-
- if (!sd->opened && open_shout_conn(sd) < 0)
- return -1;
-
- if (sd->timer)
- timer_free(sd->timer);
-
- sd->timer = timer_new(audio_format);
-
- return 0;
-}
-
-static void send_metadata(struct shout_data * sd)
-{
- static const int size = 1024;
- char song[size];
-
- if (!sd->opened || !sd->tag)
- return;
-
- if (sd->encoder->send_metadata_func(sd, song, size)) {
- shout_metadata_add(sd->shout_meta, "song", song);
- if (SHOUTERR_SUCCESS != shout_set_metadata(sd->shout_conn,
- sd->shout_meta)) {
- ERROR("error setting shout metadata\n");
- return;
- }
- }
-
- sd->tag_to_send = 0;
-}
-
-static int my_shout_play(void *data,
- const char *chunk, size_t size)
-{
- struct shout_data *sd = (struct shout_data *)data;
- int status;
-
- if (!sd->timer->started)
- timer_start(sd->timer);
-
- timer_add(sd->timer, size);
-
- if (sd->opened && sd->tag_to_send)
- send_metadata(sd);
-
- if (!sd->opened) {
- status = open_shout_conn(sd);
- if (status < 0) {
- my_shout_close_device(sd);
- return -1;
- } else if (status > 0) {
- timer_sync(sd->timer);
- return 0;
- }
- }
-
- if (sd->encoder->encode_func(sd, chunk, size)) {
- my_shout_close_device(sd);
- return -1;
- }
-
- if (write_page(sd) < 0) {
- my_shout_close_device(sd);
- return -1;
- }
-
- return 0;
-}
-
-static void my_shout_pause(void *data)
-{
- struct shout_data *sd = (struct shout_data *)data;
- static const char silence[1020];
- int ret;
-
- /* play silence until the player thread sends us a command */
-
- while (sd->opened && !audio_output_is_pending(sd->audio_output)) {
- ret = my_shout_play(data, silence, sizeof(silence));
- if (ret != 0)
- break;
- }
-}
-
-static void my_shout_set_tag(void *data,
- const struct tag *tag)
-{
- struct shout_data *sd = (struct shout_data *)data;
-
- if (sd->tag)
- tag_free(sd->tag);
- sd->tag = NULL;
- sd->tag_to_send = 0;
-
- if (!tag)
- return;
-
- sd->tag = tag_dup(tag);
- sd->tag_to_send = 1;
-}
-
-const struct audio_output_plugin shoutPlugin = {
- .name = "shout",
- .init = my_shout_init_driver,
- .finish = my_shout_finish_driver,
- .open = my_shout_open_device,
- .play = my_shout_play,
- .pause = my_shout_pause,
- .cancel = my_shout_drop_buffered_audio,
- .close = my_shout_close_device,
- .send_tag = my_shout_set_tag,
-};
-
-#else
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(shoutPlugin)
-#endif
diff --git a/src/audioOutputs/audioOutput_shout.h b/src/audioOutputs/audioOutput_shout.h
deleted file mode 100644
index 2cfe68f29..000000000
--- a/src/audioOutputs/audioOutput_shout.h
+++ /dev/null
@@ -1,93 +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
- */
-
-#ifndef AUDIO_OUTPUT_SHOUT_H
-#define AUDIO_OUTPUT_SHOUT_H
-
-#include "../output_api.h"
-
-#ifdef HAVE_SHOUT
-
-#include "../conf.h"
-#include "../timer.h"
-
-#include <shout/shout.h>
-
-struct shout_data;
-
-struct shout_encoder_plugin {
- const char *name;
- unsigned int shout_format;
-
- int (*clear_encoder_func)(struct shout_data *sd);
- int (*encode_func)(struct shout_data *sd,
- const char *chunk, size_t len);
- void (*finish_func)(struct shout_data *sd);
- int (*init_func)(struct shout_data *sd);
- int (*init_encoder_func) (struct shout_data *sd);
- /* Called when there is a new MpdTag to encode into the
- stream. If this function returns non-zero, then the
- resulting song will be passed to the shout server as
- metadata. This allows the Ogg encoder to send metadata via
- Vorbis comments in the stream, while an MP3 encoder can use
- the Shout Server's metadata API. */
- int (*send_metadata_func)(struct shout_data *sd,
- char *song, size_t size);
-};
-
-struct shout_buffer {
- unsigned char data[8192];
- size_t len;
-};
-
-struct shout_data {
- struct audio_output *audio_output;
-
- shout_t *shout_conn;
- shout_metadata_t *shout_meta;
- int shout_error;
-
- const struct shout_encoder_plugin *encoder;
- void *encoder_data;
-
- float quality;
- int bitrate;
-
- int opened;
-
- struct tag *tag;
- int tag_to_send;
-
- int timeout;
- int conn_attempts;
- time_t last_attempt;
-
- Timer *timer;
-
- /* the configured audio format */
- struct audio_format audio_format;
-
- struct shout_buffer buf;
-};
-
-extern const struct shout_encoder_plugin shout_mp3_encoder;
-extern const struct shout_encoder_plugin shout_ogg_encoder;
-
-#endif
-
-#endif
diff --git a/src/audioOutputs/audioOutput_shout_mp3.c b/src/audioOutputs/audioOutput_shout_mp3.c
deleted file mode 100644
index 722079b29..000000000
--- a/src/audioOutputs/audioOutput_shout_mp3.c
+++ /dev/null
@@ -1,189 +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 "../output_api.h"
-
-#ifdef HAVE_SHOUT_MP3
-
-#include "../utils.h"
-#include "audioOutput_shout.h"
-#include <lame/lame.h>
-
-struct lame_data {
- lame_global_flags *gfp;
-};
-
-
-static int shout_mp3_encoder_init(struct shout_data *sd)
-{
- struct lame_data *ld;
-
- if (NULL == (ld = xmalloc(sizeof(*ld))))
- FATAL("error initializing lame encoder data\n");
- sd->encoder_data = ld;
-
- return 0;
-}
-
-static int shout_mp3_encoder_clear_encoder(struct shout_data *sd)
-{
- struct lame_data *ld = (struct lame_data *)sd->encoder_data;
- struct shout_buffer *buf = &sd->buf;
- int ret;
-
- if ((ret = lame_encode_flush(ld->gfp, buf->data + buf->len,
- buf->len)) < 0)
- ERROR("error flushing lame buffers\n");
-
- return (ret > 0);
-}
-
-static void shout_mp3_encoder_finish(struct shout_data *sd)
-{
- struct lame_data *ld = (struct lame_data *)sd->encoder_data;
-
- lame_close(ld->gfp);
- ld->gfp = NULL;
-}
-
-static int shout_mp3_encoder_init_encoder(struct shout_data *sd)
-{
- struct lame_data *ld = (struct lame_data *)sd->encoder_data;
-
- if (NULL == (ld->gfp = lame_init())) {
- ERROR("error initializing lame encoder for shout\n");
- return -1;
- }
-
- if (sd->quality >= -1.0) {
- if (0 != lame_set_VBR(ld->gfp, vbr_rh)) {
- ERROR("error setting lame VBR mode\n");
- return -1;
- }
- if (0 != lame_set_VBR_q(ld->gfp, sd->quality)) {
- ERROR("error setting lame VBR quality\n");
- return -1;
- }
- } else {
- if (0 != lame_set_brate(ld->gfp, sd->bitrate)) {
- ERROR("error setting lame bitrate\n");
- return -1;
- }
- }
-
- if (0 != lame_set_num_channels(ld->gfp,
- sd->audio_format.channels)) {
- ERROR("error setting lame num channels\n");
- return -1;
- }
-
- if (0 != lame_set_in_samplerate(ld->gfp,
- sd->audio_format.sample_rate)) {
- ERROR("error setting lame sample rate\n");
- return -1;
- }
-
- if (0 > lame_init_params(ld->gfp))
- FATAL("error initializing lame params\n");
-
- return 0;
-}
-
-static int shout_mp3_encoder_send_metadata(struct shout_data *sd,
- char * song, size_t size)
-{
- char artist[size];
- char title[size];
- int i;
- struct tag *tag = sd->tag;
-
- strncpy(artist, "", size);
- strncpy(title, "", size);
-
- for (i = 0; i < tag->numOfItems; i++) {
- switch (tag->items[i]->type) {
- case TAG_ITEM_ARTIST:
- strncpy(artist, tag->items[i]->value, size);
- break;
- case TAG_ITEM_TITLE:
- strncpy(title, tag->items[i]->value, size);
- break;
-
- default:
- break;
- }
- }
- snprintf(song, size, "%s - %s", title, artist);
-
- return 1;
-}
-
-static int shout_mp3_encoder_encode(struct shout_data *sd,
- const char * chunk, size_t len)
-{
- unsigned int i;
- int j;
- float (*lamebuf)[2];
- struct shout_buffer *buf = &(sd->buf);
- unsigned int samples;
- int bytes = audio_format_sample_size(&sd->audio_format);
- struct lame_data *ld = (struct lame_data *)sd->encoder_data;
- int bytes_out;
-
- samples = len / (bytes * sd->audio_format.channels);
- /* rough estimate, from lame.h */
- lamebuf = xmalloc(sizeof(float) * (1.25 * samples + 7200));
-
- /* this is for only 16-bit audio */
-
- for (i = 0; i < samples; i++) {
- for (j = 0; j < sd->audio_format.channels; j++) {
- lamebuf[j][i] = *((const int16_t *) chunk);
- chunk += bytes;
- }
- }
-
- bytes_out = lame_encode_buffer_float(ld->gfp, lamebuf[0], lamebuf[1],
- samples, buf->data,
- sizeof(buf->data) - buf->len);
- free(lamebuf);
-
- if (0 > bytes_out) {
- ERROR("error encoding lame buffer for shout\n");
- lame_close(ld->gfp);
- ld->gfp = NULL;
- return -1;
- } else
- buf->len = bytes_out; /* signed to unsigned conversion */
-
- return 0;
-}
-
-const struct shout_encoder_plugin shout_mp3_encoder = {
- "mp3",
- SHOUT_FORMAT_MP3,
-
- shout_mp3_encoder_clear_encoder,
- shout_mp3_encoder_encode,
- shout_mp3_encoder_finish,
- shout_mp3_encoder_init,
- shout_mp3_encoder_init_encoder,
- shout_mp3_encoder_send_metadata,
-};
-
-#endif
diff --git a/src/audioOutputs/audioOutput_shout_ogg.c b/src/audioOutputs/audioOutput_shout_ogg.c
deleted file mode 100644
index 5983b4d89..000000000
--- a/src/audioOutputs/audioOutput_shout_ogg.c
+++ /dev/null
@@ -1,306 +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_shout.h"
-
-#ifdef HAVE_SHOUT_OGG
-
-#include "../utils.h"
-#include <vorbis/vorbisenc.h>
-
-struct ogg_vorbis_data {
- 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;
-};
-
-static void add_tag(struct ogg_vorbis_data *od, const char *name, char *value)
-{
- if (value) {
- union {
- const char *in;
- char *out;
- } u = { .in = name };
- vorbis_comment_add_tag(&od->vc, u.out, value);
- }
-}
-
-static void copy_tag_to_vorbis_comment(struct shout_data *sd)
-{
- struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
- if (sd->tag) {
- int i;
-
- for (i = 0; i < sd->tag->numOfItems; i++) {
- switch (sd->tag->items[i]->type) {
- case TAG_ITEM_ARTIST:
- add_tag(od, "ARTIST", sd->tag->items[i]->value);
- break;
- case TAG_ITEM_ALBUM:
- add_tag(od, "ALBUM", sd->tag->items[i]->value);
- break;
- case TAG_ITEM_TITLE:
- add_tag(od, "TITLE", sd->tag->items[i]->value);
- break;
-
- default:
- break;
- }
- }
- }
-}
-
-static int copy_ogg_buffer_to_shout_buffer(ogg_page *og,
- struct shout_buffer *buf)
-{
- if (sizeof(buf->data) - buf->len >= (size_t)og->header_len) {
- memcpy(buf->data + buf->len,
- og->header, og->header_len);
- buf->len += og->header_len;
- } else {
- ERROR("%s: not enough buffer space!\n", __func__);
- return -1;
- }
-
- if (sizeof(buf->data) - buf->len >= (size_t)og->body_len) {
- memcpy(buf->data + buf->len,
- og->body, og->body_len);
- buf->len += og->body_len;
- } else {
- ERROR("%s: not enough buffer space!\n", __func__);
- return -1;
- }
-
- return 0;
-}
-
-static int flush_ogg_buffer(struct shout_data *sd)
-{
- struct shout_buffer *buf = &sd->buf;
- struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
- int ret = 0;
-
- if (ogg_stream_flush(&od->os, &od->og))
- ret = copy_ogg_buffer_to_shout_buffer(&od->og, buf);
-
- return ret;
-}
-
-static int send_ogg_vorbis_header(struct shout_data *sd)
-{
- struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
- vorbis_analysis_headerout(&od->vd, &od->vc,
- &od->header_main,
- &od->header_comments,
- &od->header_codebooks);
-
- ogg_stream_packetin(&od->os, &od->header_main);
- ogg_stream_packetin(&od->os, &od->header_comments);
- ogg_stream_packetin(&od->os, &od->header_codebooks);
-
- return flush_ogg_buffer(sd);
-}
-
-static void finish_encoder(struct ogg_vorbis_data *od)
-{
- vorbis_analysis_wrote(&od->vd, 0);
-
- while (vorbis_analysis_blockout(&od->vd, &od->vb) == 1) {
- vorbis_analysis(&od->vb, NULL);
- vorbis_bitrate_addblock(&od->vb);
- while (vorbis_bitrate_flushpacket(&od->vd, &od->op)) {
- ogg_stream_packetin(&od->os, &od->op);
- }
- }
-}
-
-static int shout_ogg_encoder_clear_encoder(struct shout_data *sd)
-{
- struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
- int ret;
-
- finish_encoder(od);
- if ((ret = ogg_stream_pageout(&od->os, &od->og)))
- copy_ogg_buffer_to_shout_buffer(&od->og, &sd->buf);
-
- vorbis_comment_clear(&od->vc);
- ogg_stream_clear(&od->os);
- vorbis_block_clear(&od->vb);
- vorbis_dsp_clear(&od->vd);
- vorbis_info_clear(&od->vi);
-
- return ret;
-}
-
-static void shout_ogg_encoder_finish(struct shout_data *sd)
-{
- struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
- if (od) {
- free(od);
- sd->encoder_data = NULL;
- }
-}
-
-static int shout_ogg_encoder_init(struct shout_data *sd)
-{
- struct ogg_vorbis_data *od;
-
- if (NULL == (od = xmalloc(sizeof(*od))))
- FATAL("error initializing ogg vorbis encoder data\n");
- sd->encoder_data = od;
-
- return 0;
-}
-
-static int reinit_encoder(struct shout_data *sd)
-{
- struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
- vorbis_info_init(&od->vi);
-
- if (sd->quality >= -1.0) {
- if (0 != vorbis_encode_init_vbr(&od->vi,
- sd->audio_format.channels,
- sd->audio_format.sample_rate,
- sd->quality * 0.1)) {
- ERROR("error initializing vorbis vbr\n");
- vorbis_info_clear(&od->vi);
- return -1;
- }
- } else {
- if (0 != vorbis_encode_init(&od->vi,
- sd->audio_format.channels,
- sd->audio_format.sample_rate, -1.0,
- sd->bitrate * 1000, -1.0)) {
- ERROR("error initializing vorbis encoder\n");
- vorbis_info_clear(&od->vi);
- return -1;
- }
- }
-
- vorbis_analysis_init(&od->vd, &od->vi);
- vorbis_block_init(&od->vd, &od->vb);
- ogg_stream_init(&od->os, rand());
- vorbis_comment_init(&od->vc);
-
- return 0;
-}
-
-static int shout_ogg_encoder_init_encoder(struct shout_data *sd)
-{
- if (reinit_encoder(sd))
- return -1;
-
- if (send_ogg_vorbis_header(sd)) {
- ERROR("error sending ogg vorbis header for shout\n");
- return -1;
- }
-
- return 0;
-}
-
-static int shout_ogg_encoder_send_metadata(struct shout_data *sd,
- mpd_unused char * song,
- mpd_unused size_t size)
-{
- struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
- shout_ogg_encoder_clear_encoder(sd);
- if (reinit_encoder(sd))
- return 0;
-
- copy_tag_to_vorbis_comment(sd);
-
- vorbis_analysis_headerout(&od->vd, &od->vc,
- &od->header_main,
- &od->header_comments,
- &od->header_codebooks);
-
- ogg_stream_packetin(&od->os, &od->header_main);
- ogg_stream_packetin(&od->os, &od->header_comments);
- ogg_stream_packetin(&od->os, &od->header_codebooks);
-
- flush_ogg_buffer(sd);
-
- return 0;
-}
-
-static int shout_ogg_encoder_encode(struct shout_data *sd,
- const char *chunk, size_t size)
-{
- struct shout_buffer *buf = &sd->buf;
- unsigned int i;
- int j;
- float **vorbbuf;
- unsigned int samples;
- int bytes = audio_format_sample_size(&sd->audio_format);
- struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
- samples = size / (bytes * sd->audio_format.channels);
- vorbbuf = vorbis_analysis_buffer(&od->vd, samples);
-
- /* this is for only 16-bit audio */
-
- for (i = 0; i < samples; i++) {
- for (j = 0; j < sd->audio_format.channels; j++) {
- vorbbuf[j][i] = (*((const int16_t *) chunk)) / 32768.0;
- chunk += bytes;
- }
- }
-
- vorbis_analysis_wrote(&od->vd, samples);
-
- while (1 == vorbis_analysis_blockout(&od->vd, &od->vb)) {
- vorbis_analysis(&od->vb, NULL);
- vorbis_bitrate_addblock(&od->vb);
-
- while (vorbis_bitrate_flushpacket(&od->vd, &od->op)) {
- ogg_stream_packetin(&od->os, &od->op);
- }
- }
-
- if (ogg_stream_pageout(&od->os, &od->og))
- copy_ogg_buffer_to_shout_buffer(&od->og, buf);
-
- return 0;
-}
-
-const struct shout_encoder_plugin shout_ogg_encoder = {
- "ogg",
- SHOUT_FORMAT_VORBIS,
-
- shout_ogg_encoder_clear_encoder,
- shout_ogg_encoder_encode,
- shout_ogg_encoder_finish,
- shout_ogg_encoder_init,
- shout_ogg_encoder_init_encoder,
- shout_ogg_encoder_send_metadata,
-};
-
-#endif
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-01 20:47:14 +0000