static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
const FLAC__int32 * const buffer[])
{
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
guint depth = frame->header.bits_per_sample;
guint width, gdepth;
guint sample_rate = frame->header.sample_rate;
guint channels = frame->header.channels;
guint samples = frame->header.blocksize;
guint j, i;
GstMapInfo map;
gboolean caps_changed;
GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples);
if (depth == 0) {
if (flacdec->depth < 4 || flacdec->depth > 32) {
GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO",
flacdec->depth);
ret = GST_FLOW_ERROR;
goto done;
}
depth = flacdec->depth;
}
switch (depth) {
case 8:
gdepth = width = 8;
break;
case 12:
case 16:
gdepth = width = 16;
break;
case 20:
case 24:
gdepth = 24;
width = 32;
break;
case 32:
gdepth = width = 32;
break;
default:
GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
ret = GST_FLOW_ERROR;
goto done;
}
if (sample_rate == 0) {
if (flacdec->info.rate != 0) {
sample_rate = flacdec->info.rate;
} else {
GST_ERROR_OBJECT (flacdec, "unknown sample rate");
ret = GST_FLOW_ERROR;
goto done;
}
}
caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info))
|| (width != GST_AUDIO_INFO_WIDTH (&flacdec->info))
|| (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info))
|| (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info));
if (caps_changed
|| !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) {
GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate,
channels);
gst_audio_info_set_format (&flacdec->info,
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
sample_rate, channels, NULL);
memcpy (flacdec->info.position,
channel_positions[flacdec->info.channels - 1],
sizeof (GstAudioChannelPosition) * flacdec->info.channels);
gst_audio_channel_positions_to_valid_order (flacdec->info.position,
flacdec->info.channels);
/* Note: we create the inverse reordering map here */
gst_audio_get_channel_reorder_map (flacdec->info.channels,
flacdec->info.position, channel_positions[flacdec->info.channels - 1],
flacdec->channel_reorder_map);
flacdec->depth = depth;
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec),
&flacdec->info);
}
outbuf =
gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
if (width == 8) {
gint8 *outbuffer = (gint8 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint8) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint8) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint16) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint16) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 32) {
gint32 *outbuffer = (gint32 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint32) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint32) buffer[reorder_map[j]][i];
}
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &map);
GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples);
if (flacdec->error_count)
flacdec->error_count--;
ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret));
}
done:
/* we act on the flow return value later in the handle_frame function, as we
* don't want to mess up the internal decoder state by returning ABORT when
* the error is in fact non-fatal (like a pad in flushing mode) and we want
* to continue later. So just pretend everything's dandy and act later. */
flacdec->last_flow = ret;
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
gboolean
gst_sndio_prepare (struct gstsndio *sio, GstAudioRingBufferSpec *spec)
{
struct sio_par par, retpar;
unsigned nchannels;
GST_DEBUG_OBJECT (sio, "prepare");
if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_READ_WRITE,
("Only raw buffer format supported by sndio"), (NULL));
return FALSE;
}
if (!GST_AUDIO_INFO_IS_INTEGER(&spec->info)) {
GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_READ_WRITE,
("Only integer format supported"), (NULL));
return FALSE;
}
if (GST_AUDIO_INFO_DEPTH(&spec->info) % 8) {
GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_READ_WRITE,
("Only depths multiple of 8 are supported"), (NULL));
return FALSE;
}
sio_initpar (&par);
switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
case GST_AUDIO_FORMAT_S8:
case GST_AUDIO_FORMAT_U8:
case GST_AUDIO_FORMAT_S16LE:
case GST_AUDIO_FORMAT_S16BE:
case GST_AUDIO_FORMAT_U16LE:
case GST_AUDIO_FORMAT_U16BE:
case GST_AUDIO_FORMAT_S32LE:
case GST_AUDIO_FORMAT_S32BE:
case GST_AUDIO_FORMAT_U32LE:
case GST_AUDIO_FORMAT_U32BE:
case GST_AUDIO_FORMAT_S24_32LE:
case GST_AUDIO_FORMAT_S24_32BE:
case GST_AUDIO_FORMAT_U24_32LE:
case GST_AUDIO_FORMAT_U24_32BE:
case GST_AUDIO_FORMAT_S24LE:
case GST_AUDIO_FORMAT_S24BE:
case GST_AUDIO_FORMAT_U24LE:
case GST_AUDIO_FORMAT_U24BE:
break;
default:
GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_READ_WRITE,
("Unsupported audio format"),
("format = %d", GST_AUDIO_INFO_FORMAT (&spec->info)));
return FALSE;
}
par.sig = GST_AUDIO_INFO_IS_SIGNED(&spec->info);
par.bits = GST_AUDIO_INFO_WIDTH(&spec->info);
par.bps = GST_AUDIO_INFO_DEPTH(&spec->info) / 8;
if (par.bps > 1)
par.le = GST_AUDIO_INFO_IS_LITTLE_ENDIAN(&spec->info);
if (par.bits < par.bps * 8)
par.msb = 0;
par.rate = GST_AUDIO_INFO_RATE(&spec->info);
if (sio->mode == SIO_PLAY)
par.pchan = GST_AUDIO_INFO_CHANNELS(&spec->info);
else
par.rchan = GST_AUDIO_INFO_CHANNELS(&spec->info);
par.round = par.rate / 1000000. * spec->latency_time;
par.appbufsz = par.rate / 1000000. * spec->buffer_time;
if (!sio_setpar (sio->hdl, &par)) {
GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_WRITE,
("Unsupported audio encoding"), (NULL));
return FALSE;
}
if (!sio_getpar (sio->hdl, &retpar)) {
GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_WRITE,
("Couldn't get audio device parameters"), (NULL));
return FALSE;
}
#if 0
fprintf(stderr, "format = %s, "
"requested: sig = %d, bits = %d, bps = %d, le = %d, msb = %d, "
"rate = %d, pchan = %d, round = %d, appbufsz = %d; "
"returned: sig = %d, bits = %d, bps = %d, le = %d, msb = %d, "
"rate = %d, pchan = %d, round = %d, appbufsz = %d, bufsz = %d\n",
GST_AUDIO_INFO_NAME(&spec->info),
par.sig, par.bits, par.bps, par.le, par.msb,
par.rate, par.pchan, par.round, par.appbufsz,
retpar.sig, retpar.bits, retpar.bps, retpar.le, retpar.msb,
retpar.rate, retpar.pchan, retpar.round, retpar.appbufsz, retpar.bufsz);
#endif
if (par.bits != retpar.bits ||
par.bps != retpar.bps ||
par.rate != retpar.rate ||
(sio->mode == SIO_PLAY && par.pchan != retpar.pchan) ||
(sio->mode == SIO_REC && par.rchan != retpar.rchan) ||
(par.bps > 1 && par.le != retpar.le) ||
(par.bits < par.bps * 8 && par.msb != retpar.msb)) {
GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_WRITE,
("Audio device refused requested parameters"), (NULL));
return FALSE;
}
nchannels = (sio->mode == SIO_PLAY) ? retpar.pchan : retpar.rchan;
spec->segsize = retpar.round * retpar.bps * nchannels;
spec->segtotal = retpar.bufsz / retpar.round;
sio->bpf = retpar.bps * nchannels;
sio->delay = 0;
sio_onmove (sio->hdl, gst_sndio_cb, sio);
if (!sio_start (sio->hdl)) {
GST_ELEMENT_ERROR (sio->obj, RESOURCE, OPEN_READ_WRITE,
("Could not start sndio"), (NULL));
return FALSE;
}
return TRUE;
}
static gboolean
gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec)
{
gboolean ret = FALSE, is_passthrough = FALSE;
GstOsxAudioRingBuffer *osxbuf;
AudioStreamBasicDescription format;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
if (RINGBUFFER_IS_SPDIF (spec->type)) {
format.mFormatID = kAudioFormat60958AC3;
format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
format.mChannelsPerFrame = 2;
format.mFormatFlags = kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonMixable;
format.mBytesPerFrame = 0;
format.mBitsPerChannel = 16;
format.mBytesPerPacket = 6144;
format.mFramesPerPacket = 1536;
format.mReserved = 0;
spec->segsize = 6144;
spec->segtotal = 10;
is_passthrough = TRUE;
} else {
int width, depth;
/* Fill out the audio description we're going to be using */
format.mFormatID = kAudioFormatLinearPCM;
format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
format.mChannelsPerFrame = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (GST_AUDIO_INFO_IS_FLOAT (&spec->info)) {
format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
width = depth = GST_AUDIO_INFO_WIDTH (&spec->info);
} else {
format.mFormatFlags = kAudioFormatFlagIsSignedInteger;
width = GST_AUDIO_INFO_WIDTH (&spec->info);
depth = GST_AUDIO_INFO_DEPTH (&spec->info);
if (width == depth) {
format.mFormatFlags |= kAudioFormatFlagIsPacked;
} else {
format.mFormatFlags |= kAudioFormatFlagIsAlignedHigh;
}
}
if (GST_AUDIO_INFO_IS_BIG_ENDIAN (&spec->info)) {
format.mFormatFlags |= kAudioFormatFlagIsBigEndian;
}
format.mBytesPerFrame = GST_AUDIO_INFO_BPF (&spec->info);
format.mBitsPerChannel = depth;
format.mBytesPerPacket = GST_AUDIO_INFO_BPF (&spec->info);
format.mFramesPerPacket = 1;
format.mReserved = 0;
spec->segsize =
(spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = spec->buffer_time / spec->latency_time;
is_passthrough = FALSE;
}
GST_DEBUG_OBJECT (osxbuf, "Format: " CORE_AUDIO_FORMAT,
CORE_AUDIO_FORMAT_ARGS (format));
/* gst_audio_ring_buffer_set_channel_positions is not called
* since the AUs perform channel reordering themselves.
* (see gst_core_audio_set_channel_layout) */
buf->size = spec->segtotal * spec->segsize;
buf->memory = g_malloc0 (buf->size);
ret = gst_core_audio_initialize (osxbuf->core_audio, format, spec->caps,
is_passthrough);
if (!ret) {
g_free (buf->memory);
buf->memory = NULL;
buf->size = 0;
}
osxbuf->segoffset = 0;
return ret;
}
