/* copy data into @dest, skipping @skip bytes from the head buffers */
static void
copy_into_unchecked (GstAdapter * adapter, guint8 * dest, gsize skip,
gsize size)
{
GSList *g;
GstBuffer *buf;
gsize bsize, csize;
/* first step, do skipping */
/* we might well be copying where we were scanning */
if (adapter->scan_entry && (adapter->scan_offset <= skip)) {
g = adapter->scan_entry;
skip -= adapter->scan_offset;
} else {
g = adapter->buflist;
}
buf = g->data;
bsize = gst_buffer_get_size (buf);
while (G_UNLIKELY (skip >= bsize)) {
skip -= bsize;
g = g_slist_next (g);
buf = g->data;
bsize = gst_buffer_get_size (buf);
}
/* copy partial buffer */
csize = MIN (bsize - skip, size);
GST_DEBUG ("bsize %" G_GSIZE_FORMAT ", skip %" G_GSIZE_FORMAT ", csize %"
G_GSIZE_FORMAT, bsize, skip, csize);
GST_CAT_LOG_OBJECT (GST_CAT_PERFORMANCE, adapter, "extract %" G_GSIZE_FORMAT
" bytes", csize);
gst_buffer_extract (buf, skip, dest, csize);
size -= csize;
dest += csize;
/* second step, copy remainder */
while (size > 0) {
g = g_slist_next (g);
buf = g->data;
bsize = gst_buffer_get_size (buf);
if (G_LIKELY (bsize > 0)) {
csize = MIN (bsize, size);
GST_CAT_LOG_OBJECT (GST_CAT_PERFORMANCE, adapter,
"extract %" G_GSIZE_FORMAT " bytes", csize);
gst_buffer_extract (buf, 0, dest, csize);
size -= csize;
dest += csize;
}
}
}
static GstFlowReturn
output_vchain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstClockTime timestamp;
guint8 b;
gboolean jitter = video_gaps || video_overlaps || late_video;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (!jitter)
fail_unless_equals_int64 (timestamp,
(video_buffer_count % n_vbuffers) * 25 * GST_MSECOND);
timestamp =
gst_segment_to_stream_time (¤t_video_segment, GST_FORMAT_TIME,
timestamp);
if (!jitter)
fail_unless_equals_int64 (timestamp,
(video_buffer_count % n_vbuffers) * 25 * GST_MSECOND);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
timestamp =
gst_segment_to_running_time (¤t_video_segment, GST_FORMAT_TIME,
timestamp);
if (!jitter)
fail_unless_equals_int64 (timestamp, video_buffer_count * 25 * GST_MSECOND);
gst_buffer_extract (buffer, 0, &b, 1);
if (!jitter)
fail_unless_equals_int (b, video_buffer_count % n_vbuffers);
video_buffer_count++;
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static GstFlowReturn
test_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
guint8 data[2];
gst_buffer_extract (frame->buffer, 0, data, 2);
if ((GST_READ_UINT16_BE (data) & 0xffe0) == 0xffe0) {
if (GST_BUFFER_OFFSET (frame->buffer) == 0) {
GstCaps *caps;
caps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1,
"mpegaudioversion", G_TYPE_INT, 1, "layer", G_TYPE_INT, 3,
"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL);
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
}
/* this framesize is hard-coded for ../test.mp3 */
return gst_base_parse_finish_frame (parse, frame, 1045);
} else {
*skipsize = 1;
return GST_FLOW_OK;
}
}
static GstFlowReturn
gst_kate_parse_parse_packet (GstKateParse * parse, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
guint8 header[1];
gsize size;
g_assert (parse);
parse->packetno++;
size = gst_buffer_extract (buf, 0, header, 1);
GST_LOG_OBJECT (parse, "Got packet %02x, %" G_GSIZE_FORMAT " bytes",
size ? header[0] : -1, gst_buffer_get_size (buf));
if (size > 0 && header[0] & 0x80) {
GST_DEBUG_OBJECT (parse, "Found header %02x", header[0]);
/* if 0x80 is set, it's streamheader,
* so put it on the streamheader list and return */
parse->streamheader = g_list_append (parse->streamheader, buf);
ret = GST_FLOW_OK;
} else {
if (!parse->streamheader_sent) {
GST_DEBUG_OBJECT (parse, "Found non header, pushing headers seen so far");
ret = gst_kate_parse_push_headers (parse);
}
if (ret == GST_FLOW_OK) {
ret = gst_kate_parse_queue_buffer (parse, buf);
}
}
return ret;
}
/**
* gst_vaapi_profile_from_codec_data:
* @codec: a #GstVaapiCodec
* @buffer: a #GstBuffer holding code data
*
* Tries to parse VA profile from @buffer data and @codec information.
*
* Return value: the #GstVaapiProfile described in @buffer
*/
static GstVaapiProfile
gst_vaapi_profile_from_codec_data_h264 (GstBuffer * buffer)
{
/* MPEG-4 Part 15: Advanced Video Coding (AVC) file format */
guchar buf[3];
if (gst_buffer_extract (buffer, 0, buf, sizeof (buf)) != sizeof (buf))
return 0;
if (buf[0] != 1) /* configurationVersion = 1 */
return 0;
switch (buf[1]) { /* AVCProfileIndication */
case 66:
return ((buf[2] & 0x40) ?
GST_VAAPI_PROFILE_H264_CONSTRAINED_BASELINE :
GST_VAAPI_PROFILE_H264_BASELINE);
case 77:
return GST_VAAPI_PROFILE_H264_MAIN;
case 100:
return GST_VAAPI_PROFILE_H264_HIGH;
case 118:
return GST_VAAPI_PROFILE_H264_MULTIVIEW_HIGH;
case 128:
return GST_VAAPI_PROFILE_H264_STEREO_HIGH;
}
return 0;
}
static GstFlowReturn
gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)
{
GstFlowReturn ret;
GstBuffer *buf, *outbuf;
guint8 *payload, *spayload;
guint payload_len;
GstClockTime duration;
GstRTPBuffer rtp = { NULL, };
payload_len = rtpceltpay->bytes + rtpceltpay->sbytes;
duration = rtpceltpay->qduration;
GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (rtpceltpay->qduration));
/* get a big enough packet for the sizes + payloads */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
GST_BUFFER_DURATION (outbuf) = duration;
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* point to the payload for size headers and data */
spayload = gst_rtp_buffer_get_payload (&rtp);
payload = spayload + rtpceltpay->sbytes;
while ((buf = g_queue_pop_head (rtpceltpay->queue))) {
guint size;
/* copy first timestamp to output */
if (GST_BUFFER_TIMESTAMP (outbuf) == -1)
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
/* write the size to the header */
size = gst_buffer_get_size (buf);
while (size > 0xff) {
*spayload++ = 0xff;
size -= 0xff;
}
*spayload++ = size;
/* copy payload */
size = gst_buffer_get_size (buf);
gst_buffer_extract (buf, 0, payload, size);
payload += size;
gst_buffer_unref (buf);
}
gst_rtp_buffer_unmap (&rtp);
/* we consumed it all */
rtpceltpay->bytes = 0;
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf);
return ret;
}
static GstFlowReturn
gst_shm_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstShmSink *self = GST_SHM_SINK (bsink);
int rv;
GstMapInfo map;
GST_OBJECT_LOCK (self);
while (self->wait_for_connection && !self->clients) {
g_cond_wait (self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
GST_OBJECT_UNLOCK (self);
return GST_FLOW_FLUSHING;
}
}
while (!gst_shm_sink_can_render (self, GST_BUFFER_TIMESTAMP (buf))) {
g_cond_wait (self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
GST_OBJECT_UNLOCK (self);
return GST_FLOW_FLUSHING;
}
}
gst_buffer_map (buf, &map, GST_MAP_READ);
rv = sp_writer_send_buf (self->pipe, (char *) map.data, map.size,
GST_BUFFER_TIMESTAMP (buf));
gst_buffer_unmap (buf, &map);
if (rv == -1) {
ShmBlock *block = NULL;
gchar *shmbuf = NULL;
while ((block = sp_writer_alloc_block (self->pipe,
gst_buffer_get_size (buf))) == NULL) {
g_cond_wait (self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
GST_OBJECT_UNLOCK (self);
return GST_FLOW_FLUSHING;
}
}
while (self->wait_for_connection && !self->clients) {
g_cond_wait (self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
sp_writer_free_block (block);
GST_OBJECT_UNLOCK (self);
return GST_FLOW_FLUSHING;
}
}
shmbuf = sp_writer_block_get_buf (block);
gst_buffer_extract (buf, 0, shmbuf, gst_buffer_get_size (buf));
sp_writer_send_buf (self->pipe, shmbuf, gst_buffer_get_size (buf),
GST_BUFFER_TIMESTAMP (buf));
sp_writer_free_block (block);
}
GST_OBJECT_UNLOCK (self);
return GST_FLOW_OK;
}
/* submit the given buffer to the ogg sync */
static GstFlowReturn
gst_ogg_parse_submit_buffer (GstOggParse * ogg, GstBuffer * buffer)
{
gsize size;
gchar *oggbuffer;
GstFlowReturn ret = GST_FLOW_OK;
size = gst_buffer_get_size (buffer);
GST_DEBUG_OBJECT (ogg, "submitting %" G_GSIZE_FORMAT " bytes", size);
if (G_UNLIKELY (size == 0))
goto done;
oggbuffer = ogg_sync_buffer (&ogg->sync, size);
if (G_UNLIKELY (oggbuffer == NULL)) {
GST_ELEMENT_ERROR (ogg, STREAM, DECODE,
(NULL), ("failed to get ogg sync buffer"));
ret = GST_FLOW_ERROR;
goto done;
}
size = gst_buffer_extract (buffer, 0, oggbuffer, size);
if (G_UNLIKELY (ogg_sync_wrote (&ogg->sync, size) < 0)) {
GST_ELEMENT_ERROR (ogg, STREAM, DECODE, (NULL),
("failed to write %" G_GSIZE_FORMAT " bytes to the sync buffer", size));
ret = GST_FLOW_ERROR;
}
done:
gst_buffer_unref (buffer);
return ret;
}
static GstVaapiProfile
gst_vaapi_profile_from_codec_data_h265 (GstBuffer * buffer)
{
/* ISO/IEC 14496-15: HEVC file format */
guchar buf[3];
if (gst_buffer_extract (buffer, 0, buf, sizeof (buf)) != sizeof (buf))
return 0;
if (buf[0] != 1) /* configurationVersion = 1 */
return 0;
if (buf[1] & 0xc0) /* general_profile_space = 0 */
return 0;
switch (buf[1] & 0x1f) { /* HEVCProfileIndication */
case 1:
return GST_VAAPI_PROFILE_H265_MAIN;
case 2:
return GST_VAAPI_PROFILE_H265_MAIN10;
case 3:
return GST_VAAPI_PROFILE_H265_MAIN_STILL_PICTURE;
case 4:
return GST_VAAPI_PROFILE_H265_MAIN_422_10;
case 5:
return GST_VAAPI_PROFILE_H265_MAIN_444;
}
return 0;
}
static GstFlowReturn
gst_atdec_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer)
{
AudioTimeStamp timestamp = { 0 };
AudioStreamPacketDescription packet;
AudioQueueBufferRef input_buffer, output_buffer;
GstBuffer *out;
GstMapInfo info;
GstAudioInfo *audio_info;
int size, out_frames;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstATDec *atdec = GST_ATDEC (decoder);
if (buffer == NULL)
return GST_FLOW_OK;
audio_info = gst_audio_decoder_get_audio_info (decoder);
/* copy the input buffer into an AudioQueueBuffer */
size = gst_buffer_get_size (buffer);
AudioQueueAllocateBuffer (atdec->queue, size, &input_buffer);
gst_buffer_extract (buffer, 0, input_buffer->mAudioData, size);
input_buffer->mAudioDataByteSize = size;
/* assume framed input */
packet.mStartOffset = 0;
packet.mVariableFramesInPacket = 1;
packet.mDataByteSize = size;
/* enqueue the buffer. It will get free'd once the gst_atdec_buffer_emptied
* callback is called
*/
AudioQueueEnqueueBuffer (atdec->queue, input_buffer, 1, &packet);
/* figure out how many frames we need to pull out of the queue */
out_frames = GST_CLOCK_TIME_TO_FRAMES (GST_BUFFER_DURATION (buffer),
audio_info->rate);
size = out_frames * audio_info->bpf;
AudioQueueAllocateBuffer (atdec->queue, size, &output_buffer);
/* pull the frames */
AudioQueueOfflineRender (atdec->queue, ×tamp, output_buffer, out_frames);
if (output_buffer->mAudioDataByteSize) {
out =
gst_audio_decoder_allocate_output_buffer (decoder,
output_buffer->mAudioDataByteSize);
gst_buffer_map (out, &info, GST_MAP_WRITE);
memcpy (info.data, output_buffer->mAudioData,
output_buffer->mAudioDataByteSize);
gst_buffer_unmap (out, &info);
flow_ret = gst_audio_decoder_finish_frame (decoder, out, 1);
}
AudioQueueFreeBuffer (atdec->queue, output_buffer);
return flow_ret;
}
static GstFlowReturn
gst_quarktv_transform_frame (GstVideoFilter * vfilter, GstVideoFrame * in_frame,
GstVideoFrame * out_frame)
{
GstQuarkTV *filter = GST_QUARKTV (vfilter);
gint area;
guint32 *src, *dest;
GstClockTime timestamp;
GstBuffer **planetable;
gint planes, current_plane;
timestamp = GST_BUFFER_TIMESTAMP (in_frame->buffer);
timestamp =
gst_segment_to_stream_time (&GST_BASE_TRANSFORM (vfilter)->segment,
GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (GST_OBJECT (filter), timestamp);
if (G_UNLIKELY (filter->planetable == NULL))
return GST_FLOW_FLUSHING;
src = GST_VIDEO_FRAME_PLANE_DATA (in_frame, 0);
dest = GST_VIDEO_FRAME_PLANE_DATA (out_frame, 0);
GST_OBJECT_LOCK (filter);
area = filter->area;
planetable = filter->planetable;
planes = filter->planes;
current_plane = filter->current_plane;
if (planetable[current_plane])
gst_buffer_unref (planetable[current_plane]);
planetable[current_plane] = gst_buffer_ref (in_frame->buffer);
/* For each pixel */
while (--area) {
GstBuffer *rand;
/* pick a random buffer */
rand = planetable[(current_plane + (fastrand () >> 24)) % planes];
/* Copy the pixel from the random buffer to dest, FIXME, slow */
if (rand)
gst_buffer_extract (rand, area * 4, &dest[area], 4);
else
dest[area] = src[area];
}
filter->current_plane--;
if (filter->current_plane < 0)
filter->current_plane = planes - 1;
GST_OBJECT_UNLOCK (filter);
return GST_FLOW_OK;
}
GstBuffer *
mpegpsmux_prepare_aac (GstBuffer * buf, MpegPsPadData * data, MpegPsMux * mux)
{
GstBuffer *out_buf;
GstMemory *mem;
gsize out_size;
guint8 *adts_header, codec_data[2];
guint8 rate_idx = 0, channels = 0, obj_type = 0;
GST_DEBUG_OBJECT (mux, "Preparing AAC buffer for output");
adts_header = g_malloc0 (7);
/* We want the same data and metadata, and then prepend some bytes */
out_buf = gst_buffer_copy (buf);
out_size = gst_buffer_get_size (buf) + 7;
gst_buffer_extract (data->codec_data, 0, codec_data, 2);
/* Generate ADTS header */
obj_type = (codec_data[0] & 0xC) >> 2;
obj_type++;
rate_idx = (codec_data[0] & 0x3) << 1;
rate_idx |= (codec_data[1] & 0x80) >> 7;
channels = (codec_data[1] & 0x78) >> 3;
GST_DEBUG_OBJECT (mux, "Rate index %u, channels %u, object type %u", rate_idx,
channels, obj_type);
/* Sync point over a full byte */
adts_header[0] = 0xFF;
/* Sync point continued over first 4 bits + static 4 bits
* (ID, layer, protection)*/
adts_header[1] = 0xF1;
/* Object type over first 2 bits */
adts_header[2] = obj_type << 6;
/* rate index over next 4 bits */
adts_header[2] |= (rate_idx << 2);
/* channels over last 2 bits */
adts_header[2] |= (channels & 0x4) >> 2;
/* channels continued over next 2 bits + 4 bits at zero */
adts_header[3] = (channels & 0x3) << 6;
/* frame size over last 2 bits */
adts_header[3] |= (gst_buffer_get_size (out_buf) & 0x1800) >> 11;
/* frame size continued over full byte */
adts_header[4] = (out_size & 0x1FF8) >> 3;
/* frame size continued first 3 bits */
adts_header[5] = (out_size & 0x7) << 5;
/* buffer fullness (0x7FF for VBR) over 5 last bits */
adts_header[5] |= 0x1F;
/* buffer fullness (0x7FF for VBR) continued over 6 first bits + 2 zeros for
* number of raw data blocks */
adts_header[6] = 0xFC;
/* Prepend ADTS header */
mem = gst_memory_new_wrapped (0, adts_header, 7, 0, 7, adts_header, g_free);
gst_buffer_prepend_memory (out_buf, mem);
return out_buf;
}
static guint8
buffer_get_byte (GstBuffer * buffer, gint offset)
{
guint8 res;
gst_buffer_extract (buffer, offset, &res, 1);
return res;
}
static void copyGstreamerBuffersToAudioChannel(GstBufferList* buffers, AudioChannel* audioChannel)
{
float* destination = audioChannel->mutableData();
unsigned bufferCount = gst_buffer_list_length(buffers);
for (unsigned i = 0; i < bufferCount; ++i) {
GstBuffer* buffer = gst_buffer_list_get(buffers, i);
ASSERT(buffer);
gsize bufferSize = gst_buffer_get_size(buffer);
gst_buffer_extract(buffer, 0, destination, bufferSize);
destination += bufferSize / sizeof(float);
}
}
static gboolean
is_keyframe (GstBuffer * buf)
{
gsize size;
guint8 data[1];
size = gst_buffer_get_size (buf);
if (size == 0)
return FALSE;
gst_buffer_extract (buf, 0, data, 1);
return ((data[0] & 0x40) == 0);
}
static gchar *
gst_vqe_buf_to_utf8_sdp (GstBuffer * buf)
{
gsize size = gst_buffer_get_size(buf);
gchar *sdp = g_malloc(size + 1);
sdp[size] = '\0';
gst_buffer_extract(buf, 0, sdp, size);
if (!g_utf8_validate (sdp, size, NULL))
goto validate_error;
return sdp;
validate_error:
g_free(sdp);
return NULL;
}
void GStreamerReader::CopyIntoImageBuffer(GstBuffer* aBuffer,
GstBuffer** aOutBuffer,
nsRefPtr<PlanarYCbCrImage> &image)
{
*aOutBuffer = gst_buffer_new_allocate(mAllocator, gst_buffer_get_size(aBuffer), nullptr);
GstMemory *mem = gst_buffer_peek_memory(*aOutBuffer, 0);
GstMapInfo map_info;
gst_memory_map(mem, &map_info, GST_MAP_WRITE);
gst_buffer_extract(aBuffer, 0, map_info.data, gst_buffer_get_size(aBuffer));
gst_memory_unmap(mem, &map_info);
/* create a new gst buffer with the newly created memory and copy the
* metadata over from the incoming buffer */
gst_buffer_copy_into(*aOutBuffer, aBuffer,
(GstBufferCopyFlags)(GST_BUFFER_COPY_METADATA), 0, -1);
image = GetImageFromBuffer(*aOutBuffer);
}
static int
gst_ffmpegdata_peek (URLContext * h, unsigned char *buf, int size)
{
GstProtocolInfo *info;
GstBuffer *inbuf = NULL;
GstFlowReturn ret;
int total = 0;
g_return_val_if_fail (h->flags == URL_RDONLY, AVERROR (EIO));
info = (GstProtocolInfo *) h->priv_data;
GST_DEBUG ("Pulling %d bytes at position %" G_GUINT64_FORMAT, size,
info->offset);
ret = gst_pad_pull_range (info->pad, info->offset, (guint) size, &inbuf);
switch (ret) {
case GST_FLOW_OK:
total = (gint) gst_buffer_get_size (inbuf);
gst_buffer_extract (inbuf, 0, buf, total);
gst_buffer_unref (inbuf);
break;
case GST_FLOW_EOS:
total = 0;
break;
case GST_FLOW_FLUSHING:
total = -1;
break;
default:
case GST_FLOW_ERROR:
total = -2;
break;
}
GST_DEBUG ("Got %d (%s) return result %d", ret, gst_flow_get_name (ret),
total);
return total;
}
GstFlowReturn GStreamerImageStream::on_new_sample(GstAppSink *appsink, GStreamerImageStream *user_data)
{
// get the buffer from appsink
GstSample *sample = gst_app_sink_pull_sample(appsink);
GstBuffer *buffer = gst_sample_get_buffer(sample);
// upload data
GstMapInfo info;
gst_buffer_map(buffer, &info, GST_MAP_READ);
gst_buffer_extract(buffer, 0, user_data->_internal_buffer, info.size);
// data has been modified so dirty the image so the texture will updated
user_data->dirty();
// clean resources
gst_buffer_unmap(buffer, &info);
gst_sample_unref(sample);
return GST_FLOW_OK;
}
static GstFlowReturn
output_achain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstClockTime timestamp;
guint8 b;
gboolean audio_jitter = audio_nondiscont || audio_drift || early_video;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp,
(audio_buffer_count % n_abuffers) * 1 * GST_SECOND);
timestamp =
gst_segment_to_stream_time (¤t_audio_segment, GST_FORMAT_TIME,
timestamp);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp,
(audio_buffer_count % n_abuffers) * 1 * GST_SECOND);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
timestamp =
gst_segment_to_running_time (¤t_audio_segment, GST_FORMAT_TIME,
timestamp);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp, audio_buffer_count * 1 * GST_SECOND);
gst_buffer_extract (buffer, 0, &b, 1);
if (per_channel) {
fail_unless_equals_int (b, fill_value_per_channel[0]);
} else {
fail_unless_equals_int (b, fill_value);
}
audio_buffer_count++;
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_ogg_avi_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstFlowReturn result = GST_FLOW_OK;
GstOggAviParse *ogg;
guint size;
gchar *oggbuf;
gint ret = -1;
ogg = GST_OGG_AVI_PARSE (parent);
size = gst_buffer_get_size (buffer);
GST_LOG_OBJECT (ogg, "Chain function received buffer of size %d", size);
if (GST_BUFFER_IS_DISCONT (buffer)) {
ogg_sync_reset (&ogg->sync);
ogg->discont = TRUE;
}
/* write data to sync layer */
oggbuf = ogg_sync_buffer (&ogg->sync, size);
gst_buffer_extract (buffer, 0, oggbuf, size);
ogg_sync_wrote (&ogg->sync, size);
gst_buffer_unref (buffer);
/* try to get as many packets out of the stream as possible */
do {
ogg_page page;
/* try to swap out a page */
ret = ogg_sync_pageout (&ogg->sync, &page);
if (ret == 0) {
GST_DEBUG_OBJECT (ogg, "need more data");
break;
} else if (ret == -1) {
GST_DEBUG_OBJECT (ogg, "discont in pages");
ogg->discont = TRUE;
} else {
/* new unknown stream, init the ogg stream with the serial number of the
* page. */
if (ogg->serial == -1) {
ogg->serial = ogg_page_serialno (&page);
ogg_stream_init (&ogg->stream, ogg->serial);
}
/* submit page */
if (ogg_stream_pagein (&ogg->stream, &page) != 0) {
GST_WARNING_OBJECT (ogg, "ogg stream choked on page resetting stream");
ogg_sync_reset (&ogg->sync);
ogg->discont = TRUE;
continue;
}
/* try to get as many packets as possible out of the page */
do {
ogg_packet packet;
ret = ogg_stream_packetout (&ogg->stream, &packet);
GST_LOG_OBJECT (ogg, "packetout gave %d", ret);
switch (ret) {
case 0:
break;
case -1:
/* out of sync, We mark a DISCONT. */
ogg->discont = TRUE;
break;
case 1:
result = gst_ogg_avi_parse_push_packet (ogg, &packet);
if (result != GST_FLOW_OK)
goto done;
break;
default:
GST_WARNING_OBJECT (ogg,
"invalid return value %d for ogg_stream_packetout, resetting stream",
ret);
break;
}
}
while (ret != 0);
}
}
while (ret != 0);
done:
return result;
}
static GstFlowReturn
gst_dvd_spu_subpic_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstDVDSpu *dvdspu = (GstDVDSpu *) parent;
GstFlowReturn ret = GST_FLOW_OK;
gsize size;
g_return_val_if_fail (dvdspu != NULL, GST_FLOW_ERROR);
GST_INFO_OBJECT (dvdspu, "Have subpicture buffer with timestamp %"
GST_TIME_FORMAT " and size %" G_GSIZE_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), gst_buffer_get_size (buf));
DVD_SPU_LOCK (dvdspu);
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
dvdspu->subp_seg.position = GST_BUFFER_TIMESTAMP (buf);
}
if (GST_BUFFER_IS_DISCONT (buf) && dvdspu->partial_spu) {
gst_buffer_unref (dvdspu->partial_spu);
dvdspu->partial_spu = NULL;
}
if (dvdspu->partial_spu != NULL) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf))
GST_WARNING_OBJECT (dvdspu,
"Joining subpicture buffer with timestamp to previous");
dvdspu->partial_spu = gst_buffer_append (dvdspu->partial_spu, buf);
} else {
/* If we don't yet have a buffer, wait for one with a timestamp,
* since that will avoid collecting the 2nd half of a partial buf */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf))
dvdspu->partial_spu = buf;
else
gst_buffer_unref (buf);
}
if (dvdspu->partial_spu == NULL)
goto done;
size = gst_buffer_get_size (dvdspu->partial_spu);
switch (dvdspu->spu_input_type) {
case SPU_INPUT_TYPE_VOBSUB:
if (size >= 2) {
guint8 header[2];
guint16 packet_size;
gst_buffer_extract (dvdspu->partial_spu, 0, header, 2);
packet_size = GST_READ_UINT16_BE (header);
if (packet_size == size) {
submit_new_spu_packet (dvdspu, dvdspu->partial_spu);
dvdspu->partial_spu = NULL;
} else if (packet_size == 0) {
GST_LOG_OBJECT (dvdspu, "Discarding empty SPU buffer");
gst_buffer_unref (dvdspu->partial_spu);
dvdspu->partial_spu = NULL;
} else if (packet_size < size) {
/* Somehow we collected too much - something is wrong. Drop the
* packet entirely and wait for a new one */
GST_DEBUG_OBJECT (dvdspu,
"Discarding invalid SPU buffer of size %" G_GSIZE_FORMAT, size);
gst_buffer_unref (dvdspu->partial_spu);
dvdspu->partial_spu = NULL;
} else {
GST_LOG_OBJECT (dvdspu,
"SPU buffer claims to be of size %u. Collected %" G_GSIZE_FORMAT
" so far.", packet_size, size);
}
}
break;
case SPU_INPUT_TYPE_PGS:{
/* Collect until we have a command buffer that ends exactly at the size
* we've collected */
guint8 packet_type;
guint16 packet_size;
GstMapInfo map;
guint8 *ptr, *end;
gboolean invalid = FALSE;
gst_buffer_map (dvdspu->partial_spu, &map, GST_MAP_READ);
ptr = map.data;
end = ptr + map.size;
/* FIXME: There's no need to walk the command set each time. We can set a
* marker and resume where we left off next time */
/* FIXME: Move the packet parsing and sanity checking into the format-specific modules */
while (ptr != end) {
if (ptr + 3 > end)
break;
packet_type = *ptr++;
packet_size = GST_READ_UINT16_BE (ptr);
ptr += 2;
if (ptr + packet_size > end)
break;
ptr += packet_size;
/* 0x80 is the END command for PGS packets */
if (packet_type == 0x80 && ptr != end) {
/* Extra cruft on the end of the packet -> assume invalid */
invalid = TRUE;
break;
}
}
gst_buffer_unmap (dvdspu->partial_spu, &map);
if (invalid) {
gst_buffer_unref (dvdspu->partial_spu);
dvdspu->partial_spu = NULL;
} else if (ptr == end) {
GST_DEBUG_OBJECT (dvdspu,
"Have complete PGS packet of size %" G_GSIZE_FORMAT ". Enqueueing.",
map.size);
submit_new_spu_packet (dvdspu, dvdspu->partial_spu);
dvdspu->partial_spu = NULL;
}
break;
}
default:
GST_ERROR_OBJECT (dvdspu, "Input type not configured before SPU passing");
goto caps_not_set;
}
done:
DVD_SPU_UNLOCK (dvdspu);
return ret;
/* ERRORS */
caps_not_set:
{
GST_ELEMENT_ERROR (dvdspu, RESOURCE, NO_SPACE_LEFT,
(_("Subpicture format was not configured before data flow")), (NULL));
ret = GST_FLOW_ERROR;
goto done;
}
}
static gboolean
gst_amc_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstAmcAudioDec *self;
GstStructure *s;
GstAmcFormat *format;
const gchar *mime;
gboolean is_format_change = FALSE;
gboolean needs_disable = FALSE;
gchar *format_string;
gint rate, channels;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
/* Check if the caps change is a real format change or if only irrelevant
* parts of the caps have changed or nothing at all.
*/
is_format_change |= (!self->input_caps
|| !gst_caps_is_equal (self->input_caps, caps));
needs_disable = self->started;
/* If the component is not started and a real format change happens
* we have to restart the component. If no real format change
* happened we can just exit here.
*/
if (needs_disable && !is_format_change) {
/* Framerate or something minor changed */
self->input_caps_changed = TRUE;
GST_DEBUG_OBJECT (self,
"Already running and caps did not change the format");
return TRUE;
}
if (needs_disable && is_format_change) {
gst_amc_audio_dec_drain (self);
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_amc_audio_dec_stop (GST_AUDIO_DECODER (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_amc_audio_dec_close (GST_AUDIO_DECODER (self));
if (!gst_amc_audio_dec_open (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to open codec again");
return FALSE;
}
if (!gst_amc_audio_dec_start (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to start codec again");
}
}
/* srcpad task is not running at this point */
mime = caps_to_mime (caps);
if (!mime) {
GST_ERROR_OBJECT (self, "Failed to convert caps to mime");
return FALSE;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "rate", &rate) ||
!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self, "Failed to get rate/channels");
return FALSE;
}
format = gst_amc_format_new_audio (mime, rate, channels, &err);
if (!format) {
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
/* FIXME: These buffers needs to be valid until the codec is stopped again */
g_list_foreach (self->codec_datas, (GFunc) gst_buffer_unref, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
if (gst_structure_has_field (s, "codec_data")) {
const GValue *h = gst_structure_get_value (s, "codec_data");
GstBuffer *codec_data = gst_value_get_buffer (h);
GstMapInfo minfo;
guint8 *data;
gst_buffer_map (codec_data, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, "csd-0", data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (codec_data, &minfo);
} else if (gst_structure_has_field (s, "streamheader")) {
const GValue *sh = gst_structure_get_value (s, "streamheader");
gint nsheaders = gst_value_array_get_size (sh);
GstBuffer *buf;
const GValue *h;
gint i, j;
gchar *fname;
GstMapInfo minfo;
guint8 *data;
for (i = 0, j = 0; i < nsheaders; i++) {
h = gst_value_array_get_value (sh, i);
buf = gst_value_get_buffer (h);
if (strcmp (mime, "audio/vorbis") == 0) {
guint8 header_type;
gst_buffer_extract (buf, 0, &header_type, 1);
/* Only use the identification and setup packets */
if (header_type != 0x01 && header_type != 0x05)
continue;
}
fname = g_strdup_printf ("csd-%d", j);
gst_buffer_map (buf, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, fname, data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (buf, &minfo);
g_free (fname);
j++;
}
}
format_string = gst_amc_format_to_string (format, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
GST_DEBUG_OBJECT (self, "Configuring codec with format: %s",
GST_STR_NULL (format_string));
g_free (format_string);
if (!gst_amc_codec_configure (self->codec, format, 0, &err)) {
GST_ERROR_OBJECT (self, "Failed to configure codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
gst_amc_format_free (format);
if (!gst_amc_codec_start (self->codec, &err)) {
GST_ERROR_OBJECT (self, "Failed to start codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
self->spf = -1;
/* TODO: Implement for other codecs too */
if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion = -1;
gst_structure_get_int (s, "mpegversion", &mpegversion);
if (mpegversion == 1) {
gint layer = -1, mpegaudioversion = -1;
gst_structure_get_int (s, "layer", &layer);
gst_structure_get_int (s, "mpegaudioversion", &mpegaudioversion);
if (layer == 1)
self->spf = 384;
else if (layer == 2)
self->spf = 1152;
else if (layer == 3 && mpegaudioversion != -1)
self->spf = (mpegaudioversion == 1 ? 1152 : 576);
}
}
self->started = TRUE;
self->input_caps_changed = TRUE;
/* Start the srcpad loop again */
self->flushing = FALSE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
return TRUE;
}
GstFlowReturn
gst_kate_util_decoder_base_chain_kate_packet (GstKateDecoderBase * decoder,
GstElement * element, GstPad * pad, GstBuffer * buf, GstPad * srcpad,
GstPad * tagpad, GstCaps ** src_caps, const kate_event ** ev)
{
kate_packet kp;
int ret;
GstFlowReturn rflow = GST_FLOW_OK;
gboolean is_header;
guint8 *data;
gsize size;
guint8 header[1];
size = gst_buffer_extract (buf, 0, header, 1);
GST_DEBUG_OBJECT (element, "got kate packet, %u bytes, type %02x",
gst_buffer_get_size (buf), size == 0 ? -1 : header[0]);
is_header = size > 0 && (header[0] & 0x80);
if (!is_header && decoder->tags) {
/* after we've processed headers, send any tags before processing the data packet */
GST_DEBUG_OBJECT (element, "Not a header, sending tags for pad %s:%s",
GST_DEBUG_PAD_NAME (tagpad));
gst_element_found_tags_for_pad (element, tagpad, decoder->tags);
decoder->tags = NULL;
}
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
kate_packet_wrap (&kp, size, data);
ret = kate_high_decode_packetin (&decoder->k, &kp, ev);
gst_buffer_unmap (buf, data, size);
if (G_UNLIKELY (ret < 0)) {
GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
("Failed to decode Kate packet: %s",
gst_kate_util_get_error_message (ret)));
return GST_FLOW_ERROR;
}
if (G_UNLIKELY (ret > 0)) {
GST_DEBUG_OBJECT (element,
"kate_high_decode_packetin has received EOS packet");
}
/* headers may be interesting to retrieve information from */
if (G_UNLIKELY (is_header)) {
switch (header[0]) {
case 0x80: /* ID header */
GST_INFO_OBJECT (element, "Parsed ID header: language %s, category %s",
decoder->k.ki->language, decoder->k.ki->category);
if (src_caps) {
if (*src_caps) {
gst_caps_unref (*src_caps);
*src_caps = NULL;
}
if (strcmp (decoder->k.ki->category, "K-SPU") == 0 ||
strcmp (decoder->k.ki->category, "spu-subtitles") == 0) {
*src_caps = gst_caps_new_empty_simple ("video/x-dvd-subpicture");
} else if (decoder->k.ki->text_markup_type == kate_markup_none) {
*src_caps = gst_caps_new_empty_simple ("text/plain");
} else {
*src_caps = gst_caps_new_empty_simple ("text/x-pango-markup");
}
GST_INFO_OBJECT (srcpad, "Setting caps: %" GST_PTR_FORMAT, *src_caps);
if (!gst_pad_set_caps (srcpad, *src_caps)) {
GST_ERROR_OBJECT (srcpad, "Failed to set caps %" GST_PTR_FORMAT,
*src_caps);
}
}
if (decoder->k.ki->language && *decoder->k.ki->language) {
GstTagList *old = decoder->tags, *tags = gst_tag_list_new_empty ();
if (tags) {
gchar *lang_code;
/* en_GB -> en */
lang_code = g_ascii_strdown (decoder->k.ki->language, -1);
g_strdelimit (lang_code, NULL, '\0');
gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_LANGUAGE_CODE,
lang_code, NULL);
g_free (lang_code);
/* TODO: category - where should it go ? */
decoder->tags =
gst_tag_list_merge (decoder->tags, tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_free (tags);
if (old)
gst_tag_list_free (old);
}
}
/* update properties */
if (decoder->language)
g_free (decoder->language);
decoder->language = g_strdup (decoder->k.ki->language);
if (decoder->category)
g_free (decoder->category);
decoder->category = g_strdup (decoder->k.ki->category);
decoder->original_canvas_width = decoder->k.ki->original_canvas_width;
decoder->original_canvas_height = decoder->k.ki->original_canvas_height;
/* we can now send away any event we've delayed, as the src pad now has caps */
gst_kate_util_decoder_base_drain_event_queue (decoder);
break;
case 0x81: /* Vorbis comments header */
GST_INFO_OBJECT (element, "Parsed comments header");
{
gchar *encoder = NULL;
GstTagList *old = decoder->tags, *list =
gst_tag_list_from_vorbiscomment_buffer (buf,
(const guint8 *) "\201kate\0\0\0\0", 9, &encoder);
if (list) {
decoder->tags =
gst_tag_list_merge (decoder->tags, list, GST_TAG_MERGE_REPLACE);
gst_tag_list_free (list);
}
if (!decoder->tags) {
GST_ERROR_OBJECT (element, "failed to decode comment header");
decoder->tags = gst_tag_list_new_empty ();
}
if (encoder) {
gst_tag_list_add (decoder->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER, encoder, NULL);
g_free (encoder);
}
gst_tag_list_add (decoder->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_SUBTITLE_CODEC, "Kate", NULL);
gst_tag_list_add (decoder->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER_VERSION, decoder->k.ki->bitstream_version_major,
NULL);
if (old)
gst_tag_list_free (old);
if (decoder->initialized) {
gst_element_found_tags_for_pad (element, tagpad, decoder->tags);
decoder->tags = NULL;
} else {
/* Only push them as messages for the time being. *
* They will be pushed on the pad once the decoder is initialized */
gst_element_post_message (element,
gst_message_new_tag (GST_OBJECT (element),
gst_tag_list_copy (decoder->tags)));
}
}
break;
default:
break;
}
}
#if ((KATE_VERSION_MAJOR<<16)|(KATE_VERSION_MINOR<<8)|KATE_VERSION_PATCH) >= 0x000400
else if (*ev && (*ev)->meta) {
int count = kate_meta_query_count ((*ev)->meta);
if (count > 0) {
GstTagList *evtags = gst_tag_list_new_empty ();
int idx;
GST_DEBUG_OBJECT (decoder, "Kate event has %d attached metadata", count);
for (idx = 0; idx < count; ++idx) {
const char *tag, *value;
size_t len;
if (kate_meta_query ((*ev)->meta, idx, &tag, &value, &len) < 0) {
GST_WARNING_OBJECT (decoder, "Failed to retrieve metadata %d", idx);
} else {
if (gst_kate_util_is_utf8_string (value, len)) {
gchar *compound = g_strdup_printf ("%s=%s", tag, value);
GST_DEBUG_OBJECT (decoder, "Metadata %d: %s=%s (%zu bytes)", idx,
tag, value, len);
gst_tag_list_add (evtags, GST_TAG_MERGE_APPEND,
GST_TAG_EXTENDED_COMMENT, compound, NULL);
g_free (compound);
} else {
GST_INFO_OBJECT (decoder,
"Metadata %d, (%s, %zu bytes) is binary, ignored", idx, tag,
len);
}
}
}
if (gst_tag_list_is_empty (evtags))
gst_tag_list_free (evtags);
else
gst_element_found_tags_for_pad (element, tagpad, evtags);
}
}
#endif
return rflow;
}
static GstFlowReturn
gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstDtsDec *dts = GST_DTSDEC (parent);
gint first_access;
if (dts->dvdmode) {
guint8 data[2];
gsize size;
gint offset, len;
GstBuffer *subbuf;
size = gst_buffer_get_size (buf);
if (size < 2)
goto not_enough_data;
gst_buffer_extract (buf, 0, data, 2);
first_access = (data[0] << 8) | data[1];
/* Skip the first_access header */
offset = 2;
if (first_access > 1) {
/* Length of data before first_access */
len = first_access - 1;
if (len <= 0 || offset + len > size)
goto bad_first_access_parameter;
subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
ret = dts->base_chain (pad, parent, subbuf);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buf);
goto done;
}
offset += len;
len = size - offset;
if (len > 0) {
subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
ret = dts->base_chain (pad, parent, subbuf);
}
gst_buffer_unref (buf);
} else {
/* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
subbuf =
gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset,
size - offset);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
ret = dts->base_chain (pad, parent, subbuf);
gst_buffer_unref (buf);
}
} else {
ret = dts->base_chain (pad, parent, buf);
}
done:
return ret;
/* ERRORS */
not_enough_data:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Insufficient data in buffer. Can't determine first_acess"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
bad_first_access_parameter:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Bad first_access parameter (%d) in buffer", first_access));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_flac_tag_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstFlacTag *tag;
GstFlowReturn ret;
GstMapInfo map;
gsize size;
ret = GST_FLOW_OK;
tag = GST_FLAC_TAG (parent);
gst_adapter_push (tag->adapter, buffer);
/* Initial state, we don't even know if we are dealing with a flac file */
if (tag->state == GST_FLAC_TAG_STATE_INIT) {
GstBuffer *id_buffer;
if (gst_adapter_available (tag->adapter) < sizeof (FLAC_MAGIC))
goto cleanup;
id_buffer = gst_adapter_take_buffer (tag->adapter, FLAC_MAGIC_SIZE);
GST_DEBUG_OBJECT (tag, "looking for " FLAC_MAGIC " identifier");
if (gst_buffer_memcmp (id_buffer, 0, FLAC_MAGIC, FLAC_MAGIC_SIZE) == 0) {
GST_DEBUG_OBJECT (tag, "pushing " FLAC_MAGIC " identifier buffer");
ret = gst_pad_push (tag->srcpad, id_buffer);
if (ret != GST_FLOW_OK)
goto cleanup;
tag->state = GST_FLAC_TAG_STATE_METADATA_BLOCKS;
} else {
/* FIXME: does that work well with FLAC files containing ID3v2 tags ? */
gst_buffer_unref (id_buffer);
GST_ELEMENT_ERROR (tag, STREAM, WRONG_TYPE, (NULL), (NULL));
ret = GST_FLOW_ERROR;
}
}
/* The fLaC magic string has been skipped, try to detect the beginning
* of a metadata block
*/
if (tag->state == GST_FLAC_TAG_STATE_METADATA_BLOCKS) {
guint type;
gboolean is_last;
const guint8 *block_header;
g_assert (tag->metadata_block_size == 0);
g_assert (tag->metadata_last_block == FALSE);
/* The header of a flac metadata block is 4 bytes long:
* 1st bit: indicates whether this is the last metadata info block
* 7 next bits: 4 if vorbis comment block
* 24 next bits: size of the metadata to follow (big endian)
*/
if (gst_adapter_available (tag->adapter) < 4)
goto cleanup;
block_header = gst_adapter_map (tag->adapter, 4);
is_last = ((block_header[0] & 0x80) == 0x80);
type = block_header[0] & 0x7F;
size = (block_header[1] << 16)
| (block_header[2] << 8)
| block_header[3];
gst_adapter_unmap (tag->adapter);
/* The 4 bytes long header isn't included in the metadata size */
tag->metadata_block_size = size + 4;
tag->metadata_last_block = is_last;
GST_DEBUG_OBJECT (tag,
"got metadata block: %" G_GSIZE_FORMAT " bytes, type %d, "
"is vorbiscomment: %d, is last: %d",
size, type, (type == 0x04), is_last);
/* Metadata blocks of type 4 are vorbis comment blocks */
if (type == 0x04) {
tag->state = GST_FLAC_TAG_STATE_VC_METADATA_BLOCK;
} else {
tag->state = GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK;
}
}
/* Reads a metadata block */
if ((tag->state == GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK) ||
(tag->state == GST_FLAC_TAG_STATE_VC_METADATA_BLOCK)) {
GstBuffer *metadata_buffer;
if (gst_adapter_available (tag->adapter) < tag->metadata_block_size)
goto cleanup;
metadata_buffer = gst_adapter_take_buffer (tag->adapter,
tag->metadata_block_size);
/* clear the is-last flag, as the last metadata block will
* be the vorbis comment block which we will build ourselves.
*/
gst_buffer_map (metadata_buffer, &map, GST_MAP_READWRITE);
map.data[0] &= (~0x80);
gst_buffer_unmap (metadata_buffer, &map);
if (tag->state == GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK) {
GST_DEBUG_OBJECT (tag, "pushing metadata block buffer");
ret = gst_pad_push (tag->srcpad, metadata_buffer);
if (ret != GST_FLOW_OK)
goto cleanup;
} else {
tag->vorbiscomment = metadata_buffer;
}
tag->metadata_block_size = 0;
tag->state = GST_FLAC_TAG_STATE_METADATA_NEXT_BLOCK;
}
/* This state is mainly used to be able to stop as soon as we read
* a vorbiscomment block from the flac file if we are in an only output
* tags mode
*/
if (tag->state == GST_FLAC_TAG_STATE_METADATA_NEXT_BLOCK) {
/* Check if in the previous iteration we read a vorbis comment metadata
* block, and stop now if the user only wants to read tags
*/
if (tag->vorbiscomment != NULL) {
guint8 id_data[4];
/* We found some tags, try to parse them and notify the other elements
* that we encountered some tags
*/
GST_DEBUG_OBJECT (tag, "emitting vorbiscomment tags");
gst_buffer_extract (tag->vorbiscomment, 0, id_data, 4);
tag->tags = gst_tag_list_from_vorbiscomment_buffer (tag->vorbiscomment,
id_data, 4, NULL);
if (tag->tags != NULL) {
gst_pad_push_event (tag->srcpad,
gst_event_new_tag (gst_tag_list_copy (tag->tags)));
}
gst_buffer_unref (tag->vorbiscomment);
tag->vorbiscomment = NULL;
}
/* Skip to next state */
if (tag->metadata_last_block == FALSE) {
tag->state = GST_FLAC_TAG_STATE_METADATA_BLOCKS;
} else {
tag->state = GST_FLAC_TAG_STATE_ADD_VORBIS_COMMENT;
}
}
/* Creates a vorbis comment block from the metadata which was set
* on the gstreamer element, and add it to the flac stream
*/
if (tag->state == GST_FLAC_TAG_STATE_ADD_VORBIS_COMMENT) {
GstBuffer *buffer;
const GstTagList *user_tags;
GstTagList *merged_tags;
/* merge the tag lists */
user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (tag));
if (user_tags != NULL) {
merged_tags = gst_tag_list_merge (user_tags, tag->tags,
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (tag)));
} else {
merged_tags = gst_tag_list_copy (tag->tags);
}
if (merged_tags == NULL) {
/* If we get a NULL list of tags, we must generate a padding block
* which is marked as the last metadata block, otherwise we'll
* end up with a corrupted flac file.
*/
GST_WARNING_OBJECT (tag, "No tags found");
buffer = gst_buffer_new_and_alloc (12);
if (buffer == NULL)
goto no_buffer;
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 0, map.size);
map.data[0] = 0x81; /* 0x80 = Last metadata block,
* 0x01 = padding block */
gst_buffer_unmap (buffer, &map);
} else {
guchar header[4];
guint8 fbit[1];
memset (header, 0, sizeof (header));
header[0] = 0x84; /* 0x80 = Last metadata block,
* 0x04 = vorbiscomment block */
buffer = gst_tag_list_to_vorbiscomment_buffer (merged_tags, header,
sizeof (header), NULL);
GST_DEBUG_OBJECT (tag, "Writing tags %" GST_PTR_FORMAT, merged_tags);
gst_tag_list_free (merged_tags);
if (buffer == NULL)
goto no_comment;
size = gst_buffer_get_size (buffer);
if ((size < 4) || ((size - 4) > 0xFFFFFF))
goto comment_too_long;
fbit[0] = 1;
/* Get rid of the framing bit at the end of the vorbiscomment buffer
* if it exists since libFLAC seems to lose sync because of this
* bit in gstflacdec
*/
if (gst_buffer_memcmp (buffer, size - 1, fbit, 1) == 0) {
buffer = gst_buffer_make_writable (buffer);
gst_buffer_resize (buffer, 0, size - 1);
}
}
/* The 4 byte metadata block header isn't accounted for in the total
* size of the metadata block
*/
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
map.data[1] = (((map.size - 4) & 0xFF0000) >> 16);
map.data[2] = (((map.size - 4) & 0x00FF00) >> 8);
map.data[3] = ((map.size - 4) & 0x0000FF);
gst_buffer_unmap (buffer, &map);
GST_DEBUG_OBJECT (tag, "pushing %" G_GSIZE_FORMAT " byte vorbiscomment "
"buffer", map.size);
ret = gst_pad_push (tag->srcpad, buffer);
if (ret != GST_FLOW_OK) {
goto cleanup;
}
tag->state = GST_FLAC_TAG_STATE_AUDIO_DATA;
}
static GstFlowReturn
gst_shm_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstShmSink *self = GST_SHM_SINK (bsink);
int rv = 0;
GstMapInfo map;
gboolean need_new_memory = FALSE;
GstFlowReturn ret = GST_FLOW_OK;
GstMemory *memory = NULL;
GstBuffer *sendbuf = NULL;
GST_OBJECT_LOCK (self);
while (self->wait_for_connection && !self->clients) {
g_cond_wait (&self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock)
goto flushing;
}
while (!gst_shm_sink_can_render (self, GST_BUFFER_TIMESTAMP (buf))) {
g_cond_wait (&self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock)
goto flushing;
}
if (gst_buffer_n_memory (buf) > 1) {
GST_LOG_OBJECT (self, "Buffer %p has %d GstMemory, we only support a single"
" one, need to do a memcpy", buf, gst_buffer_n_memory (buf));
need_new_memory = TRUE;
} else {
memory = gst_buffer_peek_memory (buf, 0);
if (memory->allocator != GST_ALLOCATOR (self->allocator)) {
need_new_memory = TRUE;
GST_LOG_OBJECT (self, "Memory in buffer %p was not allocated by "
"%" GST_PTR_FORMAT ", will memcpy", buf, memory->allocator);
}
}
if (need_new_memory) {
if (gst_buffer_get_size (buf) > sp_writer_get_max_buf_size (self->pipe)) {
gsize area_size = sp_writer_get_max_buf_size (self->pipe);
GST_OBJECT_UNLOCK (self);
GST_ELEMENT_ERROR (self, RESOURCE, NO_SPACE_LEFT,
("Shared memory area is too small"),
("Shared memory area of size %" G_GSIZE_FORMAT " is smaller than"
"buffer of size %" G_GSIZE_FORMAT, area_size,
gst_buffer_get_size (buf)));
return GST_FLOW_ERROR;
}
while ((memory =
gst_shm_sink_allocator_alloc_locked (self->allocator,
gst_buffer_get_size (buf), &self->params)) == NULL) {
g_cond_wait (&self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock)
goto flushing;
}
while (self->wait_for_connection && !self->clients) {
g_cond_wait (&self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
gst_memory_unref (memory);
GST_OBJECT_UNLOCK (self);
return GST_FLOW_FLUSHING;
}
}
gst_memory_map (memory, &map, GST_MAP_WRITE);
gst_buffer_extract (buf, 0, map.data, map.size);
gst_memory_unmap (memory, &map);
sendbuf = gst_buffer_new ();
gst_buffer_copy_into (sendbuf, buf, GST_BUFFER_COPY_METADATA, 0, -1);
gst_buffer_append_memory (sendbuf, memory);
} else {
sendbuf = gst_buffer_ref (buf);
}
gst_buffer_map (sendbuf, &map, GST_MAP_READ);
/* Make the memory readonly as of now as we've sent it to the other side
* We know it's not mapped for writing anywhere as we just mapped it for
* reading
*/
rv = sp_writer_send_buf (self->pipe, (char *) map.data, map.size, sendbuf);
gst_buffer_unmap (sendbuf, &map);
GST_OBJECT_UNLOCK (self);
if (rv == 0) {
GST_DEBUG_OBJECT (self, "No clients connected, unreffing buffer");
gst_buffer_unref (sendbuf);
} else if (rv == -1) {
GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Invalid allocated buffer"),
("The shmpipe library rejects our buffer, this is a bug"));
ret = GST_FLOW_ERROR;
}
/* If we allocated our own memory, then unmap it */
return ret;
flushing:
GST_OBJECT_UNLOCK (self);
return GST_FLOW_FLUSHING;
}
static GstFlowReturn
gst_shm_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstShmSink *self = GST_SHM_SINK (bsink);
int rv = 0;
GstMapInfo map;
gboolean need_new_memory = FALSE;
GstFlowReturn ret = GST_FLOW_OK;
GstMemory *memory = NULL;
GstBuffer *sendbuf = NULL;
gsize written_bytes;
GST_OBJECT_LOCK (self);
while (self->wait_for_connection && !self->clients) {
g_cond_wait (&self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
GST_OBJECT_UNLOCK (self);
ret = gst_base_sink_wait_preroll (bsink);
if (ret == GST_FLOW_OK)
GST_OBJECT_LOCK (self);
else
return ret;
}
}
while (!gst_shm_sink_can_render (self, GST_BUFFER_TIMESTAMP (buf))) {
g_cond_wait (&self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
GST_OBJECT_UNLOCK (self);
ret = gst_base_sink_wait_preroll (bsink);
if (ret == GST_FLOW_OK)
GST_OBJECT_LOCK (self);
else
return ret;
}
}
if (gst_buffer_n_memory (buf) > 1) {
GST_LOG_OBJECT (self, "Buffer %p has %d GstMemory, we only support a single"
" one, need to do a memcpy", buf, gst_buffer_n_memory (buf));
need_new_memory = TRUE;
} else {
memory = gst_buffer_peek_memory (buf, 0);
if (memory->allocator != GST_ALLOCATOR (self->allocator)) {
need_new_memory = TRUE;
GST_LOG_OBJECT (self, "Memory in buffer %p was not allocated by "
"%" GST_PTR_FORMAT ", will memcpy", buf, memory->allocator);
}
}
if (need_new_memory) {
if (gst_buffer_get_size (buf) > sp_writer_get_max_buf_size (self->pipe)) {
gsize area_size = sp_writer_get_max_buf_size (self->pipe);
GST_ELEMENT_ERROR (self, RESOURCE, NO_SPACE_LEFT, (NULL),
("Shared memory area of size %" G_GSIZE_FORMAT " is smaller than"
"buffer of size %" G_GSIZE_FORMAT, area_size,
gst_buffer_get_size (buf)));
goto error;
}
while ((memory =
gst_shm_sink_allocator_alloc_locked (self->allocator,
gst_buffer_get_size (buf), &self->params)) == NULL) {
g_cond_wait (&self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
GST_OBJECT_UNLOCK (self);
ret = gst_base_sink_wait_preroll (bsink);
if (ret == GST_FLOW_OK)
GST_OBJECT_LOCK (self);
else
return ret;
}
}
while (self->wait_for_connection && !self->clients) {
g_cond_wait (&self->cond, GST_OBJECT_GET_LOCK (self));
if (self->unlock) {
GST_OBJECT_UNLOCK (self);
ret = gst_base_sink_wait_preroll (bsink);
if (ret == GST_FLOW_OK) {
GST_OBJECT_LOCK (self);
} else {
gst_memory_unref (memory);
return ret;
}
}
}
if (!gst_memory_map (memory, &map, GST_MAP_WRITE)) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to map memory"));
goto error;
}
GST_DEBUG_OBJECT (self,
"Copying %" G_GSIZE_FORMAT " bytes into map of size %" G_GSIZE_FORMAT
" bytes.", gst_buffer_get_size (buf), map.size);
written_bytes = gst_buffer_extract (buf, 0, map.data, map.size);
GST_DEBUG_OBJECT (self, "Copied %" G_GSIZE_FORMAT " bytes.", written_bytes);
gst_memory_unmap (memory, &map);
sendbuf = gst_buffer_new ();
if (!gst_buffer_copy_into (sendbuf, buf, GST_BUFFER_COPY_METADATA, 0, -1)) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to copy data into send buffer"));
gst_buffer_unref (sendbuf);
goto error;
}
gst_buffer_append_memory (sendbuf, memory);
} else {
sendbuf = gst_buffer_ref (buf);
}
if (!gst_buffer_map (sendbuf, &map, GST_MAP_READ)) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to map data into send buffer"));
goto error;
}
/* Make the memory readonly as of now as we've sent it to the other side
* We know it's not mapped for writing anywhere as we just mapped it for
* reading
*/
rv = sp_writer_send_buf (self->pipe, (char *) map.data, map.size, sendbuf);
if (rv == -1) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to send data over SHM"));
gst_buffer_unmap (sendbuf, &map);
goto error;
}
gst_buffer_unmap (sendbuf, &map);
GST_OBJECT_UNLOCK (self);
if (rv == 0) {
GST_DEBUG_OBJECT (self, "No clients connected, unreffing buffer");
gst_buffer_unref (sendbuf);
}
return ret;
error:
GST_OBJECT_UNLOCK (self);
return GST_FLOW_ERROR;
}
static GstFlowReturn
gst_rtp_asf_pay_handle_packet (GstRtpAsfPay * rtpasfpay, GstBuffer * buffer)
{
GstRTPBasePayload *rtppay;
GstAsfPacketInfo *packetinfo;
guint8 flags;
guint8 *data;
guint32 packet_util_size;
guint32 packet_offset;
guint32 size_left;
GstFlowReturn ret = GST_FLOW_OK;
rtppay = GST_RTP_BASE_PAYLOAD (rtpasfpay);
packetinfo = &rtpasfpay->packetinfo;
if (!gst_asf_parse_packet (buffer, packetinfo, TRUE,
rtpasfpay->asfinfo.packet_size)) {
GST_ERROR_OBJECT (rtpasfpay, "Error while parsing asf packet");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
if (packetinfo->packet_size == 0)
packetinfo->packet_size = rtpasfpay->asfinfo.packet_size;
GST_LOG_OBJECT (rtpasfpay, "Packet size: %" G_GUINT32_FORMAT
", padding: %" G_GUINT32_FORMAT, packetinfo->packet_size,
packetinfo->padding);
/* update padding field to 0 */
if (packetinfo->padding > 0) {
GstAsfPacketInfo info;
/* find padding field offset */
guint offset = packetinfo->err_cor_len + 2 +
gst_asf_get_var_size_field_len (packetinfo->packet_field_type) +
gst_asf_get_var_size_field_len (packetinfo->seq_field_type);
buffer = gst_buffer_make_writable (buffer);
switch (packetinfo->padd_field_type) {
case ASF_FIELD_TYPE_DWORD:
gst_buffer_memset (buffer, offset, 0, 4);
break;
case ASF_FIELD_TYPE_WORD:
gst_buffer_memset (buffer, offset, 0, 2);
break;
case ASF_FIELD_TYPE_BYTE:
gst_buffer_memset (buffer, offset, 0, 1);
break;
case ASF_FIELD_TYPE_NONE:
default:
break;
}
gst_asf_parse_packet (buffer, &info, FALSE, 0);
}
if (packetinfo->padding != 0)
packet_util_size = rtpasfpay->asfinfo.packet_size - packetinfo->padding;
else
packet_util_size = packetinfo->packet_size;
packet_offset = 0;
while (packet_util_size > 0) {
/* Even if we don't fill completely an output buffer we
* push it when we add an fragment. Because it seems that
* it is not possible to determine where a asf packet
* fragment ends inside a rtp packet payload.
* This flag tells us to push the packet.
*/
gboolean force_push = FALSE;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
/* we have no output buffer pending, create one */
if (rtpasfpay->current == NULL) {
GST_LOG_OBJECT (rtpasfpay, "Creating new output buffer");
rtpasfpay->current =
gst_rtp_buffer_new_allocate_len (GST_RTP_BASE_PAYLOAD_MTU (rtpasfpay),
0, 0);
rtpasfpay->cur_off = 0;
rtpasfpay->has_ts = FALSE;
rtpasfpay->marker = FALSE;
}
gst_rtp_buffer_map (rtpasfpay->current, GST_MAP_READWRITE, &rtp);
data = gst_rtp_buffer_get_payload (&rtp);
data += rtpasfpay->cur_off;
size_left = gst_rtp_buffer_get_payload_len (&rtp) - rtpasfpay->cur_off;
GST_DEBUG_OBJECT (rtpasfpay, "Input buffer bytes consumed: %"
G_GUINT32_FORMAT "/%" G_GSIZE_FORMAT, packet_offset,
gst_buffer_get_size (buffer));
GST_DEBUG_OBJECT (rtpasfpay, "Output rtpbuffer status");
GST_DEBUG_OBJECT (rtpasfpay, "Current offset: %" G_GUINT32_FORMAT,
rtpasfpay->cur_off);
GST_DEBUG_OBJECT (rtpasfpay, "Size left: %" G_GUINT32_FORMAT, size_left);
GST_DEBUG_OBJECT (rtpasfpay, "Has ts: %s",
rtpasfpay->has_ts ? "yes" : "no");
if (rtpasfpay->has_ts) {
GST_DEBUG_OBJECT (rtpasfpay, "Ts: %" G_GUINT32_FORMAT, rtpasfpay->ts);
}
flags = 0;
if (packetinfo->has_keyframe) {
flags = flags | 0x80;
}
flags = flags | 0x20; /* Relative timestamp is present */
if (!rtpasfpay->has_ts) {
/* this is the first asf packet, its send time is the
* rtp packet timestamp */
rtpasfpay->has_ts = TRUE;
rtpasfpay->ts = packetinfo->send_time;
}
if (size_left >= packet_util_size + 8) {
/* enough space for the rest of the packet */
if (packet_offset == 0) {
flags = flags | 0x40;
GST_WRITE_UINT24_BE (data + 1, packet_util_size);
} else {
GST_WRITE_UINT24_BE (data + 1, packet_offset);
force_push = TRUE;
}
data[0] = flags;
GST_WRITE_UINT32_BE (data + 4,
(gint32) (packetinfo->send_time) - (gint32) rtpasfpay->ts);
gst_buffer_extract (buffer, packet_offset, data + 8, packet_util_size);
/* updating status variables */
rtpasfpay->cur_off += 8 + packet_util_size;
size_left -= packet_util_size + 8;
packet_offset += packet_util_size;
packet_util_size = 0;
rtpasfpay->marker = TRUE;
} else {
/* fragment packet */
data[0] = flags;
GST_WRITE_UINT24_BE (data + 1, packet_offset);
GST_WRITE_UINT32_BE (data + 4,
(gint32) (packetinfo->send_time) - (gint32) rtpasfpay->ts);
gst_buffer_extract (buffer, packet_offset, data + 8, size_left - 8);
/* updating status variables */
rtpasfpay->cur_off += size_left;
packet_offset += size_left - 8;
packet_util_size -= size_left - 8;
size_left = 0;
force_push = TRUE;
}
/* there is not enough room for any more buffers */
if (force_push || size_left <= 8) {
gst_rtp_buffer_set_ssrc (&rtp, rtppay->current_ssrc);
gst_rtp_buffer_set_marker (&rtp, rtpasfpay->marker);
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (rtppay));
gst_rtp_buffer_set_seq (&rtp, rtppay->seqnum + 1);
gst_rtp_buffer_set_timestamp (&rtp, packetinfo->send_time);
gst_rtp_buffer_unmap (&rtp);
/* trim remaining bytes not used */
if (size_left != 0) {
gst_buffer_set_size (rtpasfpay->current,
gst_buffer_get_size (rtpasfpay->current) - size_left);
}
GST_BUFFER_TIMESTAMP (rtpasfpay->current) = GST_BUFFER_TIMESTAMP (buffer);
rtppay->seqnum++;
rtppay->timestamp = packetinfo->send_time;
GST_DEBUG_OBJECT (rtpasfpay, "Pushing rtp buffer");
ret = gst_rtp_base_payload_push (rtppay, rtpasfpay->current);
rtpasfpay->current = NULL;
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buffer);
return ret;
}
}
}
gst_buffer_unref (buffer);
return ret;
}
static GstFlowReturn
gst_decklink_sink_videosink_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstDecklinkSink *decklinksink;
IDeckLinkMutableVideoFrame *frame;
void *data;
GstFlowReturn ret;
const GstDecklinkMode *mode;
decklinksink = GST_DECKLINK_SINK (parent);
#if 0
if (!decklinksink->video_enabled) {
HRESULT ret;
ret = decklinksink->output->EnableVideoOutput (decklinksink->display_mode,
bmdVideoOutputFlagDefault);
if (ret != S_OK) {
GST_WARNING ("failed to enable video output");
//return FALSE;
}
decklinksink->video_enabled = TRUE;
}
#endif
mode = gst_decklink_get_mode (decklinksink->mode);
decklinksink->output->CreateVideoFrame (mode->width,
mode->height, mode->width * 2, decklinksink->pixel_format,
bmdFrameFlagDefault, &frame);
frame->GetBytes (&data);
gst_buffer_extract (buffer, 0, data, gst_buffer_get_size (buffer));
gst_buffer_unref (buffer);
g_mutex_lock (&decklinksink->mutex);
while (decklinksink->queued_frames > 2 && !decklinksink->stop) {
g_cond_wait (&decklinksink->cond, &decklinksink->mutex);
}
if (!decklinksink->stop) {
decklinksink->queued_frames++;
}
g_mutex_unlock (&decklinksink->mutex);
if (!decklinksink->stop) {
decklinksink->output->ScheduleVideoFrame (frame,
decklinksink->num_frames * mode->fps_d, mode->fps_d, mode->fps_n);
decklinksink->num_frames++;
if (!decklinksink->sched_started) {
decklinksink->output->StartScheduledPlayback (0, mode->fps_d, 1.0);
decklinksink->sched_started = TRUE;
}
ret = GST_FLOW_OK;
} else {
ret = GST_FLOW_FLUSHING;
}
frame->Release ();
return ret;
}
