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-rw-r--r--src/output/alsa_plugin.c52
-rw-r--r--src/output/openal_plugin.c273
-rw-r--r--src/output/recorder_output_plugin.c214
3 files changed, 537 insertions, 2 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 818c83ca..f271668b 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -183,6 +183,19 @@ get_bitformat(const struct audio_format *af)
return SND_PCM_FORMAT_UNKNOWN;
}
+static snd_pcm_format_t
+byteswap_bitformat(snd_pcm_format_t fmt)
+{
+ switch(fmt) {
+ case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
+ case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
+ case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
+ case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
+ case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;
+ case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
+ default: return SND_PCM_FORMAT_UNKNOWN;
+ }
+}
/**
* Set up the snd_pcm_t object which was opened by the caller. Set up
* the configured settings and the audio format.
@@ -208,7 +221,6 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
configure_hw:
/* configure HW params */
snd_pcm_hw_params_alloca(&hwparams);
-
cmd = "snd_pcm_hw_params_any";
err = snd_pcm_hw_params_any(ad->pcm, hwparams);
if (err < 0)
@@ -236,13 +248,38 @@ configure_hw:
}
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat);
+ if (err == -EINVAL &&
+ byteswap_bitformat(bitformat) != SND_PCM_FORMAT_UNKNOWN) {
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(bitformat));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting %u bit to reverse-endian\n",
+ alsa_device(ad), audio_format->bits);
+ audio_format->reverse_endian = 1;
+ }
+ }
if (err == -EINVAL && (audio_format->bits == 24 ||
audio_format->bits == 16)) {
/* fall back to 32 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S32);
- if (err == 0)
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n",
+ alsa_device(ad), audio_format->bits);
+ audio_format->bits = 32;
+ }
+ }
+ if (err == -EINVAL && (audio_format->bits == 24 ||
+ audio_format->bits == 16)) {
+ /* fall back to 32 bit, let pcm_convert.c do the conversion */
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(SND_PCM_FORMAT_S32));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting %u bit to 32 bit backward-endian\n",
+ alsa_device(ad), audio_format->bits);
audio_format->bits = 32;
+ audio_format->reverse_endian = 1;
+ }
}
if (err == -EINVAL && audio_format->bits != 16) {
@@ -255,6 +292,17 @@ configure_hw:
audio_format->bits = 16;
}
}
+ if (err == -EINVAL && audio_format->bits != 16) {
+ /* fall back to 16 bit, let pcm_convert.c do the conversion */
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(SND_PCM_FORMAT_S16));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting %u bit to 16 bit backward-endian\n",
+ alsa_device(ad), audio_format->bits);
+ audio_format->bits = 16;
+ audio_format->reverse_endian = 1;
+ }
+ }
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
diff --git a/src/output/openal_plugin.c b/src/output/openal_plugin.c
new file mode 100644
index 00000000..92ee82ef
--- /dev/null
+++ b/src/output/openal_plugin.c
@@ -0,0 +1,273 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "../output_api.h"
+#include "../timer.h"
+#include "config.h"
+
+#include <glib.h>
+
+#ifndef HAVE_OSX
+#include <AL/al.h>
+#include <AL/alc.h>
+#else
+#include <OpenAL/al.h>
+#include <OpenAL/alc.h>
+#endif
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "openal"
+
+/* should be enough for buffer size = 2048 */
+#define NUM_BUFFERS 16
+
+struct openal_data {
+ const char *device_name;
+ ALCdevice *device;
+ ALCcontext *context;
+ Timer *timer;
+ ALuint buffers[NUM_BUFFERS];
+ int filled;
+ ALuint source;
+ ALenum format;
+ ALuint frequency;
+};
+
+static inline GQuark
+openal_output_quark(void)
+{
+ return g_quark_from_static_string("openal_output");
+}
+
+static ALenum
+openal_audio_format(struct audio_format *audio_format)
+{
+ /* Only 8 and 16 bit samples are supported */
+ if (audio_format->bits != 16 && audio_format->bits != 8)
+ audio_format->bits = 16;
+
+ switch (audio_format->bits)
+ {
+ case 16:
+ if (audio_format->channels == 2)
+ return AL_FORMAT_STEREO16;
+ if (audio_format->channels == 1)
+ return AL_FORMAT_MONO16;
+ break;
+
+ case 8:
+ if (audio_format->channels == 2)
+ return AL_FORMAT_STEREO8;
+ if (audio_format->channels == 1)
+ return AL_FORMAT_MONO8;
+ break;
+ }
+
+ return 0;
+}
+
+static bool
+openal_setup_context(struct openal_data *od,
+ GError **error)
+{
+ od->device = alcOpenDevice(od->device_name);
+
+ if (od->device == NULL) {
+ g_set_error(error, openal_output_quark(), 0,
+ "Error opening OpenAL device \"%s\"\n",
+ od->device_name);
+ return false;
+ }
+
+ od->context = alcCreateContext(od->device, NULL);
+
+ if (od->context == NULL) {
+ g_set_error(error, openal_output_quark(), 0,
+ "Error creating context for \"%s\"\n",
+ od->device_name);
+ alcCloseDevice(od->device);
+ return false;
+ }
+
+ return true;
+}
+
+static void
+openal_unqueue_buffers(struct openal_data *od)
+{
+ ALint num;
+ ALuint buffer;
+
+ alGetSourcei(od->source, AL_BUFFERS_QUEUED, &num);
+
+ while (num--) {
+ alSourceUnqueueBuffers(od->source, 1, &buffer);
+ }
+}
+
+static void *
+openal_init(G_GNUC_UNUSED const struct audio_format *audio_format,
+ const struct config_param *param,
+ G_GNUC_UNUSED GError **error)
+{
+ const char *device_name = config_get_block_string(param, "device", NULL);
+ struct openal_data *od;
+
+ if (device_name == NULL) {
+ device_name = alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
+ }
+
+ od = g_new(struct openal_data, 1);
+ od->device_name = device_name;
+
+ return od;
+}
+
+static void
+openal_finish(void *data)
+{
+ struct openal_data *od = data;
+
+ g_free(od);
+}
+
+static bool
+openal_open(void *data, struct audio_format *audio_format,
+ GError **error)
+{
+ struct openal_data *od = data;
+
+ od->format = openal_audio_format(audio_format);
+
+ if (!od->format) {
+ g_set_error(error, openal_output_quark(), 0,
+ "Unsupported audio format (%i channels, %i bps)",
+ audio_format->channels,
+ audio_format->bits);
+ return false;
+ }
+
+ if (!openal_setup_context(od, error)) {
+ return false;
+ }
+
+ alcMakeContextCurrent(od->context);
+ alGenBuffers(NUM_BUFFERS, od->buffers);
+
+ if (alGetError() != AL_NO_ERROR) {
+ g_set_error(error, openal_output_quark(), 0,
+ "Failed to generate buffers");
+ return false;
+ }
+
+ alGenSources(1, &od->source);
+
+ if (alGetError() != AL_NO_ERROR) {
+ g_set_error(error, openal_output_quark(), 0,
+ "Failed to generate source");
+ alDeleteBuffers(NUM_BUFFERS, od->buffers);
+ return false;
+ }
+
+ od->filled = 0;
+ od->timer = timer_new(audio_format);
+ od->frequency = audio_format->sample_rate;
+
+ return true;
+}
+
+static void
+openal_close(void *data)
+{
+ struct openal_data *od = data;
+
+ timer_free(od->timer);
+ alcMakeContextCurrent(od->context);
+ alDeleteSources(1, &od->source);
+ alDeleteBuffers(NUM_BUFFERS, od->buffers);
+ alcDestroyContext(od->context);
+ alcCloseDevice(od->device);
+}
+
+static size_t
+openal_play(void *data, const void *chunk, size_t size,
+ G_GNUC_UNUSED GError **error)
+{
+ struct openal_data *od = data;
+ ALuint buffer;
+ ALint num, state;
+
+ if (alcGetCurrentContext() != od->context) {
+ alcMakeContextCurrent(od->context);
+ }
+
+ alGetSourcei(od->source, AL_BUFFERS_PROCESSED, &num);
+
+ if (od->filled < NUM_BUFFERS) {
+ /* fill all buffers */
+ buffer = od->buffers[od->filled];
+ od->filled++;
+ } else {
+ /* wait for processed buffer */
+ while (num < 1) {
+ if (!od->timer->started) {
+ timer_start(od->timer);
+ } else {
+ timer_sync(od->timer);
+ }
+
+ timer_add(od->timer, size);
+
+ alGetSourcei(od->source, AL_BUFFERS_PROCESSED, &num);
+ }
+
+ alSourceUnqueueBuffers(od->source, 1, &buffer);
+ }
+
+ alBufferData(buffer, od->format, chunk, size, od->frequency);
+ alSourceQueueBuffers(od->source, 1, &buffer);
+ alGetSourcei(od->source, AL_SOURCE_STATE, &state);
+
+ if (state != AL_PLAYING) {
+ alSourcePlay(od->source);
+ }
+
+ return size;
+}
+
+static void
+openal_cancel(void *data)
+{
+ struct openal_data *od = data;
+
+ od->filled = 0;
+ alcMakeContextCurrent(od->context);
+ alSourceStop(od->source);
+ openal_unqueue_buffers(od);
+}
+
+const struct audio_output_plugin openal_output_plugin = {
+ .name = "openal",
+ .init = openal_init,
+ .finish = openal_finish,
+ .open = openal_open,
+ .close = openal_close,
+ .play = openal_play,
+ .cancel = openal_cancel,
+};
diff --git a/src/output/recorder_output_plugin.c b/src/output/recorder_output_plugin.c
new file mode 100644
index 00000000..413e5d0d
--- /dev/null
+++ b/src/output/recorder_output_plugin.c
@@ -0,0 +1,214 @@
+/*
+ * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "output_api.h"
+#include "encoder_plugin.h"
+#include "encoder_list.h"
+
+#include <assert.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "recorder"
+
+struct recorder_output {
+ /**
+ * The configured encoder plugin.
+ */
+ struct encoder *encoder;
+
+ /**
+ * The destination file name.
+ */
+ const char *path;
+
+ /**
+ * The destination file descriptor.
+ */
+ int fd;
+
+ /**
+ * The buffer for encoder_read().
+ */
+ char buffer[32768];
+};
+
+/**
+ * The quark used for GError.domain.
+ */
+static inline GQuark
+recorder_output_quark(void)
+{
+ return g_quark_from_static_string("recorder_output");
+}
+
+static void *
+recorder_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
+ const struct config_param *param, GError **error_r)
+{
+ struct recorder_output *recorder = g_new(struct recorder_output, 1);
+ const char *encoder_name;
+ const struct encoder_plugin *encoder_plugin;
+
+ /* read configuration */
+
+ encoder_name = config_get_block_string(param, "encoder", "vorbis");
+ encoder_plugin = encoder_plugin_get(encoder_name);
+ if (encoder_plugin == NULL) {
+ g_set_error(error_r, recorder_output_quark(), 0,
+ "No such encoder: %s", encoder_name);
+ return NULL;
+ }
+
+ recorder->path = config_get_block_string(param, "path", NULL);
+ if (recorder->path == NULL) {
+ g_set_error(error_r, recorder_output_quark(), 0,
+ "'path' not configured");
+ return NULL;
+ }
+
+ /* initialize encoder */
+
+ recorder->encoder = encoder_init(encoder_plugin, param, error_r);
+ if (recorder->encoder == NULL)
+ return NULL;
+
+ return recorder;
+}
+
+static void
+recorder_output_finish(void *data)
+{
+ struct recorder_output *recorder = data;
+
+ encoder_finish(recorder->encoder);
+ g_free(recorder);
+}
+
+/**
+ * Writes pending data from the encoder to the output file.
+ */
+static bool
+recorder_output_encoder_to_file(struct recorder_output *recorder,
+ GError **error_r)
+{
+ size_t size = 0, position, nbytes;
+
+ assert(recorder->fd >= 0);
+
+ /* read from the encoder */
+
+ size = encoder_read(recorder->encoder, recorder->buffer,
+ sizeof(recorder->buffer));
+ if (size == 0)
+ return true;
+
+ /* write everything into the file */
+
+ position = 0;
+ while (true) {
+ nbytes = write(recorder->fd, recorder->buffer + position,
+ size - position);
+ if (nbytes > 0) {
+ position += (size_t)nbytes;
+ if (position >= size)
+ return true;
+ } else if (nbytes == 0) {
+ /* shouldn't happen for files */
+ g_set_error(error_r, recorder_output_quark(), 0,
+ "write() returned 0");
+ return false;
+ } else if (errno != EINTR) {
+ g_set_error(error_r, recorder_output_quark(), 0,
+ "Failed to write to '%s': %s",
+ recorder->path, g_strerror(errno));
+ return false;
+ }
+ }
+}
+
+static bool
+recorder_output_open(void *data, struct audio_format *audio_format,
+ GError **error_r)
+{
+ struct recorder_output *recorder = data;
+ bool success;
+
+ /* create the output file */
+
+ recorder->fd = creat(recorder->path, 0666);
+ if (recorder->fd < 0) {
+ g_set_error(error_r, recorder_output_quark(), 0,
+ "Failed to create '%s': %s",
+ recorder->path, g_strerror(errno));
+ return false;
+ }
+
+ /* open the encoder */
+
+ success = encoder_open(recorder->encoder, audio_format, error_r);
+ if (!success) {
+ close(recorder->fd);
+ unlink(recorder->path);
+ return false;
+ }
+
+ return true;
+}
+
+static void
+recorder_output_close(void *data)
+{
+ struct recorder_output *recorder = data;
+
+ /* flush the encoder and write the rest to the file */
+
+ if (encoder_flush(recorder->encoder, NULL))
+ recorder_output_encoder_to_file(recorder, NULL);
+
+ /* now really close everything */
+
+ encoder_close(recorder->encoder);
+
+ close(recorder->fd);
+}
+
+static size_t
+recorder_output_play(void *data, const void *chunk, size_t size,
+ GError **error_r)
+{
+ struct recorder_output *recorder = data;
+
+ return encoder_write(recorder->encoder, chunk, size, error_r) &&
+ recorder_output_encoder_to_file(recorder, error_r)
+ ? size : 0;
+}
+
+const struct audio_output_plugin recorder_output_plugin = {
+ .name = "recorder",
+ .init = recorder_output_init,
+ .finish = recorder_output_finish,
+ .open = recorder_output_open,
+ .close = recorder_output_close,
+ .play = recorder_output_play,
+};
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