gboolean
gst_pulse_fill_format_info (GstAudioRingBufferSpec * spec, pa_format_info ** f,
guint * channels)
{
pa_format_info *format;
pa_sample_format_t sf = PA_SAMPLE_INVALID;
GstAudioInfo *ainfo = &spec->info;
format = pa_format_info_new ();
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW
&& GST_AUDIO_INFO_WIDTH (ainfo) == 8) {
format->encoding = PA_ENCODING_PCM;
sf = PA_SAMPLE_ULAW;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW
&& GST_AUDIO_INFO_WIDTH (ainfo) == 8) {
format->encoding = PA_ENCODING_PCM;
sf = PA_SAMPLE_ALAW;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
format->encoding = PA_ENCODING_PCM;
if (!gstaudioformat_to_pasampleformat (GST_AUDIO_INFO_FORMAT (ainfo), &sf))
goto fail;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3) {
format->encoding = PA_ENCODING_AC3_IEC61937;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3) {
format->encoding = PA_ENCODING_EAC3_IEC61937;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS) {
format->encoding = PA_ENCODING_DTS_IEC61937;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG) {
format->encoding = PA_ENCODING_MPEG_IEC61937;
} else {
goto fail;
}
if (format->encoding == PA_ENCODING_PCM) {
pa_format_info_set_sample_format (format, sf);
pa_format_info_set_channels (format, GST_AUDIO_INFO_CHANNELS (ainfo));
}
pa_format_info_set_rate (format, GST_AUDIO_INFO_RATE (ainfo));
if (!pa_format_info_valid (format))
goto fail;
*f = format;
*channels = GST_AUDIO_INFO_CHANNELS (ainfo);
return TRUE;
fail:
if (format)
pa_format_info_free (format);
return FALSE;
}
void
gst_audio_parse_update_frame_size (GstAudioParse * ap)
{
gint framesize, width;
switch (ap->format) {
case GST_AUDIO_PARSE_FORMAT_ALAW:
case GST_AUDIO_PARSE_FORMAT_MULAW:
width = 8;
break;
case GST_AUDIO_PARSE_FORMAT_RAW:
default:
{
GstAudioInfo info;
gst_audio_info_init (&info);
/* rate, etc do not really matter here */
gst_audio_info_set_format (&info, ap->raw_format, 44100, ap->channels,
NULL);
width = GST_AUDIO_INFO_WIDTH (&info);
break;
}
}
framesize = (width / 8) * ap->channels;
gst_raw_parse_set_framesize (GST_RAW_PARSE (ap), framesize);
}
static GstFlowReturn
gst_deinterleave_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstDeinterleave *self = GST_DEINTERLEAVE (parent);
GstFlowReturn ret;
g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (GST_AUDIO_INFO_WIDTH (&self->audio_info) > 0,
GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (GST_AUDIO_INFO_CHANNELS (&self->audio_info) > 0,
GST_FLOW_NOT_NEGOTIATED);
ret = gst_deinterleave_process (self, buffer);
if (ret != GST_FLOW_OK)
GST_DEBUG_OBJECT (self, "flow return: %s", gst_flow_get_name (ret));
return ret;
}
static gboolean
gst_audio_segment_clip_set_caps (GstSegmentClip * base, GstCaps * caps)
{
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
gboolean ret;
GstAudioInfo info;
gint rate, channels, width;
gst_audio_info_init (&info);
ret = gst_audio_info_from_caps (&info, caps);
if (ret) {
rate = GST_AUDIO_INFO_RATE (&info);
channels = GST_AUDIO_INFO_CHANNELS (&info);
width = GST_AUDIO_INFO_WIDTH (&info);
GST_DEBUG_OBJECT (self, "Configured: rate %d channels %d width %d",
rate, channels, width);
self->rate = rate;
self->framesize = (width / 8) * channels;
}
return ret;
}
static gboolean
gst_deinterleave_set_process_function (GstDeinterleave * self)
{
switch (GST_AUDIO_INFO_WIDTH (&self->audio_info)) {
case 8:
self->func = (GstDeinterleaveFunc) deinterleave_8;
break;
case 16:
self->func = (GstDeinterleaveFunc) deinterleave_16;
break;
case 24:
self->func = (GstDeinterleaveFunc) deinterleave_24;
break;
case 32:
self->func = (GstDeinterleaveFunc) deinterleave_32;
break;
case 64:
self->func = (GstDeinterleaveFunc) deinterleave_64;
break;
default:
return FALSE;
}
return TRUE;
}
static gboolean
gst_wavenc_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstWavEnc *wavenc;
GstStructure *structure;
const gchar *name;
gint chans, rate;
GstCaps *ccaps;
wavenc = GST_WAVENC (gst_pad_get_parent (pad));
ccaps = gst_pad_get_current_caps (pad);
if (wavenc->sent_header && ccaps && !gst_caps_can_intersect (caps, ccaps)) {
gst_caps_unref (ccaps);
GST_WARNING_OBJECT (wavenc, "cannot change format in middle of stream");
goto fail;
}
if (ccaps)
gst_caps_unref (ccaps);
GST_DEBUG_OBJECT (wavenc, "got caps: %" GST_PTR_FORMAT, caps);
structure = gst_caps_get_structure (caps, 0);
name = gst_structure_get_name (structure);
if (!gst_structure_get_int (structure, "channels", &chans) ||
!gst_structure_get_int (structure, "rate", &rate)) {
GST_WARNING_OBJECT (wavenc, "caps incomplete");
goto fail;
}
if (strcmp (name, "audio/x-raw") == 0) {
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps))
goto fail;
if (GST_AUDIO_INFO_IS_INTEGER (&info))
wavenc->format = GST_RIFF_WAVE_FORMAT_PCM;
else if (GST_AUDIO_INFO_IS_FLOAT (&info))
wavenc->format = GST_RIFF_WAVE_FORMAT_IEEE_FLOAT;
else
goto fail;
wavenc->width = GST_AUDIO_INFO_WIDTH (&info);
} else if (strcmp (name, "audio/x-alaw") == 0) {
wavenc->format = GST_RIFF_WAVE_FORMAT_ALAW;
wavenc->width = 8;
} else if (strcmp (name, "audio/x-mulaw") == 0) {
wavenc->format = GST_RIFF_WAVE_FORMAT_MULAW;
wavenc->width = 8;
} else {
GST_WARNING_OBJECT (wavenc, "Unsupported format %s", name);
goto fail;
}
wavenc->channels = chans;
wavenc->rate = rate;
GST_LOG_OBJECT (wavenc,
"accepted caps: format=0x%04x chans=%u width=%u rate=%u",
wavenc->format, wavenc->channels, wavenc->width, wavenc->rate);
gst_object_unref (wavenc);
return TRUE;
fail:
gst_object_unref (wavenc);
return FALSE;
}
static gboolean
gst_interleave_sink_setcaps (GstInterleave * self, GstPad * pad,
const GstCaps * caps, const GstAudioInfo * info)
{
g_return_val_if_fail (GST_IS_INTERLEAVE_PAD (pad), FALSE);
/* TODO: handle caps changes */
if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps)) {
goto cannot_change_caps;
} else {
GstCaps *srccaps;
GstStructure *s;
gboolean res;
self->width = GST_AUDIO_INFO_WIDTH (info);
self->rate = GST_AUDIO_INFO_RATE (info);
gst_interleave_set_process_function (self);
srccaps = gst_caps_copy (caps);
s = gst_caps_get_structure (srccaps, 0);
gst_structure_remove_field (s, "channel-mask");
gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL);
gst_interleave_set_channel_positions (self, s);
gst_pad_set_active (self->src, TRUE);
res = gst_pad_set_caps (self->src, srccaps);
gst_caps_unref (srccaps);
if (!res)
goto src_did_not_accept;
}
if (!self->sinkcaps) {
GstCaps *sinkcaps = gst_caps_copy (caps);
GstStructure *s = gst_caps_get_structure (sinkcaps, 0);
gst_structure_remove_field (s, "channel-mask");
GST_DEBUG_OBJECT (self, "setting sinkcaps %" GST_PTR_FORMAT, sinkcaps);
gst_caps_replace (&self->sinkcaps, sinkcaps);
gst_caps_unref (sinkcaps);
}
return TRUE;
cannot_change_caps:
{
GST_WARNING_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't "
"change", self->sinkcaps);
return FALSE;
}
src_did_not_accept:
{
GST_WARNING_OBJECT (self, "src did not accept setcaps()");
return FALSE;
}
}
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 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;
}
static GstFlowReturn
gst_deinterleave_process (GstDeinterleave * self, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
guint channels = GST_AUDIO_INFO_CHANNELS (&self->audio_info);
guint pads_pushed = 0, buffers_allocated = 0;
guint nframes =
gst_buffer_get_size (buf) / channels /
(GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
guint bufsize = nframes * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
guint i;
GList *srcs;
GstBuffer **buffers_out = g_new0 (GstBuffer *, channels);
guint8 *in, *out;
GstMapInfo read_info;
GList *pending_events, *l;
/* Send any pending events to all src pads */
GST_OBJECT_LOCK (self);
pending_events = self->pending_events;
self->pending_events = NULL;
GST_OBJECT_UNLOCK (self);
if (pending_events) {
GstEvent *event;
GST_DEBUG_OBJECT (self, "Sending pending events to all src pads");
for (l = pending_events; l; l = l->next) {
event = l->data;
for (srcs = self->srcpads; srcs != NULL; srcs = srcs->next)
gst_pad_push_event (GST_PAD (srcs->data), gst_event_ref (event));
gst_event_unref (event);
}
g_list_free (pending_events);
}
gst_buffer_map (buf, &read_info, GST_MAP_READ);
/* Allocate buffers */
for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
buffers_out[i] = gst_buffer_new_allocate (NULL, bufsize, NULL);
/* Make sure we got a correct buffer. The only other case we allow
* here is an unliked pad */
if (!buffers_out[i])
goto alloc_buffer_failed;
else if (buffers_out[i] && gst_buffer_get_size (buffers_out[i]) != bufsize)
goto alloc_buffer_bad_size;
if (buffers_out[i]) {
gst_buffer_copy_into (buffers_out[i], buf, GST_BUFFER_COPY_METADATA, 0,
-1);
buffers_allocated++;
}
}
/* Return NOT_LINKED if no pad was linked */
if (!buffers_allocated) {
GST_WARNING_OBJECT (self,
"Couldn't allocate any buffers because no pad was linked");
ret = GST_FLOW_NOT_LINKED;
goto done;
}
/* deinterleave */
for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
GstPad *pad = (GstPad *) srcs->data;
GstMapInfo write_info;
in = (guint8 *) read_info.data;
in += i * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
if (buffers_out[i]) {
gst_buffer_map (buffers_out[i], &write_info, GST_MAP_WRITE);
out = (guint8 *) write_info.data;
self->func (out, in, channels, nframes);
gst_buffer_unmap (buffers_out[i], &write_info);
ret = gst_pad_push (pad, buffers_out[i]);
buffers_out[i] = NULL;
if (ret == GST_FLOW_OK)
pads_pushed++;
else if (ret == GST_FLOW_NOT_LINKED)
ret = GST_FLOW_OK;
else
goto push_failed;
}
}
/* Return NOT_LINKED if no pad was linked */
if (!pads_pushed)
ret = GST_FLOW_NOT_LINKED;
GST_DEBUG_OBJECT (self, "Pushed on %d pads", pads_pushed);
done:
gst_buffer_unmap (buf, &read_info);
gst_buffer_unref (buf);
g_free (buffers_out);
return ret;
alloc_buffer_failed:
{
GST_WARNING ("gst_pad_alloc_buffer() returned %s", gst_flow_get_name (ret));
goto clean_buffers;
}
alloc_buffer_bad_size:
{
GST_WARNING ("called alloc_buffer(), but didn't get requested bytes");
ret = GST_FLOW_NOT_NEGOTIATED;
goto clean_buffers;
}
push_failed:
{
GST_DEBUG ("push() failed, flow = %s", gst_flow_get_name (ret));
goto clean_buffers;
}
clean_buffers:
{
gst_buffer_unmap (buf, &read_info);
for (i = 0; i < channels; i++) {
if (buffers_out[i])
gst_buffer_unref (buffers_out[i]);
}
gst_buffer_unref (buf);
g_free (buffers_out);
return ret;
}
}
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;
}
