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-rw-r--r--src/decoder/_flac_common.c5
-rw-r--r--src/decoder/audiofile_plugin.c8
-rw-r--r--src/decoder/faad_plugin.c6
-rw-r--r--src/decoder/ffmpeg_plugin.c18
-rw-r--r--src/decoder/flac_plugin.c14
-rw-r--r--src/decoder/mad_plugin.c10
-rw-r--r--src/decoder/mikmod_plugin.c4
-rw-r--r--src/decoder/modplug_plugin.c6
-rw-r--r--src/decoder/mp4ff_plugin.c6
-rw-r--r--src/decoder/mpcdec_plugin.c4
-rw-r--r--src/decoder/sidplay_plugin.cxx4
-rw-r--r--src/decoder/sndfile_decoder_plugin.c244
-rw-r--r--src/decoder/vorbis_plugin.c3
-rw-r--r--src/decoder/wavpack_plugin.c6
-rw-r--r--src/decoder_api.c40
-rw-r--r--src/decoder_control.c52
-rw-r--r--src/decoder_control.h97
-rw-r--r--src/decoder_internal.c27
-rw-r--r--src/decoder_list.c4
-rw-r--r--src/decoder_thread.c47
20 files changed, 520 insertions, 85 deletions
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index e096750f3..09f7269bd 100644
--- a/src/decoder/_flac_common.c
+++ b/src/decoder/_flac_common.c
@@ -201,9 +201,8 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
- data->audio_format.bits = (int8_t)si->bits_per_sample;
- data->audio_format.sample_rate = si->sample_rate;
- data->audio_format.channels = (int8_t)si->channels;
+ audio_format_init(&data->audio_format, si->sample_rate,
+ si->bits_per_sample, si->channels);
data->total_time = ((float)si->total_samples) / (si->sample_rate);
break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
index f66d90dc1..b4959f6c2 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_plugin.c
@@ -136,11 +136,9 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, bits);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- audio_format.bits = (uint8_t)bits;
- audio_format.sample_rate =
- (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
- audio_format.channels =
- (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+
+ audio_format_init(&audio_format, afGetRate(af_fp, AF_DEFAULT_TRACK),
+ bits, afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK));
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
index d0537dd5b..1b8b2b784 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_plugin.c
@@ -262,11 +262,7 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
decoder_buffer_consume(buffer, nbytes);
- *audio_format = (struct audio_format){
- .bits = 16,
- .channels = channels,
- .sample_rate = sample_rate,
- };
+ audio_format_init(audio_format, sample_rate, 16, channels);
return true;
}
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
index abccdf977..e6646f649 100644
--- a/src/decoder/ffmpeg_plugin.c
+++ b/src/decoder/ffmpeg_plugin.c
@@ -267,6 +267,7 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
struct audio_format audio_format;
enum decoder_command cmd;
int total_time;
+ uint8_t bits;
total_time = 0;
@@ -275,13 +276,13 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
}
#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
- audio_format.bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
+ bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
#else
/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
- audio_format.bits = (uint8_t) 16;
+ bits = (uint8_t) 16;
#endif
- audio_format.sample_rate = (unsigned int)codec_context->sample_rate;
- audio_format.channels = codec_context->channels;
+ audio_format_init(&audio_format, codec_context->sample_rate, bits,
+ codec_context->channels);
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
@@ -342,8 +343,9 @@ static void
ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
enum tag_type type, const char *name)
{
- AVMetadataTag *mt = av_metadata_get(m, name, NULL, 0);
- if (mt != NULL)
+ AVMetadataTag *mt = NULL;
+
+ while ((mt = av_metadata_get(m, name, mt, 0)) != NULL)
tag_add_item(tag, type, mt->value);
}
#endif
@@ -351,13 +353,15 @@ ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
static bool ffmpeg_tag_internal(struct ffmpeg_context *ctx)
{
struct tag *tag = (struct tag *) ctx->tag;
- const AVFormatContext *f = ctx->format_context;
+ AVFormatContext *f = ctx->format_context;
tag->time = 0;
if (f->duration != (int64_t)AV_NOPTS_VALUE)
tag->time = f->duration / AV_TIME_BASE;
#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
+ av_metadata_conv(f, NULL, f->iformat->metadata_conv);
+
ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_TITLE, "title");
ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ARTIST, "author");
ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ALBUM, "album");
diff --git a/src/decoder/flac_plugin.c b/src/decoder/flac_plugin.c
index 1d7a9f868..89a812f52 100644
--- a/src/decoder/flac_plugin.c
+++ b/src/decoder/flac_plugin.c
@@ -300,6 +300,8 @@ flac_cue_tag_load(const char *file)
FLAC__uint64 track_time = 0;
#ifdef HAVE_CUE /* libcue */
FLAC__StreamMetadata* vc = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT);
+ char* cs_filename;
+ FILE* cs_file;
#endif /* libcue */
FLAC__StreamMetadata* si = FLAC__metadata_object_new(FLAC__METADATA_TYPE_STREAMINFO);
FLAC__StreamMetadata* cs = FLAC__metadata_object_new(FLAC__METADATA_TYPE_CUESHEET);
@@ -328,6 +330,18 @@ flac_cue_tag_load(const char *file)
}
FLAC__metadata_object_delete(vc);
}
+
+ if (tag == NULL) {
+ cs_filename = g_strconcat(file, ".cue", NULL);
+
+ cs_file = fopen(cs_filename, "rt");
+ g_free(cs_filename);
+
+ if (cs_file != NULL) {
+ tag = cue_tag_file(cs_file, tnum);
+ fclose(cs_file);
+ }
+ }
#endif /* libcue */
if (tag == NULL)
diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c
index 1ef7183fa..c5287564f 100644
--- a/src/decoder/mad_plugin.c
+++ b/src/decoder/mad_plugin.c
@@ -1170,13 +1170,6 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
return ret != DECODE_BREAK;
}
-static void mp3_audio_format(struct mp3_data *data, struct audio_format *af)
-{
- af->bits = 24;
- af->sample_rate = (data->frame).header.samplerate;
- af->channels = MAD_NCHANNELS(&(data->frame).header);
-}
-
static void
mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
{
@@ -1192,7 +1185,8 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
return;
}
- mp3_audio_format(&data, &audio_format);
+ audio_format_init(&audio_format, data.frame.header.samplerate, 24,
+ MAD_NCHANNELS(&data.frame.header));
decoder_initialized(decoder, &audio_format,
data.input_stream->seekable, data.total_time);
diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c
index 065c34319..e7b7bfb03 100644
--- a/src/decoder/mikmod_plugin.c
+++ b/src/decoder/mikmod_plugin.c
@@ -175,9 +175,7 @@ mod_decode(struct decoder *decoder, const char *path)
return;
}
- audio_format.bits = 16;
- audio_format.sample_rate = 44100;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 44100, 16, 2);
secPerByte =
1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_plugin.c
index f636f2fa6..6c375e6a0 100644
--- a/src/decoder/modplug_plugin.c
+++ b/src/decoder/modplug_plugin.c
@@ -121,9 +121,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
return;
}
- audio_format.bits = 16;
- audio_format.sample_rate = 44100;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 44100, 16, 2);
sec_perbyte =
1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
@@ -186,7 +184,7 @@ static struct tag *mod_tagdup(const char *file)
return NULL;
}
ret = tag_new();
- ret->time = 0;
+ ret->time = ModPlug_GetLength(f) / 1000;
title = g_strdup(ModPlug_GetName(f));
if (title)
diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c
index cf9382904..d2c63f983 100644
--- a/src/decoder/mp4ff_plugin.c
+++ b/src/decoder/mp4ff_plugin.c
@@ -131,11 +131,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
}
*track_r = track;
- *audio_format = (struct audio_format){
- .bits = 16,
- .channels = channels,
- .sample_rate = sample_rate,
- };
+ audio_format_init(audio_format, sample_rate, 16, channels);
if (!audio_format_valid(audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c
index 26349f93a..a684da104 100644
--- a/src/decoder/mpcdec_plugin.c
+++ b/src/decoder/mpcdec_plugin.c
@@ -193,9 +193,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
mpc_demux_get_info(demux, &info);
#endif
- audio_format.bits = 24;
- audio_format.channels = info.channels;
- audio_format.sample_rate = info.sample_freq;
+ audio_format_init(&audio_format, info.sample_freq, 24, info.channels);
if (!audio_format_valid(&audio_format)) {
#ifndef MPC_IS_OLD_API
diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx
index c62e6b4b6..54ab746e2 100644
--- a/src/decoder/sidplay_plugin.cxx
+++ b/src/decoder/sidplay_plugin.cxx
@@ -103,9 +103,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
/* initialize the MPD decoder */
struct audio_format audio_format;
- audio_format.sample_rate = 48000;
- audio_format.bits = 16;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 48000, 16, 2);
decoder_initialized(decoder, &audio_format, false, -1);
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
new file mode 100644
index 000000000..4cc64459f
--- /dev/null
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -0,0 +1,244 @@
+/*
+ * 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 "decoder_api.h"
+
+#include <sndfile.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "sndfile"
+
+static sf_count_t
+sndfile_vio_get_filelen(void *user_data)
+{
+ const struct input_stream *is = user_data;
+
+ return is->size;
+}
+
+static sf_count_t
+sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
+{
+ struct input_stream *is = user_data;
+ bool success;
+
+ success = input_stream_seek(is, offset, whence);
+ if (!success)
+ return -1;
+
+ return is->offset;
+}
+
+static sf_count_t
+sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
+{
+ struct input_stream *is = user_data;
+ size_t nbytes;
+
+ nbytes = input_stream_read(is, ptr, count);
+ if (nbytes == 0 && is->error != 0)
+ return -1;
+
+ return nbytes;
+}
+
+static sf_count_t
+sndfile_vio_write(G_GNUC_UNUSED const void *ptr,
+ G_GNUC_UNUSED sf_count_t count,
+ G_GNUC_UNUSED void *user_data)
+{
+ /* no writing! */
+ return -1;
+}
+
+static sf_count_t
+sndfile_vio_tell(void *user_data)
+{
+ const struct input_stream *is = user_data;
+
+ return is->offset;
+}
+
+/**
+ * This SF_VIRTUAL_IO implementation wraps MPD's #input_stream to a
+ * libsndfile stream.
+ */
+static SF_VIRTUAL_IO vio = {
+ .get_filelen = sndfile_vio_get_filelen,
+ .seek = sndfile_vio_seek,
+ .read = sndfile_vio_read,
+ .write = sndfile_vio_write,
+ .tell = sndfile_vio_tell,
+};
+
+/**
+ * Converts a frame number to a timestamp (in seconds).
+ */
+static float
+frame_to_time(sf_count_t frame, const struct audio_format *audio_format)
+{
+ return (float)frame / (float)audio_format->sample_rate;
+}
+
+/**
+ * Converts a timestamp (in seconds) to a frame number.
+ */
+static sf_count_t
+time_to_frame(float t, const struct audio_format *audio_format)
+{
+ return (sf_count_t)(t * audio_format->sample_rate);
+}
+
+static void
+sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
+{
+ SNDFILE *sf;
+ SF_INFO info;
+ struct audio_format audio_format;
+ size_t frame_size;
+ sf_count_t read_frames, num_frames, position = 0;
+ int buffer[4096];
+ enum decoder_command cmd;
+
+ info.format = 0;
+
+ sf = sf_open_virtual(&vio, SFM_READ, &info, is);
+ if (sf == NULL) {
+ g_warning("sf_open_virtual() failed");
+ return;
+ }
+
+ /* for now, always read 32 bit samples. Later, we could lower
+ MPD's CPU usage by reading 16 bit samples with
+ sf_readf_short() on low-quality source files. */
+ audio_format_init(&audio_format, info.samplerate, 32, info.channels);
+
+ if (!audio_format_valid(&audio_format)) {
+ g_warning("invalid audio format");
+ return;
+ }
+
+ decoder_initialized(decoder, &audio_format, info.seekable,
+ frame_to_time(info.frames, &audio_format));
+
+ frame_size = audio_format_frame_size(&audio_format);
+ read_frames = sizeof(buffer) / frame_size;
+
+ do {
+ num_frames = sf_readf_int(sf, buffer, read_frames);
+ if (num_frames <= 0)
+ break;
+
+ cmd = decoder_data(decoder, is,
+ buffer, num_frames * frame_size,
+ frame_to_time(position, &audio_format),
+ 0, NULL);
+ if (cmd == DECODE_COMMAND_SEEK) {
+ sf_count_t c =
+ time_to_frame(decoder_seek_where(decoder),
+ &audio_format);
+ c = sf_seek(sf, c, SEEK_SET);
+ if (c < 0)
+ decoder_seek_error(decoder);
+ else
+ decoder_command_finished(decoder);
+ cmd = DECODE_COMMAND_NONE;
+ }
+ } while (cmd == DECODE_COMMAND_NONE);
+
+ sf_close(sf);
+}
+
+static struct tag *
+sndfile_tag_dup(const char *path_fs)
+{
+ SNDFILE *sf;
+ SF_INFO info;
+ struct tag *tag;
+ const char *p;
+
+ info.format = 0;
+
+ sf = sf_open(path_fs, SFM_READ, &info);
+ if (sf == NULL)
+ return NULL;
+
+ if (!audio_valid_sample_rate(info.samplerate)) {
+ sf_close(sf);
+ g_warning("Invalid sample rate in %s\n", path_fs);
+ return NULL;
+ }
+
+ tag = tag_new();
+ tag->time = info.frames / info.samplerate;
+
+ p = sf_get_string(sf, SF_STR_TITLE);
+ if (p != NULL)
+ tag_add_item(tag, TAG_ITEM_TITLE, p);
+
+ p = sf_get_string(sf, SF_STR_ARTIST);
+ if (p != NULL)
+ tag_add_item(tag, TAG_ITEM_ARTIST, p);
+
+ p = sf_get_string(sf, SF_STR_DATE);
+ if (p != NULL)
+ tag_add_item(tag, TAG_ITEM_DATE, p);
+
+ sf_close(sf);
+
+ return tag;
+}
+
+static const char *const sndfile_suffixes[] = {
+ "wav", "aiff", "aif", /* Microsoft / SGI / Apple */
+ "au", "snd", /* Sun / DEC / NeXT */
+ "paf", /* Paris Audio File */
+ "iff", "svx", /* Commodore Amiga IFF / SVX */
+ "sf", /* IRCAM */
+ "voc", /* Creative */
+ "w64", /* Soundforge */
+ "pvf", /* Portable Voice Format */
+ "xi", /* Fasttracker */
+ "htk", /* HMM Tool Kit */
+ "caf", /* Apple */
+ "sd2", /* Sound Designer II */
+
+ /* libsndfile also supports FLAC and Ogg Vorbis, but only by
+ linking with libFLAC and libvorbis - we can do better, we
+ have native plugins for these libraries */
+
+ NULL
+};
+
+static const char *const sndfile_mime_types[] = {
+ "audio/x-wav",
+ "audio/x-aiff",
+
+ /* what are the MIME types of the other supported formats? */
+
+ NULL
+};
+
+const struct decoder_plugin sndfile_decoder_plugin = {
+ .name = "sndfile",
+ .stream_decode = sndfile_stream_decode,
+ .tag_dup = sndfile_tag_dup,
+ .suffixes = sndfile_suffixes,
+ .mime_types = sndfile_mime_types,
+};
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c
index d4f81e91f..bab1d57ec 100644
--- a/src/decoder/vorbis_plugin.c
+++ b/src/decoder/vorbis_plugin.c
@@ -324,8 +324,7 @@ vorbis_stream_decode(struct decoder *decoder,
vorbis_info *vi = ov_info(&vf, -1);
struct replay_gain_info *new_rgi;
- audio_format.channels = vi->channels;
- audio_format.sample_rate = vi->rate;
+ audio_format_init(&audio_format, vi->rate, 16, vi->channels);
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
index 821536fb5..f3d701144 100644
--- a/src/decoder/wavpack_plugin.c
+++ b/src/decoder/wavpack_plugin.c
@@ -145,9 +145,9 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
int bytes_per_sample, output_sample_size;
int position;
- audio_format.sample_rate = WavpackGetSampleRate(wpc);
- audio_format.channels = WavpackGetReducedChannels(wpc);
- audio_format.bits = WavpackGetBitsPerSample(wpc);
+ audio_format_init(&audio_format, WavpackGetSampleRate(wpc),
+ WavpackGetBitsPerSample(wpc),
+ WavpackGetReducedChannels(wpc));
/* round bitwidth to 8-bit units */
audio_format.bits = (audio_format.bits + 7) & (~7);
diff --git a/src/decoder_api.c b/src/decoder_api.c
index 2ece3bb98..4cff9916c 100644
--- a/src/decoder_api.c
+++ b/src/decoder_api.c
@@ -57,7 +57,10 @@ void decoder_initialized(G_GNUC_UNUSED struct decoder * decoder,
dc.seekable = seekable;
dc.total_time = total_time;
+ decoder_lock();
dc.state = DECODE_STATE_DECODE;
+ decoder_unlock();
+
notify_signal(&pc.notify);
g_debug("audio_format=%u:%u:%u, seekable=%s",
@@ -88,6 +91,8 @@ enum decoder_command decoder_get_command(G_GNUC_UNUSED struct decoder * decoder)
void decoder_command_finished(G_GNUC_UNUSED struct decoder * decoder)
{
+ decoder_lock();
+
assert(dc.command != DECODE_COMMAND_NONE);
assert(dc.command != DECODE_COMMAND_SEEK ||
dc.seek_error || decoder->seeking);
@@ -105,6 +110,8 @@ void decoder_command_finished(G_GNUC_UNUSED struct decoder * decoder)
}
dc.command = DECODE_COMMAND_NONE;
+ decoder_unlock();
+
notify_signal(&pc.notify);
}
@@ -225,21 +232,24 @@ decoder_data(struct decoder *decoder,
struct replay_gain_info *replay_gain_info)
{
const char *data = _data;
+ GError *error = NULL;
+ enum decoder_command cmd;
assert(dc.state == DECODE_STATE_DECODE);
assert(dc.pipe != NULL);
assert(length % audio_format_frame_size(&dc.in_audio_format) == 0);
- if (dc.command == DECODE_COMMAND_STOP ||
- dc.command == DECODE_COMMAND_SEEK ||
+ decoder_lock();
+ cmd = dc.command;
+ decoder_unlock();
+
+ if (cmd == DECODE_COMMAND_STOP || cmd == DECODE_COMMAND_SEEK ||
length == 0)
- return dc.command;
+ return cmd;
/* send stream tags */
if (update_stream_tag(decoder, is)) {
- enum decoder_command cmd;
-
if (decoder->decoder_tag != NULL) {
/* merge with tag from decoder plugin */
struct tag *tag;
@@ -259,14 +269,15 @@ decoder_data(struct decoder *decoder,
if (!audio_format_equals(&dc.in_audio_format, &dc.out_audio_format)) {
data = pcm_convert(&decoder->conv_state,
&dc.in_audio_format, data, length,
- &dc.out_audio_format, &length);
-
- /* under certain circumstances, pcm_convert() may
- return an empty buffer - this condition should be
- investigated further, but for now, do this check as
- a workaround: */
- if (data == NULL)
- return DECODE_COMMAND_NONE;
+ &dc.out_audio_format, &length,
+ &error);
+ if (data == NULL) {
+ /* the PCM conversion has failed - stop
+ playback, since we have no better way to
+ bail out */
+ g_warning("%s", error->message);
+ return DECODE_COMMAND_STOP;
+ }
}
while (length > 0) {
@@ -301,8 +312,7 @@ decoder_data(struct decoder *decoder,
/* apply replay gain or normalization */
- if (replay_gain_info != NULL &&
- replay_gain_mode != REPLAY_GAIN_OFF)
+ if (replay_gain_mode != REPLAY_GAIN_OFF)
replay_gain_apply(replay_gain_info, dest, nbytes,
&dc.out_audio_format);
else if (normalizationEnabled)
diff --git a/src/decoder_control.c b/src/decoder_control.c
index 44bb63e15..3b993431c 100644
--- a/src/decoder_control.c
+++ b/src/decoder_control.c
@@ -18,6 +18,7 @@
*/
#include "decoder_control.h"
+#include "notify.h"
#include <assert.h>
@@ -25,36 +26,63 @@ struct decoder_control dc;
void dc_init(void)
{
- notify_init(&dc.notify);
+ dc.mutex = g_mutex_new();
+ dc.cond = g_cond_new();
+
dc.state = DECODE_STATE_STOP;
dc.command = DECODE_COMMAND_NONE;
}
void dc_deinit(void)
{
- notify_deinit(&dc.notify);
+ g_cond_free(dc.cond);
+ g_mutex_free(dc.mutex);
}
-void
-dc_command_wait(struct notify *notify)
+static void
+dc_command_wait_locked(struct notify *notify)
{
while (dc.command != DECODE_COMMAND_NONE) {
- notify_signal(&dc.notify);
+ decoder_signal();
+ decoder_unlock();
+
notify_wait(notify);
+
+ decoder_lock();
}
}
+void
+dc_command_wait(struct notify *notify)
+{
+ decoder_lock();
+ dc_command_wait_locked(notify);
+ decoder_unlock();
+}
+
static void
-dc_command(struct notify *notify, enum decoder_command cmd)
+dc_command_locked(struct notify *notify, enum decoder_command cmd)
{
dc.command = cmd;
- dc_command_wait(notify);
+ dc_command_wait_locked(notify);
+}
+
+static void
+dc_command(struct notify *notify, enum decoder_command cmd)
+{
+ decoder_lock();
+ dc_command_locked(notify, cmd);
+ decoder_unlock();
}
static void dc_command_async(enum decoder_command cmd)
{
+ decoder_lock();
+
dc.command = cmd;
- notify_signal(&dc.notify);
+ decoder_signal();
+
+ decoder_unlock();
}
void
@@ -80,15 +108,19 @@ dc_start_async(struct song *song)
void
dc_stop(struct notify *notify)
{
+ decoder_lock();
+
if (dc.command != DECODE_COMMAND_NONE)
/* Attempt to cancel the current command. If it's too
late and the decoder thread is already executing
the old command, we'll call STOP again in this
function (see below). */
- dc_command(notify, DECODE_COMMAND_STOP);
+ dc_command_locked(notify, DECODE_COMMAND_STOP);
if (dc.state != DECODE_STATE_STOP && dc.state != DECODE_STATE_ERROR)
- dc_command(notify, DECODE_COMMAND_STOP);
+ dc_command_locked(notify, DECODE_COMMAND_STOP);
+
+ decoder_unlock();
}
bool
diff --git a/src/decoder_control.h b/src/decoder_control.h
index 703ea256c..7e861f970 100644
--- a/src/decoder_control.h
+++ b/src/decoder_control.h
@@ -22,13 +22,16 @@
#include "decoder_command.h"
#include "audio_format.h"
-#include "notify.h"
+
+#include <glib.h>
#include <assert.h>
#define DECODE_TYPE_FILE 0
#define DECODE_TYPE_URL 1
+struct notify;
+
enum decoder_state {
DECODE_STATE_STOP = 0,
DECODE_STATE_START,
@@ -48,14 +51,25 @@ struct decoder_control {
thread isn't running */
GThread *thread;
- struct notify notify;
+ /**
+ * This lock protects #state and #command.
+ */
+ GMutex *mutex;
+
+ /**
+ * Trigger this object after you have modified #command. This
+ * is also used by the decoder thread to notify the caller
+ * when it has finished a command.
+ */
+ GCond *cond;
+
+ enum decoder_state state;
+ enum decoder_command command;
- volatile enum decoder_state state;
- volatile enum decoder_command command;
bool quit;
bool seek_error;
bool seekable;
- volatile double seek_where;
+ double seek_where;
/** the format of the song file */
struct audio_format in_audio_format;
@@ -80,6 +94,46 @@ void dc_init(void);
void dc_deinit(void);
+/**
+ * Locks the #decoder_control object.
+ */
+static inline void
+decoder_lock(void)
+{
+ g_mutex_lock(dc.mutex);
+}
+
+/**
+ * Unlocks the #decoder_control object.
+ */
+static inline void
+decoder_unlock(void)
+{
+ g_mutex_unlock(dc.mutex);
+}
+
+/**
+ * Waits for a signal on the #decoder_control object. This function
+ * is only valid in the decoder thread. The object must be locked
+ * prior to calling this function.
+ */
+static inline void
+decoder_wait(void)
+{
+ g_cond_wait(dc.cond, dc.mutex);
+}
+
+/**
+ * Signals the #decoder_control object. This function is only valid
+ * in the player thread. The object should be locked prior to calling
+ * this function.
+ */
+static inline void
+decoder_signal(void)
+{
+ g_cond_signal(dc.cond);
+}
+
static inline bool decoder_is_idle(void)
{
return (dc.state == DECODE_STATE_STOP ||
@@ -100,6 +154,39 @@ static inline bool decoder_has_failed(void)
return dc.state == DECODE_STATE_ERROR;
}
+static inline bool decoder_lock_is_idle(void)
+{
+ bool ret;
+
+ decoder_lock();
+ ret = decoder_is_idle();
+ decoder_unlock();
+
+ return ret;
+}
+
+static inline bool decoder_lock_is_starting(void)
+{
+ bool ret;
+
+ decoder_lock();
+ ret = decoder_is_starting();
+ decoder_unlock();
+
+ return ret;
+}
+
+static inline bool decoder_lock_has_failed(void)
+{
+ bool ret;
+
+ decoder_lock();
+ ret = decoder_has_failed();
+ decoder_unlock();
+
+ return ret;
+}
+
static inline struct song *
decoder_current_song(void)
{
diff --git a/src/decoder_internal.c b/src/decoder_internal.c
index 4a56fa5f3..1b064d0aa 100644
--- a/src/decoder_internal.c
+++ b/src/decoder_internal.c
@@ -28,6 +28,24 @@
#include <assert.h>
/**
+ * This is a wrapper for input_stream_buffer(). It assumes that the
+ * decoder is currently locked, and temporarily unlocks it while
+ * calling input_stream_buffer(). We shouldn't hold the lock during a
+ * potentially blocking operation.
+ */
+static int
+decoder_input_buffer(struct input_stream *is)
+{
+ int ret;
+
+ decoder_unlock();
+ ret = input_stream_buffer(is) > 0;
+ decoder_lock();
+
+ return ret;
+}
+
+/**
* All chunks are full of decoded data; wait for the player to free
* one.
*/
@@ -38,9 +56,12 @@ need_chunks(struct input_stream *is, bool do_wait)
dc.command == DECODE_COMMAND_SEEK)
return dc.command;
- if ((is == NULL || input_stream_buffer(is) <= 0) && do_wait) {
- notify_wait(&dc.notify);
+ if ((is == NULL || decoder_input_buffer(is) <= 0) && do_wait) {
+ decoder_wait();
+
+ decoder_unlock();
notify_signal(&pc.notify);
+ decoder_lock();
return dc.command;
}
@@ -63,7 +84,9 @@ decoder_get_chunk(struct decoder *decoder, struct input_stream *is)
if (decoder->chunk != NULL)
return decoder->chunk;
+ decoder_lock();
cmd = need_chunks(is, true);
+ decoder_unlock();
} while (cmd == DECODE_COMMAND_NONE);
return NULL;
diff --git a/src/decoder_list.c b/src/decoder_list.c
index a42585e34..177ac46e4 100644
--- a/src/decoder_list.c
+++ b/src/decoder_list.c
@@ -31,6 +31,7 @@ extern const struct decoder_plugin mad_decoder_plugin;
extern const struct decoder_plugin vorbis_decoder_plugin;
extern const struct decoder_plugin flac_decoder_plugin;
extern const struct decoder_plugin oggflac_decoder_plugin;
+extern const struct decoder_plugin sndfile_decoder_plugin;
extern const struct decoder_plugin audiofile_decoder_plugin;
extern const struct decoder_plugin mp4ff_decoder_plugin;
extern const struct decoder_plugin faad_decoder_plugin;
@@ -56,6 +57,9 @@ static const struct decoder_plugin *const decoder_plugins[] = {
#ifdef HAVE_FLAC
&flac_decoder_plugin,
#endif
+#ifdef ENABLE_SNDFILE
+ &sndfile_decoder_plugin,
+#endif
#ifdef HAVE_AUDIOFILE
&audiofile_decoder_plugin,
#endif
diff --git a/src/decoder_thread.c b/src/decoder_thread.c
index 2b1a6299a..be37896c1 100644
--- a/src/decoder_thread.c
+++ b/src/decoder_thread.c
@@ -49,11 +49,15 @@ decoder_stream_decode(const struct decoder_plugin *plugin,
assert(input_stream->ready);
assert(dc.state == DECODE_STATE_START);
+ decoder_unlock();
+
/* rewind the stream, so each plugin gets a fresh start */
input_stream_seek(input_stream, 0, SEEK_SET);
decoder_plugin_stream_decode(plugin, decoder, input_stream);
+ decoder_lock();
+
assert(dc.state == DECODE_STATE_START ||
dc.state == DECODE_STATE_DECODE);
@@ -73,8 +77,12 @@ decoder_file_decode(const struct decoder_plugin *plugin,
assert(path[0] == '/');
assert(dc.state == DECODE_STATE_START);
+ decoder_unlock();
+
decoder_plugin_file_decode(plugin, decoder, path);
+ decoder_lock();
+
assert(dc.state == DECODE_STATE_START ||
dc.state == DECODE_STATE_DECODE);
@@ -103,28 +111,40 @@ static void decoder_run_song(const struct song *song, const char *uri)
dc.state = DECODE_STATE_START;
dc.command = DECODE_COMMAND_NONE;
+
+ decoder_unlock();
notify_signal(&pc.notify);
+ decoder_lock();
/* wait for the input stream to become ready; its metadata
will be available then */
while (!input_stream.ready) {
if (dc.command == DECODE_COMMAND_STOP) {
+ decoder_unlock();
input_stream_close(&input_stream);
+ decoder_lock();
dc.state = DECODE_STATE_STOP;
return;
}
+ decoder_unlock();
ret = input_stream_buffer(&input_stream);
if (ret < 0) {
input_stream_close(&input_stream);
+ decoder_lock();
dc.state = DECODE_STATE_ERROR;
return;
}
+
+ decoder_lock();
}
if (dc.command == DECODE_COMMAND_STOP) {
+ decoder_unlock();
input_stream_close(&input_stream);
+ decoder_lock();
+
dc.state = DECODE_STATE_STOP;
return;
}
@@ -179,7 +199,10 @@ static void decoder_run_song(const struct song *song, const char *uri)
const char *s = uri_get_suffix(uri);
while ((plugin = decoder_plugin_from_suffix(s, next++))) {
if (plugin->file_decode != NULL) {
+ decoder_unlock();
input_stream_close(&input_stream);
+ decoder_lock();
+
close_instream = false;
ret = decoder_file_decode(plugin,
&decoder, uri);
@@ -191,7 +214,13 @@ static void decoder_run_song(const struct song *song, const char *uri)
been closed before
decoder_file_decode() -
reopen it */
- if (input_stream_open(&input_stream, uri))
+ bool success;
+
+ decoder_unlock();
+ success = input_stream_open(&input_stream, uri);
+ decoder_lock();
+
+ if (success)
close_instream = true;
else
continue;
@@ -205,6 +234,8 @@ static void decoder_run_song(const struct song *song, const char *uri)
}
}
+ decoder_unlock();
+
pcm_convert_deinit(&decoder.conv_state);
/* flush the last chunk */
@@ -223,6 +254,8 @@ static void decoder_run_song(const struct song *song, const char *uri)
if (decoder.decoder_tag != NULL)
tag_free(decoder.decoder_tag);
+ decoder_lock();
+
dc.state = ret ? DECODE_STATE_STOP : DECODE_STATE_ERROR;
}
@@ -249,6 +282,8 @@ static void decoder_run(void)
static gpointer decoder_task(G_GNUC_UNUSED gpointer arg)
{
+ decoder_lock();
+
do {
assert(dc.state == DECODE_STATE_STOP ||
dc.state == DECODE_STATE_ERROR);
@@ -259,20 +294,28 @@ static gpointer decoder_task(G_GNUC_UNUSED gpointer arg)
decoder_run();
dc.command = DECODE_COMMAND_NONE;
+
+ decoder_unlock();
notify_signal(&pc.notify);
+ decoder_lock();
break;
case DECODE_COMMAND_STOP:
dc.command = DECODE_COMMAND_NONE;
+
+ decoder_unlock();
notify_signal(&pc.notify);
+ decoder_lock();
break;
case DECODE_COMMAND_NONE:
- notify_wait(&dc.notify);
+ decoder_wait();
break;
}
} while (dc.command != DECODE_COMMAND_NONE || !dc.quit);
+ decoder_unlock();
+
return NULL;
}
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-18 05:39:27 +0000