/**
* Decode one "DSD" chunk.
*/
static bool
dsdiff_decode_chunk(struct decoder *decoder, struct input_stream *is,
unsigned channels,
uint64_t chunk_size,
bool fileisdff,
bool bitreverse)
{
uint8_t buffer[8192];
/* Scratch buffer for DSF samples to convert to the needed
normal Left/Right regime of samples */
uint8_t dsf_scratch_buffer[8192];
const size_t sample_size = sizeof(buffer[0]);
const size_t frame_size = channels * sample_size;
const unsigned buffer_frames = sizeof(buffer) / frame_size;
const unsigned buffer_samples = buffer_frames * frame_size;
const size_t buffer_size = buffer_samples * sample_size;
while (chunk_size > 0) {
/* see how much aligned data from the remaining chunk
fits into the local buffer */
unsigned now_frames = buffer_frames;
size_t now_size = buffer_size;
if (chunk_size < (uint64_t)now_size) {
now_frames = (unsigned)chunk_size / frame_size;
now_size = now_frames * frame_size;
}
size_t nbytes = decoder_read(decoder, is, buffer, now_size);
if (nbytes != now_size)
return false;
chunk_size -= nbytes;
if (lsbitfirst || bitreverse)
bit_reverse_buffer(buffer, buffer + nbytes);
if (!fileisdff)
dsf_to_pcm_order(buffer, dsf_scratch_buffer, nbytes);
enum decoder_command cmd =
decoder_data(decoder, is, buffer, nbytes, 0);
switch (cmd) {
case DECODE_COMMAND_NONE:
break;
case DECODE_COMMAND_START:
case DECODE_COMMAND_STOP:
return false;
case DECODE_COMMAND_SEEK:
/* not implemented yet */
decoder_seek_error(decoder);
break;
}
}
return dsdiff_skip(decoder, is, chunk_size);
}
static void
sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
{
GError *error = NULL;
SNDFILE *sf;
SF_INFO info;
struct audio_format audio_format;
size_t frame_size;
sf_count_t read_frames, num_frames;
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. */
if (!audio_format_init_checked(&audio_format, info.samplerate,
SAMPLE_FORMAT_S32,
info.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
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,
0);
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 void
pcm_stream_decode(struct decoder *decoder, struct input_stream *is)
{
static const struct audio_format audio_format = {
.sample_rate = 44100,
.format = SAMPLE_FORMAT_S16,
.channels = 2,
};
GError *error = NULL;
enum decoder_command cmd;
double time_to_size = audio_format_time_to_size(&audio_format);
float total_time = -1;
if (is->size >= 0)
total_time = is->size / time_to_size;
decoder_initialized(decoder, &audio_format, is->seekable, total_time);
do {
char buffer[4096];
size_t nbytes = decoder_read(decoder, is,
buffer, sizeof(buffer));
if (nbytes == 0 && input_stream_eof(is))
break;
cmd = nbytes > 0
? decoder_data(decoder, is,
buffer, nbytes, 0)
: decoder_get_command(decoder);
if (cmd == DECODE_COMMAND_SEEK) {
goffset offset = (goffset)(time_to_size *
decoder_seek_where(decoder));
if (input_stream_seek(is, offset, SEEK_SET, &error)) {
decoder_command_finished(decoder);
} else {
g_warning("seeking failed: %s", error->message);
g_error_free(error);
decoder_seek_error(decoder);
}
cmd = DECODE_COMMAND_NONE;
}
} while (cmd == DECODE_COMMAND_NONE);
}
static const char *const pcm_mime_types[] = {
/* for streams obtained by the cdio_paranoia input plugin */
"audio/x-mpd-cdda-pcm",
NULL
};
const struct decoder_plugin pcm_decoder_plugin = {
.name = "pcm",
.stream_decode = pcm_stream_decode,
.mime_types = pcm_mime_types,
};
static void
flac_decoder_loop(struct flac_data *data, FLAC__StreamDecoder *flac_dec,
FLAC__uint64 t_start, FLAC__uint64 t_end)
{
struct decoder *decoder = data->decoder;
enum decoder_command cmd;
data->first_frame = t_start;
while (true) {
if (data->tag != NULL && !tag_is_empty(data->tag)) {
cmd = decoder_tag(data->decoder, data->input_stream,
data->tag);
tag_free(data->tag);
data->tag = tag_new();
} else
cmd = decoder_get_command(decoder);
if (cmd == DECODE_COMMAND_SEEK) {
FLAC__uint64 seek_sample = t_start +
decoder_seek_where(decoder) *
data->audio_format.sample_rate;
if (seek_sample >= t_start &&
(t_end == 0 || seek_sample <= t_end) &&
FLAC__stream_decoder_seek_absolute(flac_dec, seek_sample)) {
data->next_frame = seek_sample;
data->position = 0;
decoder_command_finished(decoder);
} else
decoder_seek_error(decoder);
} else if (cmd == DECODE_COMMAND_STOP ||
FLAC__stream_decoder_get_state(flac_dec) == FLAC__STREAM_DECODER_END_OF_STREAM)
break;
if (t_end != 0 && data->next_frame >= t_end)
/* end of this sub track */
break;
if (!FLAC__stream_decoder_process_single(flac_dec)) {
cmd = decoder_get_command(decoder);
if (cmd != DECODE_COMMAND_SEEK)
break;
}
}
if (cmd != DECODE_COMMAND_STOP) {
flacPrintErroredState(FLAC__stream_decoder_get_state(flac_dec));
FLAC__stream_decoder_finish(flac_dec);
}
}
static bool
mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
{
struct decoder *decoder = data->decoder;
enum mp3_action ret;
enum decoder_command cmd;
mp3_update_timer_next_frame(data);
switch (data->mute_frame) {
case MUTEFRAME_SKIP:
data->mute_frame = MUTEFRAME_NONE;
break;
case MUTEFRAME_SEEK:
if (data->elapsed_time >= data->seek_where)
data->mute_frame = MUTEFRAME_NONE;
break;
case MUTEFRAME_NONE:
cmd = mp3_synth_and_send(data,
replay_gain_info_r != NULL
? *replay_gain_info_r : NULL);
if (cmd == DECODE_COMMAND_SEEK) {
unsigned long j;
assert(data->input_stream->seekable);
j = mp3_time_to_frame(data,
decoder_seek_where(decoder));
if (j < data->highest_frame) {
if (mp3_seek(data, data->frame_offsets[j])) {
data->current_frame = j;
decoder_command_finished(decoder);
} else
decoder_seek_error(decoder);
} else {
data->seek_where = decoder_seek_where(decoder);
data->mute_frame = MUTEFRAME_SEEK;
decoder_command_finished(decoder);
}
} else if (cmd != DECODE_COMMAND_NONE)
return false;
}
while (true) {
bool skip = false;
do {
struct tag *tag = NULL;
ret = decode_next_frame_header(data, &tag,
replay_gain_info_r);
if (tag != NULL) {
decoder_tag(decoder, data->input_stream, tag);
tag_free(tag);
}
} while (ret == DECODE_CONT);
if (ret == DECODE_BREAK)
return false;
else if (ret == DECODE_SKIP)
skip = true;
if (data->mute_frame == MUTEFRAME_NONE) {
do {
ret = decodeNextFrame(data);
} while (ret == DECODE_CONT);
if (ret == DECODE_BREAK)
return false;
}
if (!skip && ret == DECODE_OK)
break;
}
return ret != DECODE_BREAK;
}
/* public */
static void
vorbis_stream_decode(struct decoder *decoder,
struct input_stream *input_stream)
{
GError *error = NULL;
OggVorbis_File vf;
struct vorbis_input_stream vis;
struct audio_format audio_format;
float total_time;
int current_section;
int prev_section = -1;
long ret;
char chunk[OGG_CHUNK_SIZE];
long bitRate = 0;
long test;
const vorbis_info *vi;
enum decoder_command cmd = DECODE_COMMAND_NONE;
if (ogg_stream_type_detect(input_stream) != VORBIS)
return;
/* rewind the stream, because ogg_stream_type_detect() has
moved it */
input_stream_lock_seek(input_stream, 0, SEEK_SET, NULL);
if (!vorbis_is_open(&vis, &vf, decoder, input_stream))
return;
vi = ov_info(&vf, -1);
if (vi == NULL) {
g_warning("ov_info() has failed");
return;
}
if (!audio_format_init_checked(&audio_format, vi->rate,
SAMPLE_FORMAT_S16,
vi->channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
return;
}
total_time = ov_time_total(&vf, -1);
if (total_time < 0)
total_time = 0;
decoder_initialized(decoder, &audio_format, vis.seekable, total_time);
do {
if (cmd == DECODE_COMMAND_SEEK) {
double seek_where = decoder_seek_where(decoder);
if (0 == ov_time_seek_page(&vf, seek_where)) {
decoder_command_finished(decoder);
} else
decoder_seek_error(decoder);
}
ret = ov_read(&vf, chunk, sizeof(chunk),
OGG_DECODE_USE_BIGENDIAN, 2, 1, ¤t_section);
if (ret == OV_HOLE) /* bad packet */
ret = 0;
else if (ret <= 0)
/* break on EOF or other error */
break;
if (current_section != prev_section) {
char **comments;
vi = ov_info(&vf, -1);
if (vi == NULL) {
g_warning("ov_info() has failed");
break;
}
if (vi->rate != (long)audio_format.sample_rate ||
vi->channels != (int)audio_format.channels) {
/* we don't support audio format
change yet */
g_warning("audio format change, stopping here");
break;
}
comments = ov_comment(&vf, -1)->user_comments;
vorbis_send_comments(decoder, input_stream, comments);
struct replay_gain_info rgi;
if (vorbis_comments_to_replay_gain(&rgi, comments))
decoder_replay_gain(decoder, &rgi);
prev_section = current_section;
}
if ((test = ov_bitrate_instant(&vf)) > 0)
bitRate = test / 1000;
cmd = decoder_data(decoder, input_stream,
chunk, ret,
bitRate);
} while (cmd != DECODE_COMMAND_STOP);
ov_clear(&vf);
}
/*
* This does the main decoding thing.
* Requires an already opened WavpackContext.
*/
static void
wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
struct replay_gain_info *replay_gain_info)
{
struct audio_format audio_format;
format_samples_t format_samples;
char chunk[CHUNK_SIZE];
int samples_requested, samples_got;
float total_time, current_time;
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);
/* round bitwidth to 8-bit units */
audio_format.bits = (audio_format.bits + 7) & (~7);
/* mpd handles max 24-bit samples */
if (audio_format.bits > 24) {
audio_format.bits = 24;
}
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
audio_format.sample_rate,
audio_format.bits,
audio_format.channels);
return;
}
if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT) {
format_samples = format_samples_float;
} else {
format_samples = format_samples_int;
}
total_time = WavpackGetNumSamples(wpc);
total_time /= audio_format.sample_rate;
bytes_per_sample = WavpackGetBytesPerSample(wpc);
output_sample_size = audio_format_frame_size(&audio_format);
/* wavpack gives us all kind of samples in a 32-bit space */
samples_requested = sizeof(chunk) / (4 * audio_format.channels);
decoder_initialized(decoder, &audio_format, can_seek, total_time);
position = 0;
do {
if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
if (can_seek) {
int where;
where = decoder_seek_where(decoder);
where *= audio_format.sample_rate;
if (WavpackSeekSample(wpc, where)) {
position = where;
decoder_command_finished(decoder);
} else {
decoder_seek_error(decoder);
}
} else {
decoder_seek_error(decoder);
}
}
if (decoder_get_command(decoder) == DECODE_COMMAND_STOP) {
break;
}
samples_got = WavpackUnpackSamples(
wpc, (int32_t *)chunk, samples_requested
);
if (samples_got > 0) {
int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
1000 + 0.5);
position += samples_got;
current_time = position;
current_time /= audio_format.sample_rate;
format_samples(
bytes_per_sample, chunk,
samples_got * audio_format.channels
);
decoder_data(
decoder, NULL, chunk,
samples_got * output_sample_size,
current_time, bitrate,
replay_gain_info
);
}
} while (samples_got > 0);
}
/* public */
static void
vorbis_stream_decode(struct decoder *decoder,
struct input_stream *input_stream)
{
GError *error = NULL;
if (ogg_codec_detect(decoder, input_stream) != OGG_CODEC_VORBIS)
return;
/* rewind the stream, because ogg_codec_detect() has
moved it */
input_stream_lock_seek(input_stream, 0, SEEK_SET, NULL);
struct vorbis_input_stream vis;
OggVorbis_File vf;
if (!vorbis_is_open(&vis, &vf, decoder, input_stream))
return;
const vorbis_info *vi = ov_info(&vf, -1);
if (vi == NULL) {
g_warning("ov_info() has failed");
return;
}
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, vi->rate,
#ifdef HAVE_TREMOR
SAMPLE_FORMAT_S16,
#else
SAMPLE_FORMAT_FLOAT,
#endif
vi->channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
return;
}
float total_time = ov_time_total(&vf, -1);
if (total_time < 0)
total_time = 0;
decoder_initialized(decoder, &audio_format, vis.seekable, total_time);
enum decoder_command cmd = decoder_get_command(decoder);
#ifdef HAVE_TREMOR
char buffer[4096];
#else
float buffer[2048];
const int frames_per_buffer =
G_N_ELEMENTS(buffer) / audio_format.channels;
const unsigned frame_size = sizeof(buffer[0]) * audio_format.channels;
#endif
int prev_section = -1;
unsigned kbit_rate = 0;
do {
if (cmd == DECODE_COMMAND_SEEK) {
double seek_where = decoder_seek_where(decoder);
if (0 == ov_time_seek_page(&vf, seek_where)) {
decoder_command_finished(decoder);
} else
decoder_seek_error(decoder);
}
int current_section;
#ifdef HAVE_TREMOR
long nbytes = ov_read(&vf, buffer, sizeof(buffer),
VORBIS_BIG_ENDIAN, 2, 1,
¤t_section);
#else
float **per_channel;
long nframes = ov_read_float(&vf, &per_channel,
frames_per_buffer,
¤t_section);
long nbytes = nframes;
if (nframes > 0) {
vorbis_interleave(buffer,
(const float*const*)per_channel,
nframes, audio_format.channels);
nbytes *= frame_size;
}
#endif
if (nbytes == OV_HOLE) /* bad packet */
nbytes = 0;
else if (nbytes <= 0)
/* break on EOF or other error */
break;
if (current_section != prev_section) {
vi = ov_info(&vf, -1);
if (vi == NULL) {
g_warning("ov_info() has failed");
break;
}
if (vi->rate != (long)audio_format.sample_rate ||
vi->channels != (int)audio_format.channels) {
/* we don't support audio format
change yet */
g_warning("audio format change, stopping here");
break;
}
char **comments = ov_comment(&vf, -1)->user_comments;
vorbis_send_comments(decoder, input_stream, comments);
struct replay_gain_info rgi;
if (vorbis_comments_to_replay_gain(&rgi, comments))
decoder_replay_gain(decoder, &rgi);
prev_section = current_section;
}
long test = ov_bitrate_instant(&vf);
if (test > 0)
kbit_rate = test / 1000;
cmd = decoder_data(decoder, input_stream,
buffer, nbytes,
kbit_rate);
} while (cmd != DECODE_COMMAND_STOP);
ov_clear(&vf);
}
/*
* This does the main decoding thing.
* Requires an already opened WavpackContext.
*/
static void
wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
{
GError *error = NULL;
bool is_float;
enum sample_format sample_format;
struct audio_format audio_format;
format_samples_t format_samples;
float total_time;
int bytes_per_sample, output_sample_size;
is_float = (WavpackGetMode(wpc) & MODE_FLOAT) != 0;
sample_format =
wavpack_bits_to_sample_format(is_float,
WavpackGetBytesPerSample(wpc));
if (!audio_format_init_checked(&audio_format,
WavpackGetSampleRate(wpc),
sample_format,
WavpackGetNumChannels(wpc), &error)) {
g_warning("%s", error->message);
g_error_free(error);
return;
}
if (is_float) {
format_samples = format_samples_float;
} else {
format_samples = format_samples_int;
}
total_time = WavpackGetNumSamples(wpc);
total_time /= audio_format.sample_rate;
bytes_per_sample = WavpackGetBytesPerSample(wpc);
output_sample_size = audio_format_frame_size(&audio_format);
/* wavpack gives us all kind of samples in a 32-bit space */
int32_t chunk[1024];
const uint32_t samples_requested = G_N_ELEMENTS(chunk) /
audio_format.channels;
decoder_initialized(decoder, &audio_format, can_seek, total_time);
enum decoder_command cmd = decoder_get_command(decoder);
while (cmd != DECODE_COMMAND_STOP) {
if (cmd == DECODE_COMMAND_SEEK) {
if (can_seek) {
unsigned where = decoder_seek_where(decoder) *
audio_format.sample_rate;
if (WavpackSeekSample(wpc, where)) {
decoder_command_finished(decoder);
} else {
decoder_seek_error(decoder);
}
} else {
decoder_seek_error(decoder);
}
}
uint32_t samples_got = WavpackUnpackSamples(wpc, chunk,
samples_requested);
if (samples_got == 0)
break;
int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 +
0.5);
format_samples(bytes_per_sample, chunk,
samples_got * audio_format.channels);
cmd = decoder_data(decoder, NULL, chunk,
samples_got * output_sample_size,
bitrate);
}
}
