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
flac_got_stream_info(struct flac_data *data,
const FLAC__StreamMetadata_StreamInfo *stream_info)
{
if (data->initialized || data->unsupported)
return;
GError *error = NULL;
if (!audio_format_init_checked(&data->audio_format,
stream_info->sample_rate,
flac_sample_format(stream_info->bits_per_sample),
stream_info->channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
data->unsupported = true;
return;
}
data->frame_size = audio_format_frame_size(&data->audio_format);
if (data->total_frames == 0)
data->total_frames = stream_info->total_samples;
data->initialized = true;
}
/**
* This function attempts to call decoder_initialized() in case there
* was no STREAMINFO block. This is allowed for nonseekable streams,
* where the server sends us only a part of the file, without
* providing the STREAMINFO block from the beginning of the file
* (e.g. when seeking with SqueezeBox Server).
*/
static bool
flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header)
{
if (data->unsupported)
return false;
GError *error = NULL;
if (!audio_format_init_checked(&data->audio_format,
header->sample_rate,
flac_sample_format(header->bits_per_sample),
header->channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
data->unsupported = true;
return false;
}
data->frame_size = audio_format_frame_size(&data->audio_format);
decoder_initialized(data->decoder, &data->audio_format,
data->input_stream->seekable,
(float)data->total_frames /
(float)data->audio_format.sample_rate);
data->initialized = true;
return true;
}
static void
mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
{
struct mp3_data data;
GError *error = NULL;
struct tag *tag = NULL;
struct audio_format audio_format;
if (!mp3_open(input_stream, &data, decoder, &tag)) {
if (decoder_get_command(decoder) == DECODE_COMMAND_NONE)
g_warning
("Input does not appear to be a mp3 bit stream.\n");
return;
}
if (!audio_format_init_checked(&audio_format,
data.frame.header.samplerate,
SAMPLE_FORMAT_S24_P32,
MAD_NCHANNELS(&data.frame.header),
&error)) {
g_warning("%s", error->message);
g_error_free(error);
if (tag != NULL)
tag_free(tag);
mp3_data_finish(&data);
return;
}
decoder_initialized(decoder, &audio_format,
data.input_stream->seekable, data.total_time);
if (tag != NULL) {
decoder_tag(decoder, input_stream, tag);
tag_free(tag);
}
while (mp3_read(&data)) ;
mp3_data_finish(&data);
}
static void
dsf_stream_decode(struct decoder *decoder, struct input_stream *is)
{
struct dsf_metadata metadata = {
.sample_rate = 0,
.channels = 0,
};
/* check if it is a proper DSF file */
if (!dsf_read_metadata(decoder, is, &metadata))
return;
GError *error = NULL;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
SAMPLE_FORMAT_DSD,
metadata.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
return;
}
/* success: file was recognized */
decoder_initialized(decoder, &audio_format, false, -1);
if (!dsf_decode_chunk(decoder, is, metadata.channels,
metadata.chunk_size,
metadata.bitreverse))
return;
}
static bool
dsf_scan_stream(struct input_stream *is,
G_GNUC_UNUSED const struct tag_handler *handler,
G_GNUC_UNUSED void *handler_ctx)
{
struct dsf_metadata metadata = {
.sample_rate = 0,
.channels = 0,
};
/* check DSF metadata */
if (!dsf_read_metadata(NULL, is, &metadata))
return false;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
SAMPLE_FORMAT_DSD,
metadata.channels, NULL))
/* refuse to parse files which we cannot play anyway */
return false;
return true;
}
static const char *const dsf_suffixes[] = {
"dsf",
NULL
};
static const char *const dsf_mime_types[] = {
"application/x-dsf",
NULL
};
const struct decoder_plugin dsf_decoder_plugin = {
.name = "dsf",
.stream_decode = dsf_stream_decode,
.scan_stream = dsf_scan_stream,
.suffixes = dsf_suffixes,
.mime_types = dsf_mime_types,
};
/* 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);
}
static void
dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is)
{
struct dsdiff_metadata metadata = {
.sample_rate = 0,
.channels = 0,
};
struct dsdiff_chunk_header chunk_header;
/* First see if it is is a DFF file */
if (!dsdiff_read_metadata(decoder, is, &metadata, &chunk_header))
{
/* It was not a DFF file, now check if it is a DSF file */
if (!dsf_read_metadata(decoder, is, &metadata))
return;
}
GError *error = NULL;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
SAMPLE_FORMAT_DSD,
metadata.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
return;
}
/* success: file was recognized */
decoder_initialized(decoder, &audio_format, false, -1);
if (!metadata.fileisdff) {
uint64_t chunk_size = metadata.chunk_size;
if (!dsdiff_decode_chunk(decoder, is,
metadata.channels,
chunk_size,
metadata.fileisdff,
metadata.bitreverse))
return;
} else {
/* every iteration of the following loop decodes one "DSD"
chunk from a DFF file */
while (true) {
uint64_t chunk_size = dsdiff_chunk_size(&chunk_header);
if (dsdiff_id_equals(&chunk_header.id, "DSD ")) {
if (!dsdiff_decode_chunk(decoder, is,
metadata.channels,
chunk_size,
metadata.fileisdff,
/* Set bitreverse to
false for DFF files */
false))
break;
} else {
/* ignore other chunks */
if (!dsdiff_skip(decoder, is, chunk_size))
break;
}
/* read next chunk header; the first one was read by
dsdiff_read_metadata() */
if (!dsdiff_read_chunk_header(decoder,
is, &chunk_header))
break;
}
}
}
static bool
dsdiff_scan_stream(struct input_stream *is,
G_GNUC_UNUSED const struct tag_handler *handler,
G_GNUC_UNUSED void *handler_ctx)
{
struct dsdiff_metadata metadata = {
.sample_rate = 0,
.channels = 0,
};
struct dsdiff_chunk_header chunk_header;
/* First check for DFF metadata */
if (!dsdiff_read_metadata(NULL, is, &metadata, &chunk_header))
{
/* It was not an DFF file, now check for DSF metadata */
if (!dsf_read_metadata(NULL, is, &metadata))
return false;
}
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
SAMPLE_FORMAT_DSD,
metadata.channels, NULL))
/* refuse to parse files which we cannot play anyway */
return false;
/* no total time estimate, no tags implemented yet */
return true;
}
static const char *const dsdiff_suffixes[] = {
"dff",
"dsf",
NULL
};
static const char *const dsdiff_mime_types[] = {
"application/x-dff",
"application/x-dsf",
NULL
};
const struct decoder_plugin dsdiff_decoder_plugin = {
.name = "dsdiff",
.init = dsdiff_init,
.stream_decode = dsdiff_stream_decode,
.scan_stream = dsdiff_scan_stream,
.suffixes = dsdiff_suffixes,
.mime_types = dsdiff_mime_types,
};
static void
audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
{
GError *error = NULL;
AFvirtualfile *vf;
int fs, frame_count;
AFfilehandle af_fp;
struct audio_format audio_format;
float total_time;
uint16_t bit_rate;
int ret;
char chunk[CHUNK_SIZE];
enum decoder_command cmd;
if (!is->seekable) {
g_warning("not seekable");
return;
}
vf = setup_virtual_fops(is);
af_fp = afOpenVirtualFile(vf, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
g_warning("failed to input stream\n");
return;
}
if (!audio_format_init_checked(&audio_format,
afGetRate(af_fp, AF_DEFAULT_TRACK),
audiofile_setup_sample_format(af_fp),
afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK),
&error)) {
g_warning("%s", error->message);
g_error_free(error);
afCloseFile(af_fp);
return;
}
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
total_time = ((float)frame_count / (float)audio_format.sample_rate);
bit_rate = (uint16_t)(is->size * 8.0 / total_time / 1000.0 + 0.5);
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
decoder_initialized(decoder, &audio_format, true, total_time);
do {
ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
CHUNK_SIZE / fs);
if (ret <= 0)
break;
cmd = decoder_data(decoder, NULL,
chunk, ret * fs,
bit_rate);
if (cmd == DECODE_COMMAND_SEEK) {
AFframecount frame = decoder_seek_where(decoder) *
audio_format.sample_rate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, frame);
decoder_command_finished(decoder);
cmd = DECODE_COMMAND_NONE;
}
} while (cmd == DECODE_COMMAND_NONE);
afCloseFile(af_fp);
}
/* 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);
}
}
