static GstFlowReturn
gst_siren_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstSirenDec *dec;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint i, size, num_frames;
gint out_size, in_size;
gint decode_ret;
GstMapInfo inmap, outmap;
dec = GST_SIREN_DEC (bdec);
size = gst_buffer_get_size (buf);
GST_LOG_OBJECT (dec, "Received buffer of size %u", size);
g_return_val_if_fail (size % 40 == 0, GST_FLOW_ERROR);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
/* process 40 input bytes into 640 output bytes */
num_frames = size / 40;
/* this is the input/output size */
in_size = num_frames * 40;
out_size = num_frames * 640;
GST_LOG_OBJECT (dec, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
out_buf = gst_audio_decoder_allocate_output_buffer (bdec, out_size);
if (out_buf == NULL)
goto alloc_failed;
/* get the input data for all the frames */
gst_buffer_map (buf, &inmap, GST_MAP_READ);
gst_buffer_map (out_buf, &outmap, GST_MAP_WRITE);
in_data = inmap.data;
out_data = outmap.data;
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (dec, "Decoding frame %u/%u", i, num_frames);
/* decode 40 input bytes to 640 output bytes */
decode_ret = Siren7_DecodeFrame (dec->decoder, in_data, out_data);
if (decode_ret != 0)
goto decode_error;
/* move to next frame */
out_data += 640;
in_data += 40;
}
gst_buffer_unmap (buf, &inmap);
gst_buffer_unmap (out_buf, &outmap);
GST_LOG_OBJECT (dec, "Finished decoding");
/* might really be multiple frames,
* but was treated as one for all purposes here */
ret = gst_audio_decoder_finish_frame (bdec, out_buf, 1);
done:
return ret;
/* ERRORS */
alloc_failed:
{
GST_DEBUG_OBJECT (dec, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
decode_error:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("Error decoding frame: %d", decode_ret), ret);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
gst_buffer_unref (out_buf);
goto done;
}
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gsize size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
GstMapInfo map, omap;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
gst_opus_dec_negotiate (dec, NULL);
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_DEBUG
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
"Mapping table", dec->n_channels, dec->channel_mapping);
#endif
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
dec->n_stereo_streams);
dec->state =
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %u",
gst_buffer_get_size (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = (dec->use_inband_fec
&& gst_buffer_get_size (dec->last_buffer) >
0) ? dec->last_buffer : buffer;
if (buf && gst_buffer_get_size (buf) > 0) {
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (dec, "Using buffer of size %u", size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
packet_size = samples * dec->n_channels * 2;
outbuf = gst_buffer_new_and_alloc (packet_size);
if (!outbuf) {
goto buffer_failed;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
if (dec->use_inband_fec) {
if (dec->last_buffer) {
/* normal delayed decode */
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
}
gst_buffer_unmap (outbuf, &omap);
if (buf)
gst_buffer_unmap (buf, &map);
if (n < 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
gst_buffer_unref (outbuf);
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
/* Skip any samples that need skipping */
if (dec->pre_skip > 0) {
guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
dec->pre_skip -= scaled_skip;
GST_INFO_OBJECT (dec,
"Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip,
scaled_skip, dec->pre_skip);
}
if (gst_buffer_get_size (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
dec->n_channels, dec->opus_pos, dec->info.position);
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
gsize rsize;
unsigned int i, nsamples;
double volume = dec->r128_gain_volume;
gint16 *samples;
gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
samples = (gint16 *) omap.data;
rsize = omap.size;
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
nsamples = rsize / 2;
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
gst_buffer_unmap (outbuf, &omap);
}
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
}
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
done:
return res;
creation_failed:
GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
return GST_FLOW_ERROR;
buffer_failed:
GST_ERROR_OBJECT (dec, "Failed to create %u byte buffer", packet_size);
return GST_FLOW_ERROR;
}
static gboolean
gst_bz2enc_event (GstPad * pad, GstObject * parent, GstEvent * e)
{
GstBz2enc *b;
gboolean ret;
b = GST_BZ2ENC (parent);
switch (GST_EVENT_TYPE (e)) {
case GST_EVENT_EOS:{
GstFlowReturn flow;
int r = BZ_FINISH_OK;
do {
GstBuffer *out;
GstMapInfo omap;
guint n;
out = gst_buffer_new_and_alloc (b->buffer_size);
gst_buffer_map (out, &omap, GST_MAP_WRITE);
b->stream.next_out = (char *) omap.data;
b->stream.avail_out = omap.size;
r = BZ2_bzCompress (&b->stream, BZ_FINISH);
gst_buffer_unmap (out, &omap);
if ((r != BZ_FINISH_OK) && (r != BZ_STREAM_END)) {
GST_ELEMENT_ERROR (b, STREAM, ENCODE, (NULL),
("Failed to finish to compress (error code %i).", r));
gst_buffer_unref (out);
break;
}
n = gst_buffer_get_size (out);
if (b->stream.avail_out >= n) {
gst_buffer_unref (out);
break;
}
gst_buffer_resize (out, 0, n - b->stream.avail_out);
n = gst_buffer_get_size (out);
GST_BUFFER_OFFSET (out) = b->stream.total_out_lo32 - n;
flow = gst_pad_push (b->src, out);
if (flow != GST_FLOW_OK) {
GST_DEBUG_OBJECT (b, "push on EOS failed: %s",
gst_flow_get_name (flow));
break;
}
} while (r != BZ_STREAM_END);
ret = gst_pad_event_default (pad, parent, e);
if (r != BZ_STREAM_END || flow != GST_FLOW_OK)
ret = FALSE;
gst_bz2enc_compress_init (b);
break;
}
default:
ret = gst_pad_event_default (pad, parent, e);
break;
}
return ret;
}
/* Reads in buffers, parses them, reframes into one-buffer-per-ogg-page, submits
* pages to output pad.
*/
static GstFlowReturn
gst_ogg_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstOggParse *ogg;
GstFlowReturn result = GST_FLOW_OK;
gint ret = -1;
guint32 serialno;
GstBuffer *pagebuffer;
GstClockTime buffertimestamp = GST_BUFFER_TIMESTAMP (buffer);
ogg = GST_OGG_PARSE (parent);
GST_LOG_OBJECT (ogg,
"Chain function received buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
gst_ogg_parse_submit_buffer (ogg, buffer);
while (ret != 0 && result == GST_FLOW_OK) {
ogg_page page;
/* We use ogg_sync_pageseek() rather than ogg_sync_pageout() so that we can
* track how many bytes the ogg layer discarded (in the case of sync errors,
* etc.); this allows us to accurately track the current stream offset
*/
ret = ogg_sync_pageseek (&ogg->sync, &page);
if (ret == 0) {
/* need more data, that's fine... */
break;
} else if (ret < 0) {
/* discontinuity; track how many bytes we skipped (-ret) */
ogg->offset -= ret;
} else {
gint64 granule = ogg_page_granulepos (&page);
#ifndef GST_DISABLE_GST_DEBUG
int bos = ogg_page_bos (&page);
#endif
guint64 startoffset = ogg->offset;
GstOggStream *stream;
gboolean keyframe;
serialno = ogg_page_serialno (&page);
stream = gst_ogg_parse_find_stream (ogg, serialno);
GST_LOG_OBJECT (ogg, "Timestamping outgoing buffer as %" GST_TIME_FORMAT,
GST_TIME_ARGS (buffertimestamp));
if (stream) {
buffertimestamp = gst_ogg_stream_get_end_time_for_granulepos (stream,
granule);
if (ogg->video_stream) {
if (stream == ogg->video_stream) {
keyframe = gst_ogg_stream_granulepos_is_key_frame (stream, granule);
} else {
keyframe = FALSE;
}
} else {
keyframe = TRUE;
}
} else {
buffertimestamp = GST_CLOCK_TIME_NONE;
keyframe = TRUE;
}
pagebuffer = gst_ogg_parse_buffer_from_page (&page, startoffset,
buffertimestamp);
/* We read out 'ret' bytes, so we set the next offset appropriately */
ogg->offset += ret;
GST_LOG_OBJECT (ogg,
"processing ogg page (serial %08x, pageno %ld, "
"granule pos %" G_GUINT64_FORMAT ", bos %d, offset %"
G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT ") keyframe=%d",
serialno, ogg_page_pageno (&page),
granule, bos, startoffset, ogg->offset, keyframe);
if (ogg_page_bos (&page)) {
/* If we've seen this serialno before, this is technically an error,
* we log this case but accept it - this one replaces the previous
* stream with this serialno. We can do this since we're streaming, and
* not supporting seeking...
*/
GstOggStream *stream = gst_ogg_parse_find_stream (ogg, serialno);
if (stream != NULL) {
GST_LOG_OBJECT (ogg, "Incorrect stream; repeats serial number %08x "
"at offset %" G_GINT64_FORMAT, serialno, ogg->offset);
}
if (ogg->last_page_not_bos) {
GST_LOG_OBJECT (ogg, "Deleting all referenced streams, found a new "
"chain starting with serial %u", serialno);
gst_ogg_parse_delete_all_streams (ogg);
}
stream = gst_ogg_parse_new_stream (ogg, &page);
ogg->last_page_not_bos = FALSE;
gst_buffer_ref (pagebuffer);
stream->headers = g_list_append (stream->headers, pagebuffer);
if (!ogg->in_headers) {
GST_LOG_OBJECT (ogg,
"Found start of new chain at offset %" G_GUINT64_FORMAT,
startoffset);
ogg->in_headers = 1;
}
/* For now, we just keep the header buffer in the stream->headers list;
* it actually gets output once we've collected the entire set
*/
} else {
/* Non-BOS page. Either: we're outside headers, and this isn't a
* header (normal data), outside headers and this is (error!), inside
* headers, this is (append header), or inside headers and this isn't
* (we've found the end of headers; flush the lot!)
*
* Before that, we flag that the last page seen (this one) was not a
* BOS page; that way we know that when we next see a BOS page it's a
* new chain, and we can flush all existing streams.
*/
page_type type;
GstOggStream *stream = gst_ogg_parse_find_stream (ogg, serialno);
if (!stream) {
GST_LOG_OBJECT (ogg,
"Non-BOS page unexpectedly found at %" G_GINT64_FORMAT,
ogg->offset);
goto failure;
}
ogg->last_page_not_bos = TRUE;
type = gst_ogg_parse_is_header (ogg, stream, &page);
if (type == PAGE_PENDING && ogg->in_headers) {
gst_buffer_ref (pagebuffer);
stream->unknown_pages = g_list_append (stream->unknown_pages,
pagebuffer);
} else if (type == PAGE_HEADER) {
if (!ogg->in_headers) {
GST_LOG_OBJECT (ogg, "Header page unexpectedly found outside "
"headers at offset %" G_GINT64_FORMAT, ogg->offset);
goto failure;
} else {
/* Append the header to the buffer list, after any unknown previous
* pages
*/
stream->headers = g_list_concat (stream->headers,
stream->unknown_pages);
g_list_free (stream->unknown_pages);
gst_buffer_ref (pagebuffer);
stream->headers = g_list_append (stream->headers, pagebuffer);
}
} else { /* PAGE_DATA, or PAGE_PENDING but outside headers */
if (ogg->in_headers) {
/* First non-header page... set caps, flush headers.
*
* First up, we build a single GValue list of all the pagebuffers
* we're using for the headers, in order.
* Then we set this on the caps structure. Then we can start pushing
* buffers for the headers, and finally we send this non-header
* page.
*/
GstCaps *caps;
GstStructure *structure;
GValue array = { 0 };
gint count = 0;
gboolean found_pending_headers = FALSE;
GSList *l;
g_value_init (&array, GST_TYPE_ARRAY);
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
if (g_list_length (stream->headers) == 0) {
GST_LOG_OBJECT (ogg, "No primary header found for stream %08x",
stream->serialno);
goto failure;
}
gst_ogg_parse_append_header (&array,
GST_BUFFER (stream->headers->data));
count++;
}
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
GList *j;
/* already appended the first header, now do headers 2-N */
for (j = stream->headers->next; j != NULL; j = j->next) {
gst_ogg_parse_append_header (&array, GST_BUFFER (j->data));
count++;
}
}
caps = gst_pad_query_caps (ogg->srcpad, NULL);
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
gst_structure_take_value (structure, "streamheader", &array);
gst_pad_set_caps (ogg->srcpad, caps);
if (ogg->caps)
gst_caps_unref (ogg->caps);
ogg->caps = caps;
GST_LOG_OBJECT (ogg, "Set \"streamheader\" caps with %d buffers "
"(one per page)", count);
/* Now, we do the same thing, but push buffers... */
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
GstBuffer *buf = GST_BUFFER (stream->headers->data);
result = gst_pad_push (ogg->srcpad, buf);
if (result != GST_FLOW_OK)
return result;
}
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
GList *j;
/* pushed the first one for each stream already, now do 2-N */
for (j = stream->headers->next; j != NULL; j = j->next) {
GstBuffer *buf = GST_BUFFER (j->data);
result = gst_pad_push (ogg->srcpad, buf);
if (result != GST_FLOW_OK)
return result;
}
}
ogg->in_headers = 0;
/* And finally the pending data pages */
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
GList *k;
if (stream->unknown_pages == NULL)
continue;
if (found_pending_headers) {
GST_WARNING_OBJECT (ogg, "Incorrectly muxed headers found at "
"approximate offset %" G_GINT64_FORMAT, ogg->offset);
}
found_pending_headers = TRUE;
GST_LOG_OBJECT (ogg, "Pushing %d pending pages after headers",
g_list_length (stream->unknown_pages) + 1);
for (k = stream->unknown_pages; k != NULL; k = k->next) {
GstBuffer *buf = GST_BUFFER (k->data);
result = gst_pad_push (ogg->srcpad, buf);
if (result != GST_FLOW_OK)
return result;
}
g_list_foreach (stream->unknown_pages,
(GFunc) gst_mini_object_unref, NULL);
g_list_free (stream->unknown_pages);
stream->unknown_pages = NULL;
}
}
if (granule == -1) {
stream->stored_buffers = g_list_append (stream->stored_buffers,
pagebuffer);
} else {
while (stream->stored_buffers) {
GstBuffer *buf = stream->stored_buffers->data;
buf = gst_buffer_make_writable (buf);
GST_BUFFER_TIMESTAMP (buf) = buffertimestamp;
if (!keyframe) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DELTA_UNIT);
} else {
keyframe = FALSE;
}
result = gst_pad_push (ogg->srcpad, buf);
if (result != GST_FLOW_OK)
return result;
stream->stored_buffers =
g_list_delete_link (stream->stored_buffers,
stream->stored_buffers);
}
pagebuffer = gst_buffer_make_writable (pagebuffer);
if (!keyframe) {
GST_BUFFER_FLAG_SET (pagebuffer, GST_BUFFER_FLAG_DELTA_UNIT);
} else {
keyframe = FALSE;
}
result = gst_pad_push (ogg->srcpad, pagebuffer);
if (result != GST_FLOW_OK)
return result;
}
}
}
}
}
return result;
failure:
gst_pad_push_event (GST_PAD (ogg->srcpad), gst_event_new_eos ());
return GST_FLOW_ERROR;
}
/* for off == 4 initial code; returns TRUE if code starts a frame
* otherwise returns TRUE if code terminates preceding frame */
static gboolean
gst_mpegv_parse_process_sc (GstMpegvParse * mpvparse,
GstBuffer * buf, gint off, guint8 code)
{
gboolean ret = FALSE, packet = TRUE;
g_return_val_if_fail (buf && gst_buffer_get_size (buf) >= 4, FALSE);
GST_LOG_OBJECT (mpvparse, "process startcode %x (%s)", code,
picture_start_code_name (code));
switch (code) {
case GST_MPEG_VIDEO_PACKET_PICTURE:
GST_LOG_OBJECT (mpvparse, "startcode is PICTURE");
/* picture is aggregated with preceding sequence/gop, if any.
* so, picture start code only ends if already a previous one */
if (mpvparse->pic_offset < 0)
mpvparse->pic_offset = off;
else
ret = (off != mpvparse->pic_offset);
/* but it's a valid starting one */
if (off == 4)
ret = TRUE;
break;
case GST_MPEG_VIDEO_PACKET_SEQUENCE:
GST_LOG_OBJECT (mpvparse, "startcode is SEQUENCE");
if (mpvparse->seq_offset < 0)
mpvparse->seq_offset = off;
ret = TRUE;
break;
case GST_MPEG_VIDEO_PACKET_GOP:
GST_LOG_OBJECT (mpvparse, "startcode is GOP");
if (mpvparse->seq_offset >= 0)
ret = mpvparse->gop_split;
else
ret = TRUE;
break;
case GST_MPEG_VIDEO_PACKET_EXTENSION:
GST_LOG_OBJECT (mpvparse, "startcode is VIDEO PACKET EXTENSION");
parse_picture_extension (mpvparse, buf, off);
if (mpvparse->ext_count < G_N_ELEMENTS (mpvparse->ext_offsets))
mpvparse->ext_offsets[mpvparse->ext_count++] = off;
/* fall-through */
default:
packet = FALSE;
break;
}
/* set size to avoid processing config again */
if (mpvparse->seq_offset >= 0 && off != mpvparse->seq_offset &&
!mpvparse->seq_size && packet) {
/* should always be at start */
g_assert (mpvparse->seq_offset <= 4);
gst_mpegv_parse_process_config (mpvparse, buf, off - mpvparse->seq_offset);
mpvparse->seq_size = off - mpvparse->seq_offset;
}
/* extract some picture info if there is any in the frame being terminated */
if (ret && mpvparse->pic_offset >= 0 && mpvparse->pic_offset < off) {
GstMapInfo map;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (gst_mpeg_video_parse_picture_header (&mpvparse->pichdr,
map.data, map.size, mpvparse->pic_offset))
GST_LOG_OBJECT (mpvparse, "picture_coding_type %d (%s), ending"
"frame of size %d", mpvparse->pichdr.pic_type,
picture_type_name (mpvparse->pichdr.pic_type), off - 4);
else
GST_LOG_OBJECT (mpvparse, "Couldn't parse picture at offset %d",
mpvparse->pic_offset);
gst_buffer_unmap (buf, &map);
}
return ret;
}
static void
gst_type_find_element_loop (GstPad * pad)
{
GstTypeFindElement *typefind;
GstFlowReturn ret = GST_FLOW_OK;
typefind = GST_TYPE_FIND_ELEMENT (GST_PAD_PARENT (pad));
if (typefind->need_stream_start) {
gchar *stream_id;
GstEvent *event;
stream_id = gst_pad_create_stream_id (typefind->src,
GST_ELEMENT_CAST (typefind), NULL);
GST_DEBUG_OBJECT (typefind, "Pushing STREAM_START");
event = gst_event_new_stream_start (stream_id);
gst_event_set_group_id (event, gst_util_group_id_next ());
gst_pad_push_event (typefind->src, event);
typefind->need_stream_start = FALSE;
g_free (stream_id);
}
if (typefind->mode == MODE_TYPEFIND) {
GstPad *peer = NULL;
GstCaps *found_caps = NULL;
GstTypeFindProbability probability = GST_TYPE_FIND_NONE;
GST_DEBUG_OBJECT (typefind, "find type in pull mode");
GST_OBJECT_LOCK (typefind);
if (typefind->force_caps) {
found_caps = gst_caps_ref (typefind->force_caps);
probability = GST_TYPE_FIND_MAXIMUM;
}
GST_OBJECT_UNLOCK (typefind);
if (!found_caps) {
peer = gst_pad_get_peer (pad);
if (peer) {
gint64 size;
gchar *ext;
if (!gst_pad_query_duration (peer, GST_FORMAT_BYTES, &size)) {
GST_WARNING_OBJECT (typefind, "Could not query upstream length!");
gst_object_unref (peer);
ret = GST_FLOW_ERROR;
goto pause;
}
/* the size if 0, we cannot continue */
if (size == 0) {
/* keep message in sync with message in sink event handler */
GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND,
(_("Stream contains no data.")), ("Can't typefind empty stream"));
gst_object_unref (peer);
ret = GST_FLOW_ERROR;
goto pause;
}
ext = gst_type_find_get_extension (typefind, pad);
found_caps =
gst_type_find_helper_get_range (GST_OBJECT_CAST (peer),
GST_OBJECT_PARENT (peer),
(GstTypeFindHelperGetRangeFunction) (GST_PAD_GETRANGEFUNC (peer)),
(guint64) size, ext, &probability);
g_free (ext);
GST_DEBUG ("Found caps %" GST_PTR_FORMAT, found_caps);
gst_object_unref (peer);
}
}
if (!found_caps || probability < typefind->min_probability) {
GST_DEBUG ("Trying to guess using extension");
gst_caps_replace (&found_caps, NULL);
found_caps =
gst_type_find_guess_by_extension (typefind, pad, &probability);
}
if (!found_caps || probability < typefind->min_probability) {
GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
gst_caps_replace (&found_caps, NULL);
ret = GST_FLOW_ERROR;
goto pause;
}
GST_DEBUG ("Emiting found caps %" GST_PTR_FORMAT, found_caps);
gst_type_find_element_emit_have_type (typefind, probability, found_caps);
typefind->mode = MODE_NORMAL;
gst_caps_unref (found_caps);
} else if (typefind->mode == MODE_NORMAL) {
GstBuffer *outbuf = NULL;
if (typefind->need_segment) {
typefind->need_segment = FALSE;
gst_pad_push_event (typefind->src,
gst_event_new_segment (&typefind->segment));
}
/* Pull 4k blocks and send downstream */
ret = gst_pad_pull_range (typefind->sink, typefind->offset, 4096, &outbuf);
if (ret != GST_FLOW_OK)
goto pause;
typefind->offset += gst_buffer_get_size (outbuf);
ret = gst_pad_push (typefind->src, outbuf);
if (ret != GST_FLOW_OK)
goto pause;
} else {
/* Error out */
ret = GST_FLOW_ERROR;
goto pause;
}
return;
pause:
{
const gchar *reason = gst_flow_get_name (ret);
gboolean push_eos = FALSE;
GST_LOG_OBJECT (typefind, "pausing task, reason %s", reason);
gst_pad_pause_task (typefind->sink);
if (ret == GST_FLOW_EOS) {
/* perform EOS logic */
if (typefind->segment.flags & GST_SEGMENT_FLAG_SEGMENT) {
gint64 stop;
/* for segment playback we need to post when (in stream time)
* we stopped, this is either stop (when set) or the duration. */
if ((stop = typefind->segment.stop) == -1)
stop = typefind->offset;
GST_LOG_OBJECT (typefind, "Sending segment done, at end of segment");
gst_element_post_message (GST_ELEMENT (typefind),
gst_message_new_segment_done (GST_OBJECT (typefind),
GST_FORMAT_BYTES, stop));
gst_pad_push_event (typefind->src,
gst_event_new_segment_done (GST_FORMAT_BYTES, stop));
} else {
push_eos = TRUE;
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
/* for fatal errors we post an error message */
GST_ELEMENT_ERROR (typefind, STREAM, FAILED, (NULL),
("stream stopped, reason %s", reason));
push_eos = TRUE;
}
if (push_eos) {
/* send EOS, and prevent hanging if no streams yet */
GST_LOG_OBJECT (typefind, "Sending EOS, at end of stream");
gst_pad_push_event (typefind->src, gst_event_new_eos ());
}
return;
}
}
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 > 4) {
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 < 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 GstFlowReturn
gst_interleave_collected (GstCollectPads * pads, GstInterleave * self)
{
guint size;
GstBuffer *outbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
GSList *collected;
guint nsamples;
guint ncollected = 0;
gboolean empty = TRUE;
gint width = self->width / 8;
GstMapInfo write_info;
GstClockTime timestamp = -1;
/* FIXME: send caps and tags after stream-start */
#if 0
if (self->send_stream_start) {
gchar s_id[32];
/* stream-start (FIXME: create id based on input ids) */
g_snprintf (s_id, sizeof (s_id), "interleave-%08x", g_random_int ());
gst_pad_push_event (self->src, gst_event_new_stream_start (s_id));
self->send_stream_start = FALSE;
}
#endif
size = gst_collect_pads_available (pads);
if (size == 0)
goto eos;
g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (self->width > 0, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (self->rate > 0, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (size % width == 0, GST_FLOW_ERROR);
GST_DEBUG_OBJECT (self, "Starting to collect %u bytes from %d channels", size,
self->channels);
nsamples = size / width;
outbuf = gst_buffer_new_allocate (NULL, size * self->channels, NULL);
if (outbuf == NULL || gst_buffer_get_size (outbuf) < size * self->channels) {
gst_buffer_unref (outbuf);
return GST_FLOW_NOT_NEGOTIATED;
}
gst_buffer_map (outbuf, &write_info, GST_MAP_WRITE);
memset (write_info.data, 0, size * self->channels);
for (collected = pads->data; collected != NULL; collected = collected->next) {
GstCollectData *cdata;
GstBuffer *inbuf;
guint8 *outdata;
GstMapInfo input_info;
cdata = (GstCollectData *) collected->data;
inbuf = gst_collect_pads_take_buffer (pads, cdata, size);
if (inbuf == NULL) {
GST_DEBUG_OBJECT (cdata->pad, "No buffer available");
goto next;
}
ncollected++;
gst_buffer_map (inbuf, &input_info, GST_MAP_READ);
if (timestamp == -1)
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))
goto next;
empty = FALSE;
outdata =
write_info.data + width * GST_INTERLEAVE_PAD_CAST (cdata->pad)->channel;
self->func (outdata, input_info.data, self->channels, nsamples);
gst_buffer_unmap (inbuf, &input_info);
next:
if (inbuf)
gst_buffer_unref (inbuf);
}
if (ncollected == 0) {
gst_buffer_unmap (outbuf, &write_info);
goto eos;
}
GST_OBJECT_LOCK (self);
if (self->pending_segment) {
GstEvent *event;
GstSegment segment;
event = self->pending_segment;
self->pending_segment = NULL;
GST_OBJECT_UNLOCK (self);
/* convert the input segment to time now */
gst_event_copy_segment (event, &segment);
if (segment.format != GST_FORMAT_TIME) {
gst_event_unref (event);
/* not time, convert */
switch (segment.format) {
case GST_FORMAT_BYTES:
segment.start *= width;
if (segment.stop != -1)
segment.stop *= width;
if (segment.position != -1)
segment.position *= width;
/* fallthrough for the samples case */
case GST_FORMAT_DEFAULT:
segment.start =
gst_util_uint64_scale_int (segment.start, GST_SECOND, self->rate);
if (segment.stop != -1)
segment.stop =
gst_util_uint64_scale_int (segment.stop, GST_SECOND,
self->rate);
if (segment.position != -1)
segment.position =
gst_util_uint64_scale_int (segment.position, GST_SECOND,
self->rate);
break;
default:
GST_WARNING ("can't convert segment values");
segment.start = 0;
segment.stop = -1;
segment.position = 0;
break;
}
event = gst_event_new_segment (&segment);
}
gst_pad_push_event (self->src, event);
GST_OBJECT_LOCK (self);
}
GST_OBJECT_UNLOCK (self);
if (timestamp != -1) {
self->offset = gst_util_uint64_scale_int (timestamp, self->rate,
GST_SECOND);
self->timestamp = timestamp;
}
GST_BUFFER_TIMESTAMP (outbuf) = self->timestamp;
GST_BUFFER_OFFSET (outbuf) = self->offset;
self->offset += nsamples;
self->timestamp = gst_util_uint64_scale_int (self->offset,
GST_SECOND, self->rate);
GST_BUFFER_DURATION (outbuf) =
self->timestamp - GST_BUFFER_TIMESTAMP (outbuf);
if (empty)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
gst_buffer_unmap (outbuf, &write_info);
GST_LOG_OBJECT (self, "pushing outbuf, timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
ret = gst_pad_push (self->src, outbuf);
return ret;
eos:
{
GST_DEBUG_OBJECT (self, "no data available, must be EOS");
if (outbuf)
gst_buffer_unref (outbuf);
gst_pad_push_event (self->src, gst_event_new_eos ());
return GST_FLOW_EOS;
}
}
static GstFlowReturn
mpegpsmux_create_stream (MpegPsMux * mux, MpegPsPadData * ps_data, GstPad * pad)
{
/* Create a steam. Fill in codec specific information */
GstFlowReturn ret = GST_FLOW_ERROR;
GstCaps *caps;
GstStructure *s;
gboolean is_video = FALSE;
caps = gst_pad_get_current_caps (pad);
if (caps == NULL) {
GST_DEBUG_OBJECT (pad, "Sink pad caps were not set before pushing");
return GST_FLOW_NOT_NEGOTIATED;
}
s = gst_caps_get_structure (caps, 0);
g_return_val_if_fail (s != NULL, FALSE);
if (gst_structure_has_name (s, "video/x-dirac")) {
GST_DEBUG_OBJECT (pad, "Creating Dirac stream");
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_DIRAC);
is_video = TRUE;
} else if (gst_structure_has_name (s, "audio/x-ac3")) {
GST_DEBUG_OBJECT (pad, "Creating AC3 stream");
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_PS_AUDIO_AC3);
} else if (gst_structure_has_name (s, "audio/x-dts")) {
GST_DEBUG_OBJECT (pad, "Creating DTS stream");
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_PS_AUDIO_DTS);
} else if (gst_structure_has_name (s, "audio/x-lpcm")) {
GST_DEBUG_OBJECT (pad, "Creating LPCM stream");
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_PS_AUDIO_LPCM);
} else if (gst_structure_has_name (s, "video/x-h264")) {
const GValue *value;
GST_DEBUG_OBJECT (pad, "Creating H264 stream");
/* Codec data contains SPS/PPS which need to go in stream for valid ES */
value = gst_structure_get_value (s, "codec_data");
if (value) {
ps_data->codec_data = gst_buffer_ref (gst_value_get_buffer (value));
GST_DEBUG_OBJECT (pad, "%" G_GSIZE_FORMAT " bytes of codec data",
gst_buffer_get_size (ps_data->codec_data));
ps_data->prepare_func = mpegpsmux_prepare_h264;
} else {
ps_data->codec_data = NULL;
}
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_H264);
is_video = TRUE;
} else if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion;
if (!gst_structure_get_int (s, "mpegversion", &mpegversion)) {
GST_ELEMENT_ERROR (pad, STREAM, FORMAT,
("Invalid data format presented"),
("Caps with type audio/mpeg did not have mpegversion"));
goto beach;
}
switch (mpegversion) {
case 1:
GST_DEBUG_OBJECT (pad, "Creating MPEG Audio, version 1 stream");
ps_data->stream =
psmux_create_stream (mux->psmux, PSMUX_ST_AUDIO_MPEG1);
break;
case 2:
GST_DEBUG_OBJECT (pad, "Creating MPEG Audio, version 2 stream");
ps_data->stream =
psmux_create_stream (mux->psmux, PSMUX_ST_AUDIO_MPEG2);
break;
case 4:
{
const GValue *value;
/* Codec data contains SPS/PPS which need to go in stream for valid ES */
GST_DEBUG_OBJECT (pad, "Creating MPEG Audio, version 4 stream");
value = gst_structure_get_value (s, "codec_data");
if (value) {
ps_data->codec_data = gst_buffer_ref (gst_value_get_buffer (value));
GST_DEBUG_OBJECT (pad, "%" G_GSIZE_FORMAT " bytes of codec data",
gst_buffer_get_size (ps_data->codec_data));
ps_data->prepare_func = mpegpsmux_prepare_aac;
} else {
ps_data->codec_data = NULL;
}
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_AUDIO_AAC);
break;
}
default:
GST_WARNING_OBJECT (pad, "unsupported mpegversion %d", mpegversion);
goto beach;
}
} else if (gst_structure_has_name (s, "video/mpeg")) {
gint mpegversion;
if (!gst_structure_get_int (s, "mpegversion", &mpegversion)) {
GST_ELEMENT_ERROR (mux, STREAM, FORMAT,
("Invalid data format presented"),
("Caps with type video/mpeg did not have mpegversion"));
goto beach;
}
if (mpegversion == 1) {
GST_DEBUG_OBJECT (pad, "Creating MPEG Video, version 1 stream");
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_MPEG1);
} else if (mpegversion == 2) {
GST_DEBUG_OBJECT (pad, "Creating MPEG Video, version 2 stream");
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_MPEG2);
} else {
GST_DEBUG_OBJECT (pad, "Creating MPEG Video, version 4 stream");
ps_data->stream = psmux_create_stream (mux->psmux, PSMUX_ST_VIDEO_MPEG4);
}
is_video = TRUE;
}
if (ps_data->stream != NULL) {
ps_data->stream_id = ps_data->stream->stream_id;
ps_data->stream_id_ext = ps_data->stream->stream_id_ext;
GST_DEBUG_OBJECT (pad, "Stream created, stream_id=%04x, stream_id_ext=%04x",
ps_data->stream_id, ps_data->stream_id_ext);
gst_structure_get_int (s, "rate", &ps_data->stream->audio_sampling);
gst_structure_get_int (s, "channels", &ps_data->stream->audio_channels);
gst_structure_get_int (s, "bitrate", &ps_data->stream->audio_bitrate);
ret = GST_FLOW_OK;
if (is_video && mux->video_stream_id == 0) {
mux->video_stream_id = ps_data->stream_id;
GST_INFO_OBJECT (mux, "video pad stream_id 0x%02x", mux->video_stream_id);
}
}
beach:
return ret;
}
static GstFlowReturn
gst_raw_base_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
gsize in_size, out_size;
guint frame_size;
guint num_out_frames;
gsize units_n, units_d;
guint64 buffer_duration;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstEvent *new_caps_event = NULL;
GstRawBaseParse *raw_base_parse = GST_RAW_BASE_PARSE (parse);
GstRawBaseParseClass *klass = GST_RAW_BASE_PARSE_GET_CLASS (parse);
g_assert (klass->is_config_ready);
g_assert (klass->get_caps_from_config);
g_assert (klass->get_config_frame_size);
g_assert (klass->get_units_per_second);
/* We never skip any bytes this way. Instead, subclass takes care
* of skipping any overhead (necessary, since the way it needs to
* be skipped is completely subclass specific). */
*skipsize = 0;
/* The operations below access the current config. Protect
* against race conditions by using the object lock. */
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (raw_base_parse);
/* If the source pad caps haven't been set yet, or need to be
* set again, do so now, BEFORE any buffers are pushed out */
if (G_UNLIKELY (!raw_base_parse->src_caps_set)) {
GstCaps *new_src_caps;
if (G_UNLIKELY (!klass->is_config_ready (raw_base_parse,
GST_RAW_BASE_PARSE_CONFIG_CURRENT))) {
/* The current configuration is not ready. No caps can be
* generated out of it.
* The most likely reason for this is that the sink caps config
* is the current one and no valid sink caps have been pushed
* by upstream. Report the problem and exit. */
if (gst_raw_base_parse_is_using_sink_caps (raw_base_parse)) {
goto config_not_ready;
} else {
/* This should not be reached if the property config is active */
g_assert_not_reached ();
}
}
GST_DEBUG_OBJECT (parse,
"setting src caps since this has not been done yet");
/* Convert the current config to a caps structure to
* inform downstream about the new format */
if (!klass->get_caps_from_config (raw_base_parse,
GST_RAW_BASE_PARSE_CONFIG_CURRENT, &new_src_caps)) {
GST_ERROR_OBJECT (raw_base_parse,
"could not get src caps from current config");
flow_ret = GST_FLOW_NOT_NEGOTIATED;
goto error_locked;
}
new_caps_event = gst_event_new_caps (new_src_caps);
gst_caps_unref (new_src_caps);
raw_base_parse->src_caps_set = TRUE;
}
frame_size =
klass->get_config_frame_size (raw_base_parse,
GST_RAW_BASE_PARSE_CONFIG_CURRENT);
in_size = gst_buffer_get_size (frame->buffer);
/* gst_base_parse_set_min_frame_size() is called when the current
* configuration changes and the change affects the frame size. This
* means that a buffer must contain at least as many bytes as indicated
* by the frame size. If there are fewer inside an error occurred;
* either something in the parser went wrong, or the min frame size
* wasn't updated properly. */
g_assert (in_size >= frame_size);
/* Determine how many complete frames would fit in the input buffer.
* Then check if this amount exceeds the maximum number of frames
* as indicated by the subclass. */
num_out_frames = (in_size / frame_size);
if (klass->get_max_frames_per_buffer) {
guint max_num_out_frames = klass->get_max_frames_per_buffer (raw_base_parse,
GST_RAW_BASE_PARSE_CONFIG_CURRENT);
num_out_frames = MIN (num_out_frames, max_num_out_frames);
}
/* Ensure that the size of the buffers that get pushed downstream
* is always an integer multiple of the frame size to prevent cases
* where downstream gets buffers with incomplete frames. */
out_size = num_out_frames * frame_size;
/* Set the overhead size to ensure that timestamping excludes these
* extra overhead bytes. */
frame->overhead =
klass->get_overhead_size ? klass->get_overhead_size (raw_base_parse,
GST_RAW_BASE_PARSE_CONFIG_CURRENT) : 0;
g_assert (out_size >= (guint) (frame->overhead));
out_size -= frame->overhead;
GST_LOG_OBJECT (raw_base_parse,
"%" G_GSIZE_FORMAT " bytes input %" G_GSIZE_FORMAT
" bytes output (%u frame(s)) %d bytes overhead", in_size, out_size,
num_out_frames, frame->overhead);
/* Calculate buffer duration */
klass->get_units_per_second (raw_base_parse, GST_FORMAT_BYTES,
GST_RAW_BASE_PARSE_CONFIG_CURRENT, &units_n, &units_d);
if (units_n == 0 || units_d == 0)
buffer_duration = GST_CLOCK_TIME_NONE;
else
buffer_duration =
gst_util_uint64_scale (out_size, GST_SECOND * units_d, units_n);
if (klass->process) {
GstBuffer *processed_data = NULL;
if (!klass->process (raw_base_parse, GST_RAW_BASE_PARSE_CONFIG_CURRENT,
frame->buffer, in_size, out_size, &processed_data))
goto process_error;
frame->out_buffer = processed_data;
} else {
frame->out_buffer = NULL;
}
/* Set the duration of the output buffer, or if none exists, of
* the input buffer. Do this after the process() call, since in
* case out_buffer is set, the subclass has created a new buffer.
* Instead of requiring subclasses to set the duration (which
* anyway must always be buffer_duration), let's do it here. */
if (frame->out_buffer != NULL)
GST_BUFFER_DURATION (frame->out_buffer) = buffer_duration;
else
GST_BUFFER_DURATION (frame->buffer) = buffer_duration;
/* Access to the current config is not needed in subsequent
* operations, so the lock can be released */
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (raw_base_parse);
/* If any new caps have to be pushed downstrean, do so
* *before* the frame is finished */
if (G_UNLIKELY (new_caps_event != NULL)) {
gst_pad_push_event (GST_BASE_PARSE_SRC_PAD (raw_base_parse),
new_caps_event);
new_caps_event = NULL;
}
gst_base_parse_finish_frame (parse, frame, out_size + frame->overhead);
return flow_ret;
config_not_ready:
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (raw_base_parse);
GST_ELEMENT_ERROR (parse, STREAM, FORMAT,
("sink caps config is the current config, and it is not ready -"
"upstream may not have pushed a caps event yet"), (NULL));
flow_ret = GST_FLOW_ERROR;
goto error_end;
process_error:
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (raw_base_parse);
GST_ELEMENT_ERROR (parse, STREAM, DECODE, ("could not process data"), (NULL));
flow_ret = GST_FLOW_ERROR;
goto error_end;
error_locked:
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (raw_base_parse);
goto error_end;
error_end:
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_DROP;
if (new_caps_event != NULL)
gst_event_unref (new_caps_event);
return flow_ret;
}
static GstFlowReturn
gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstBuffer *buf = NULL;
GstDTMFSrcEvent *event;
GstDTMFSrc *dtmfsrc;
GstClock *clock;
GstClockID *clockid;
GstClockReturn clockret;
dtmfsrc = GST_DTMF_SRC (basesrc);
do {
if (dtmfsrc->last_event == NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
switch (event->event_type) {
case DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
"Received a DTMF stop event when already stopped");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case DTMF_EVENT_TYPE_START:
gst_dtmf_prepare_timestamps (dtmfsrc);
event->packet_count = 0;
dtmfsrc->last_event = event;
event = NULL;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed",
dtmfsrc->last_event);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
if (event)
g_slice_free (GstDTMFSrcEvent, event);
} else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
MIN_DUTY_CYCLE) {
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
switch (event->event_type) {
case DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
"Received two consecutive DTMF start events");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case DTMF_EVENT_TYPE_STOP:
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
g_slice_free (GstDTMFSrcEvent, event);
}
}
} while (dtmfsrc->last_event == NULL);
GST_LOG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
GST_OBJECT_LOCK (dtmfsrc);
if (!dtmfsrc->paused) {
dtmfsrc->clockid = clockid;
GST_OBJECT_UNLOCK (dtmfsrc);
clockret = gst_clock_id_wait (clockid, NULL);
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused)
clockret = GST_CLOCK_UNSCHEDULED;
} else {
clockret = GST_CLOCK_UNSCHEDULED;
}
gst_clock_id_unref (clockid);
dtmfsrc->clockid = NULL;
GST_OBJECT_UNLOCK (dtmfsrc);
if (clockret == GST_CLOCK_UNSCHEDULED) {
goto paused;
}
buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
GST_LOG_OBJECT (dtmfsrc, "Created buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buf));
*buffer = buf;
return GST_FLOW_OK;
paused_locked:
GST_OBJECT_UNLOCK (dtmfsrc);
paused:
if (dtmfsrc->last_event) {
GST_DEBUG_OBJECT (dtmfsrc, "Stopping current event");
/* Don't forget to release the stream lock */
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
}
return GST_FLOW_FLUSHING;
}
static GstFlowReturn
gst_tcp_mix_src_pad_read (GstTCPMixSrcPad * pad, GstBuffer ** outbuf)
{
GstTCPMixSrc *src = GST_TCP_MIX_SRC (GST_PAD_PARENT (pad));
gssize avail, receivedBytes;
GstMapInfo map;
GError *err = NULL;
/* if we have a client, wait for read */
GST_LOG_OBJECT (pad, "asked for a buffer");
if (!pad->client) {
if (src->mode == MODE_LOOP)
goto loop_read;
else
goto no_client;
}
/* read the buffer header */
read_available_bytes:
avail = g_socket_get_available_bytes (pad->client);
if (avail < 0) {
goto socket_get_available_bytes_error;
} else if (avail == 0) {
GIOCondition condition;
if (!g_socket_condition_wait (pad->client,
G_IO_IN | G_IO_PRI | G_IO_ERR | G_IO_HUP, pad->cancellable, &err))
goto socket_condition_wait_error;
condition = g_socket_condition_check (pad->client,
G_IO_IN | G_IO_PRI | G_IO_ERR | G_IO_HUP);
if ((condition & G_IO_ERR))
goto socket_condition_error;
else if ((condition & G_IO_HUP))
goto socket_condition_hup;
avail = g_socket_get_available_bytes (pad->client);
if (avail < 0)
goto socket_get_available_bytes_error;
}
if (0 < avail) {
gsize readBytes = MIN (avail, MAX_READ_SIZE);
*outbuf = gst_buffer_new_and_alloc (readBytes);
gst_buffer_map (*outbuf, &map, GST_MAP_READWRITE);
receivedBytes = g_socket_receive (pad->client, (gchar *) map.data,
readBytes, pad->cancellable, &err);
} else {
/* Connection closed */
receivedBytes = 0;
*outbuf = NULL;
}
if (receivedBytes == 0)
goto socket_connection_closed;
else if (receivedBytes < 0)
goto socket_receive_error;
gst_buffer_unmap (*outbuf, &map);
gst_buffer_resize (*outbuf, 0, receivedBytes);
#if 0
GST_LOG_OBJECT (pad,
"Returning buffer from _get of size %" G_GSIZE_FORMAT
", ts %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT
", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
gst_buffer_get_size (*outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (*outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (*outbuf)),
GST_BUFFER_OFFSET (*outbuf), GST_BUFFER_OFFSET_END (*outbuf));
#endif
g_clear_error (&err);
return GST_FLOW_OK;
/* Handling Errors */
no_client:
{
GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL),
("No client socket (%s)", GST_PAD_NAME (pad)));
if (src->mode == MODE_LOOP)
goto loop_read;
return GST_FLOW_ERROR;
}
socket_get_available_bytes_error:
{
GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL),
("Failed to get available bytes from socket"));
gst_tcp_mix_src_pad_reset (pad);
if (src->mode == MODE_LOOP)
goto loop_read;
return GST_FLOW_ERROR;
}
socket_condition_wait_error:
{
GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL),
("Select failed: %s", err->message));
g_clear_error (&err);
gst_tcp_mix_src_pad_reset (pad);
if (src->mode == MODE_LOOP)
goto loop_read;
return GST_FLOW_ERROR;
}
socket_condition_error:
{
GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL), ("Socket in error state"));
*outbuf = NULL;
gst_tcp_mix_src_pad_reset (pad);
if (src->mode == MODE_LOOP)
goto loop_read;
return GST_FLOW_ERROR;
}
socket_condition_hup:
{
GST_DEBUG_OBJECT (pad, "Connection closed");
*outbuf = NULL;
gst_tcp_mix_src_pad_reset (pad);
if (src->mode == MODE_LOOP)
goto loop_read;
return GST_FLOW_EOS;
}
socket_connection_closed:
{
GST_DEBUG_OBJECT (pad, "Connection closed");
if (*outbuf) {
gst_buffer_unmap (*outbuf, &map);
gst_buffer_unref (*outbuf);
}
*outbuf = NULL;
gst_tcp_mix_src_pad_reset (pad);
if (src->mode == MODE_LOOP)
goto loop_read;
return GST_FLOW_EOS;
}
socket_receive_error:
{
gst_buffer_unmap (*outbuf, &map);
gst_buffer_unref (*outbuf);
*outbuf = NULL;
if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
GST_DEBUG_OBJECT (pad, "Cancelled reading from socket");
if (src->mode == MODE_LOOP)
goto loop_read;
return GST_FLOW_FLUSHING;
} else {
GST_ELEMENT_ERROR (pad, RESOURCE, READ, (NULL),
("Failed to read from socket: %s", err->message));
if (src->mode == MODE_LOOP)
goto loop_read;
return GST_FLOW_ERROR;
}
}
loop_read:
{
#if 0
GstEvent *event = gst_event_new_flush_start ();
if (!gst_pad_push_event (pad, event)) {
GST_ERROR_OBJECT (src, "Failed to flush data on %s.%s",
GST_ELEMENT_NAME (src), GST_PAD_NAME (pad));
}
#endif
#if 0
GST_DEBUG_OBJECT (pad, "Looping");
#endif
if (src->fill == FILL_NONE) {
gst_tcp_mix_src_pad_wait_for_client (pad);
goto read_available_bytes;
}
enum
{ buffer_size = 1024 };
*outbuf = gst_buffer_new_and_alloc (buffer_size);
switch (src->fill) {
case FILL_ZERO:
break;
case FILL_RAND:
{
guchar *p;
gst_buffer_map (*outbuf, &map, GST_MAP_READWRITE);
for (p = map.data; p < map.data + buffer_size; p += 4) {
*((int *) p) = rand ();
}
} break;
}
return GST_FLOW_OK;
}
}
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;
}
static GstFlowReturn
gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
basepayload, GstBuffer * buffer)
{
GstRTPBaseAudioPayload *payload;
GstRTPBaseAudioPayloadPrivate *priv;
guint payload_len;
GstFlowReturn ret;
guint available;
guint min_payload_len;
guint max_payload_len;
guint align;
guint size;
gboolean discont;
GstClockTime timestamp;
ret = GST_FLOW_OK;
payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
priv = payload->priv;
timestamp = GST_BUFFER_PTS (buffer);
discont = GST_BUFFER_IS_DISCONT (buffer);
if (discont) {
GST_DEBUG_OBJECT (payload, "Got DISCONT");
/* flush everything out of the adapter, mark DISCONT */
ret = gst_rtp_base_audio_payload_flush (payload, -1, -1);
priv->discont = TRUE;
/* get the distance between the timestamp gap and produce the same gap in
* the RTP timestamps */
if (priv->last_timestamp != -1 && timestamp != -1) {
/* we had a last timestamp, compare it to the new timestamp and update the
* offset counter for RTP timestamps. The effect is that we will produce
* output buffers containing the same RTP timestamp gap as the gap
* between the GST timestamps. */
if (timestamp > priv->last_timestamp) {
GstClockTime diff;
guint64 bytes;
/* we're only going to apply a positive gap, otherwise we let the marker
* bit do its thing. simply convert to bytes and add the current
* offset */
diff = timestamp - priv->last_timestamp;
bytes = priv->time_to_bytes (payload, diff);
priv->offset += bytes;
GST_DEBUG_OBJECT (payload,
"elapsed time %" GST_TIME_FORMAT ", bytes %" G_GUINT64_FORMAT
", new offset %" G_GUINT64_FORMAT, GST_TIME_ARGS (diff), bytes,
priv->offset);
}
}
}
if (!gst_rtp_base_audio_payload_get_lengths (basepayload,
gst_rtp_base_payload_get_source_count (basepayload, buffer),
&min_payload_len, &max_payload_len, &align))
goto config_error;
GST_DEBUG_OBJECT (payload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
size = gst_buffer_get_size (buffer);
/* shortcut, we don't need to use the adapter when the packet can be pushed
* through directly. */
available = gst_adapter_available (priv->adapter);
GST_DEBUG_OBJECT (payload, "got buffer size %u, available %u",
size, available);
if (available == 0 && (size >= min_payload_len && size <= max_payload_len) &&
(size % align == 0)) {
/* If buffer fits on an RTP packet, let's just push it through
* this will check against max_ptime and max_mtu */
GST_DEBUG_OBJECT (payload, "Fast packet push");
ret = gst_rtp_base_audio_payload_push_buffer (payload, buffer, timestamp);
} else {
/* push the buffer in the adapter */
gst_adapter_push (priv->adapter, buffer);
available += size;
GST_DEBUG_OBJECT (payload, "available now %u", available);
/* as long as we have full frames */
/* TODO: Use buffer lists here */
while (available >= min_payload_len) {
/* get multiple of alignment */
payload_len = MIN (max_payload_len, available);
payload_len = ALIGN_DOWN (payload_len, align);
/* and flush out the bytes from the adapter, automatically set the
* timestamp. */
ret = gst_rtp_base_audio_payload_flush (payload, payload_len, -1);
available -= payload_len;
GST_DEBUG_OBJECT (payload, "available after push %u", available);
}
}
return ret;
/* ERRORS */
config_error:
{
GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL),
("subclass did not configure us properly"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_monoscope_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
GstMonoscope *monoscope;
monoscope = GST_MONOSCOPE (parent);
if (monoscope->rate == 0) {
gst_buffer_unref (inbuf);
flow_ret = GST_FLOW_NOT_NEGOTIATED;
goto out;
}
/* Make sure have an output format */
flow_ret = ensure_negotiated (monoscope);
if (flow_ret != GST_FLOW_OK) {
gst_buffer_unref (inbuf);
goto out;
}
/* don't try to combine samples from discont buffer */
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (monoscope->adapter);
monoscope->next_ts = GST_CLOCK_TIME_NONE;
}
/* Match timestamps from the incoming audio */
if (GST_BUFFER_TIMESTAMP (inbuf) != GST_CLOCK_TIME_NONE)
monoscope->next_ts = GST_BUFFER_TIMESTAMP (inbuf);
GST_LOG_OBJECT (monoscope, "in buffer has %d samples, ts=%" GST_TIME_FORMAT,
gst_buffer_get_size (inbuf) / monoscope->bps,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)));
gst_adapter_push (monoscope->adapter, inbuf);
inbuf = NULL;
/* Collect samples until we have enough for an output frame */
while (flow_ret == GST_FLOW_OK) {
gint16 *samples;
GstBuffer *outbuf = NULL;
guint32 *pixels, avail, bytesperframe;
avail = gst_adapter_available (monoscope->adapter);
GST_LOG_OBJECT (monoscope, "bytes avail now %u", avail);
bytesperframe = monoscope->spf * monoscope->bps;
if (avail < bytesperframe)
break;
/* FIXME: something is wrong with QoS, we are skipping way too much
* stuff even with very low CPU loads */
#if 0
if (monoscope->next_ts != -1) {
gboolean need_skip;
gint64 qostime;
qostime = gst_segment_to_running_time (&monoscope->segment,
GST_FORMAT_TIME, monoscope->next_ts);
GST_OBJECT_LOCK (monoscope);
/* check for QoS, don't compute buffers that are known to be late */
need_skip =
GST_CLOCK_TIME_IS_VALID (monoscope->earliest_time) &&
qostime <= monoscope->earliest_time;
GST_OBJECT_UNLOCK (monoscope);
if (need_skip) {
GST_WARNING_OBJECT (monoscope,
"QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT,
GST_TIME_ARGS (qostime), GST_TIME_ARGS (monoscope->earliest_time));
goto skip;
}
}
#endif
samples = (gint16 *) gst_adapter_map (monoscope->adapter, bytesperframe);
if (monoscope->spf < 512) {
gint16 in_data[512], i;
for (i = 0; i < 512; ++i) {
gdouble off;
off = ((gdouble) i * (gdouble) monoscope->spf) / 512.0;
in_data[i] = samples[MIN ((guint) off, monoscope->spf)];
}
pixels = monoscope_update (monoscope->visstate, in_data);
} else {
/* not really correct, but looks much prettier */
pixels = monoscope_update (monoscope->visstate, samples);
}
GST_LOG_OBJECT (monoscope, "allocating output buffer");
flow_ret = gst_buffer_pool_acquire_buffer (monoscope->pool, &outbuf, NULL);
if (flow_ret != GST_FLOW_OK) {
gst_adapter_unmap (monoscope->adapter);
goto out;
}
gst_buffer_fill (outbuf, 0, pixels, monoscope->outsize);
GST_BUFFER_TIMESTAMP (outbuf) = monoscope->next_ts;
GST_BUFFER_DURATION (outbuf) = monoscope->frame_duration;
flow_ret = gst_pad_push (monoscope->srcpad, outbuf);
#if 0
skip:
#endif
if (GST_CLOCK_TIME_IS_VALID (monoscope->next_ts))
monoscope->next_ts += monoscope->frame_duration;
gst_adapter_flush (monoscope->adapter, bytesperframe);
}
out:
return flow_ret;
}
static void
gst_wildmidi_loop (GstPad * sinkpad)
{
GstWildmidi *wildmidi = GST_WILDMIDI (GST_PAD_PARENT (sinkpad));
GstFlowReturn ret;
switch (wildmidi->state) {
case GST_WILDMIDI_STATE_LOAD:
{
GstBuffer *buffer = NULL;
GST_DEBUG_OBJECT (wildmidi, "loading song");
ret =
gst_pad_pull_range (wildmidi->sinkpad, wildmidi->offset, -1, &buffer);
if (ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (wildmidi, "Song loaded");
wildmidi->state = GST_WILDMIDI_STATE_PARSE;
} else if (ret != GST_FLOW_OK) {
GST_ELEMENT_ERROR (wildmidi, STREAM, DECODE, (NULL),
("Unable to read song"));
goto pause;
} else {
GST_DEBUG_OBJECT (wildmidi, "pushing buffer");
gst_adapter_push (wildmidi->adapter, buffer);
wildmidi->offset += gst_buffer_get_size (buffer);
}
break;
}
case GST_WILDMIDI_STATE_PARSE:
ret = gst_wildmidi_parse_song (wildmidi);
if (ret != GST_FLOW_OK)
goto pause;
wildmidi->state = GST_WILDMIDI_STATE_PLAY;
break;
case GST_WILDMIDI_STATE_PLAY:
ret = gst_wildmidi_do_play (wildmidi);
if (ret != GST_FLOW_OK)
goto pause;
break;
default:
break;
}
return;
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GstEvent *event;
GST_DEBUG_OBJECT (wildmidi, "pausing task, reason %s", reason);
gst_pad_pause_task (sinkpad);
if (ret == GST_FLOW_EOS) {
/* perform EOS logic */
event = gst_event_new_eos ();
gst_pad_push_event (wildmidi->srcpad, event);
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
event = gst_event_new_eos ();
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (wildmidi, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused, reason %s (%d)", reason, ret));
gst_pad_push_event (wildmidi->srcpad, event);
}
}
}
/* FIXME: this entire function looks completely wrong, if there is or
* was a bug in GStreamer it should be fixed there (tpm) */
static GstPadProbeReturn
brasero_transcode_buffer_handler (GstPad *pad,
GstPadProbeInfo *info,
gpointer user_data)
{
BraseroTranscodePrivate *priv;
BraseroTranscode *self = user_data;
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info);
GstPad *peer;
gint64 size;
priv = BRASERO_TRANSCODE_PRIVATE (self);
size = gst_buffer_get_size (buffer);
if (priv->segment_start <= 0 && priv->segment_end <= 0)
return GST_PAD_PROBE_OK;
/* what we do here is more or less what gstreamer does when seeking:
* it reads and process from 0 to the seek position (I tried).
* It even forwards the data before the seek position to the sink (which
* is a problem in our case as it would be written) */
if (priv->size > priv->segment_end) {
priv->size += size;
return GST_PAD_PROBE_DROP;
}
if (priv->size + size > priv->segment_end) {
GstBuffer *new_buffer;
int data_size;
/* the entire the buffer is not interesting for us */
/* create a new buffer and push it on the pad:
* NOTE: we're going to receive it ... */
data_size = priv->segment_end - priv->size;
new_buffer = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_METADATA, 0, data_size);
/* FIXME: we can now modify the probe buffer in 0.11 */
/* Recursive: the following calls ourselves BEFORE we finish */
peer = gst_pad_get_peer (pad);
gst_pad_push (peer, new_buffer);
priv->size += size - data_size;
/* post an EOS event to stop pipeline */
gst_pad_push_event (peer, gst_event_new_eos ());
gst_object_unref (peer);
return GST_PAD_PROBE_DROP;
}
/* see if the buffer is in the segment */
if (priv->size < priv->segment_start) {
GstBuffer *new_buffer;
gint data_size;
/* see if all the buffer is interesting for us */
if (priv->size + size < priv->segment_start) {
priv->size += size;
return GST_PAD_PROBE_DROP;
}
/* create a new buffer and push it on the pad:
* NOTE: we're going to receive it ... */
data_size = priv->size + size - priv->segment_start;
new_buffer = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_METADATA, size - data_size, data_size);
/* FIXME: this looks dodgy (tpm) */
GST_BUFFER_TIMESTAMP (new_buffer) = GST_BUFFER_TIMESTAMP (buffer) + data_size;
/* move forward by the size of bytes we dropped */
priv->size += size - data_size;
/* FIXME: we can now modify the probe buffer in 0.11 */
/* this is recursive the following calls ourselves
* BEFORE we finish */
peer = gst_pad_get_peer (pad);
gst_pad_push (peer, new_buffer);
gst_object_unref (peer);
return GST_PAD_PROBE_DROP;
}
priv->size += size;
priv->pos += size;
return GST_PAD_PROBE_OK;
}
static GstFlowReturn
gst_identity_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstIdentity *identity = GST_IDENTITY (trans);
GstClockTime rundts = GST_CLOCK_TIME_NONE;
GstClockTime runpts = GST_CLOCK_TIME_NONE;
GstClockTime ts, duration, runtimestamp;
gsize size;
size = gst_buffer_get_size (buf);
if (identity->check_imperfect_timestamp)
gst_identity_check_imperfect_timestamp (identity, buf);
if (identity->check_imperfect_offset)
gst_identity_check_imperfect_offset (identity, buf);
/* update prev values */
identity->prev_timestamp = GST_BUFFER_TIMESTAMP (buf);
identity->prev_duration = GST_BUFFER_DURATION (buf);
identity->prev_offset_end = GST_BUFFER_OFFSET_END (buf);
identity->prev_offset = GST_BUFFER_OFFSET (buf);
if (identity->error_after >= 0) {
identity->error_after--;
if (identity->error_after == 0)
goto error_after;
}
if (identity->drop_probability > 0.0) {
if ((gfloat) (1.0 * rand () / (RAND_MAX)) < identity->drop_probability)
goto dropped;
}
if (identity->dump) {
GstMapInfo info;
gst_buffer_map (buf, &info, GST_MAP_READ);
gst_util_dump_mem (info.data, info.size);
gst_buffer_unmap (buf, &info);
}
if (!identity->silent) {
gst_identity_update_last_message_for_buffer (identity, "chain", buf, size);
}
if (identity->datarate > 0) {
GstClockTime time = gst_util_uint64_scale_int (identity->offset,
GST_SECOND, identity->datarate);
GST_BUFFER_PTS (buf) = GST_BUFFER_DTS (buf) = time;
GST_BUFFER_DURATION (buf) = size * GST_SECOND / identity->datarate;
}
if (identity->signal_handoffs)
g_signal_emit (identity, gst_identity_signals[SIGNAL_HANDOFF], 0, buf);
if (trans->segment.format == GST_FORMAT_TIME) {
rundts = gst_segment_to_running_time (&trans->segment,
GST_FORMAT_TIME, GST_BUFFER_DTS (buf));
runpts = gst_segment_to_running_time (&trans->segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (buf));
}
if (GST_CLOCK_TIME_IS_VALID (rundts))
runtimestamp = rundts;
else if (GST_CLOCK_TIME_IS_VALID (runpts))
runtimestamp = runpts;
else
runtimestamp = 0;
ret = gst_identity_do_sync (identity, runtimestamp);
identity->offset += size;
if (identity->sleep_time && ret == GST_FLOW_OK)
g_usleep (identity->sleep_time);
if (identity->single_segment && (trans->segment.format == GST_FORMAT_TIME)
&& (ret == GST_FLOW_OK)) {
GST_BUFFER_DTS (buf) = rundts;
GST_BUFFER_PTS (buf) = runpts;
GST_BUFFER_OFFSET (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET_END (buf) = GST_CLOCK_TIME_NONE;
}
return ret;
/* ERRORS */
error_after:
{
GST_ELEMENT_ERROR (identity, CORE, FAILED,
(_("Failed after iterations as requested.")), (NULL));
return GST_FLOW_ERROR;
}
dropped:
{
if (!identity->silent) {
gst_identity_update_last_message_for_buffer (identity, "dropping", buf,
size);
}
ts = GST_BUFFER_TIMESTAMP (buf);
if (GST_CLOCK_TIME_IS_VALID (ts)) {
duration = GST_BUFFER_DURATION (buf);
gst_pad_push_event (GST_BASE_TRANSFORM_SRC_PAD (identity),
gst_event_new_gap (ts, duration));
}
/* return DROPPED to basetransform. */
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gsize size;
guint8 *data;
GstBuffer *outbuf, *bufd;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
GstMapInfo map, omap;
GstAudioClippingMeta *cmeta = NULL;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
if (!gst_opus_dec_negotiate (dec, NULL))
return GST_FLOW_NOT_NEGOTIATED;
}
if (dec->n_channels == 2 && dec->n_streams == 1
&& dec->n_stereo_streams == 0) {
/* if we are automatically decoding 2 channels, but only have
a single encoded one, direct both channels to it */
dec->channel_mapping[1] = 0;
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_GST_DEBUG
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
"Mapping table", dec->n_channels, dec->channel_mapping);
#endif
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
dec->n_stereo_streams);
dec->state =
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = (dec->use_inband_fec
&& gst_buffer_get_size (dec->last_buffer) >
0) ? dec->last_buffer : buffer;
/* That's the buffer we get duration from */
bufd = dec->use_inband_fec ? dec->last_buffer : buffer;
if (buf && gst_buffer_get_size (buf) > 0) {
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
if (gst_buffer_get_size (bufd) == 0) {
GstClockTime const opus_plc_alignment = 2500 * GST_USECOND;
GstClockTime aligned_missing_duration;
GstClockTime missing_duration = GST_BUFFER_DURATION (bufd);
if (!GST_CLOCK_TIME_IS_VALID (missing_duration) || missing_duration == 0) {
if (GST_CLOCK_TIME_IS_VALID (dec->last_known_buffer_duration)) {
missing_duration = dec->last_known_buffer_duration;
GST_WARNING_OBJECT (dec,
"Missing duration, using last duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (missing_duration));
} else {
GST_WARNING_OBJECT (dec,
"Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms");
missing_duration = 20 * GST_MSECOND;
}
}
GST_DEBUG_OBJECT (dec,
"missing buffer, doing PLC duration %" GST_TIME_FORMAT
" plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration),
GST_TIME_ARGS (dec->leftover_plc_duration));
/* add the leftover PLC duration to that of the buffer */
missing_duration += dec->leftover_plc_duration;
/* align the combined buffer and leftover PLC duration to multiples
* of 2.5ms, rounding to nearest, and store excess duration for later */
aligned_missing_duration =
((missing_duration +
opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment;
dec->leftover_plc_duration = missing_duration - aligned_missing_duration;
/* Opus' PLC cannot operate with less than 2.5ms; skip PLC
* and accumulate the missing duration in the leftover_plc_duration
* for the next PLC attempt */
if (aligned_missing_duration < opus_plc_alignment) {
GST_DEBUG_OBJECT (dec,
"current duration %" GST_TIME_FORMAT
" of missing data not enough for PLC (minimum needed: %"
GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration),
GST_TIME_ARGS (opus_plc_alignment));
goto done;
}
/* convert the duration (in nanoseconds) to sample count */
samples =
gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate,
GST_SECOND);
GST_DEBUG_OBJECT (dec,
"calculated PLC frame length: %" GST_TIME_FORMAT
" num frame samples: %d new leftover: %" GST_TIME_FORMAT,
GST_TIME_ARGS (aligned_missing_duration), samples,
GST_TIME_ARGS (dec->leftover_plc_duration));
} else {
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
}
packet_size = samples * dec->n_channels * 2;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
if (size > 0)
dec->last_known_buffer_duration = packet_duration_opus (data, size);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
do {
if (dec->use_inband_fec) {
if (gst_buffer_get_size (dec->last_buffer) > 0) {
/* normal delayed decode */
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
}
if (n == OPUS_BUFFER_TOO_SMALL) {
/* if too small, add 2.5 milliseconds and try again, up to the
* Opus max size of 120 milliseconds */
if (samples >= 120 * dec->sample_rate / 1000)
break;
samples += 25 * dec->sample_rate / 10000;
packet_size = samples * dec->n_channels * 2;
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
}
} while (n == OPUS_BUFFER_TOO_SMALL);
gst_buffer_unmap (outbuf, &omap);
if (data != NULL)
gst_buffer_unmap (buf, &map);
if (n < 0) {
GstFlowReturn ret = GST_FLOW_ERROR;
gst_buffer_unref (outbuf);
GST_AUDIO_DECODER_ERROR (dec, 1, STREAM, DECODE, (NULL),
("Decoding error (%d): %s", n, opus_strerror (n)), ret);
return ret;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
GST_BUFFER_DURATION (outbuf) = samples * GST_SECOND / dec->sample_rate;
samples = n;
cmeta = gst_buffer_get_audio_clipping_meta (buf);
g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);
/* Skip any samples that need skipping */
if (cmeta && cmeta->start) {
guint pre_skip = cmeta->start;
guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
GST_INFO_OBJECT (dec,
"Skipping %u samples at the beginning (%u at 48000 Hz)",
skip, scaled_skip);
}
if (cmeta && cmeta->end) {
guint post_skip = cmeta->end;
guint scaled_post_skip = post_skip * dec->sample_rate / 48000;
guint skip = scaled_post_skip > n ? n : scaled_post_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
guint outsize = gst_buffer_get_size (outbuf);
guint skip_bytes = skip * 2 * dec->n_channels;
if (outsize > skip_bytes)
outsize -= skip_bytes;
else
outsize = 0;
gst_buffer_resize (outbuf, 0, outsize);
GST_INFO_OBJECT (dec,
"Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip);
}
if (gst_buffer_get_size (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
dec->n_channels, dec->opus_pos, dec->info.position);
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
gsize rsize;
unsigned int i, nsamples;
double volume = dec->r128_gain_volume;
gint16 *samples;
gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
samples = (gint16 *) omap.data;
rsize = omap.size;
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
nsamples = rsize / 2;
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
gst_buffer_unmap (outbuf, &omap);
}
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
}
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
done:
return res;
creation_failed:
GST_ELEMENT_ERROR (dec, LIBRARY, INIT, ("Failed to create Opus decoder"),
("Failed to create Opus decoder (%d): %s", err, opus_strerror (err)));
return GST_FLOW_ERROR;
buffer_failed:
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to create %u byte buffer", packet_size));
return GST_FLOW_ERROR;
}
static GstFlowReturn
gst_cutter_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstCutter *filter;
GstMapInfo map;
gint16 *in_data;
gint bpf, rate;
gsize in_size;
guint num_samples;
gdouble NCS = 0.0; /* Normalized Cumulative Square of buffer */
gdouble RMS = 0.0; /* RMS of signal in buffer */
gdouble NMS = 0.0; /* Normalized Mean Square of buffer */
GstBuffer *prebuf; /* pointer to a prebuffer element */
GstClockTime duration;
filter = GST_CUTTER (parent);
if (GST_AUDIO_INFO_FORMAT (&filter->info) == GST_AUDIO_FORMAT_UNKNOWN)
goto not_negotiated;
bpf = GST_AUDIO_INFO_BPF (&filter->info);
rate = GST_AUDIO_INFO_RATE (&filter->info);
gst_buffer_map (buf, &map, GST_MAP_READ);
in_data = (gint16 *) map.data;
in_size = map.size;
GST_LOG_OBJECT (filter, "length of prerec buffer: %" GST_TIME_FORMAT,
GST_TIME_ARGS (filter->pre_run_length));
/* calculate mean square value on buffer */
switch (GST_AUDIO_INFO_FORMAT (&filter->info)) {
case GST_AUDIO_FORMAT_S16:
num_samples = in_size / 2;
gst_cutter_calculate_gint16 (in_data, num_samples, &NCS);
NMS = NCS / num_samples;
break;
case GST_AUDIO_FORMAT_S8:
num_samples = in_size;
gst_cutter_calculate_gint8 ((gint8 *) in_data, num_samples, &NCS);
NMS = NCS / num_samples;
break;
default:
/* this shouldn't happen */
g_warning ("no mean square function for format");
break;
}
gst_buffer_unmap (buf, &map);
filter->silent_prev = filter->silent;
duration = gst_util_uint64_scale (in_size / bpf, GST_SECOND, rate);
RMS = sqrt (NMS);
/* if RMS below threshold, add buffer length to silent run length count
* if not, reset
*/
GST_LOG_OBJECT (filter, "buffer stats: NMS %f, RMS %f, audio length %f", NMS,
RMS, gst_guint64_to_gdouble (duration));
if (RMS < filter->threshold_level)
filter->silent_run_length += gst_guint64_to_gdouble (duration);
else {
filter->silent_run_length = 0 * GST_SECOND;
filter->silent = FALSE;
}
if (filter->silent_run_length > filter->threshold_length)
/* it has been silent long enough, flag it */
filter->silent = TRUE;
/* has the silent status changed ? if so, send right signal
* and, if from silent -> not silent, flush pre_record buffer
*/
if (filter->silent != filter->silent_prev) {
if (filter->silent) {
GstMessage *m =
gst_cutter_message_new (filter, FALSE, GST_BUFFER_TIMESTAMP (buf));
GST_DEBUG_OBJECT (filter, "signaling CUT_STOP");
gst_element_post_message (GST_ELEMENT (filter), m);
} else {
gint count = 0;
GstMessage *m =
gst_cutter_message_new (filter, TRUE, GST_BUFFER_TIMESTAMP (buf));
GST_DEBUG_OBJECT (filter, "signaling CUT_START");
gst_element_post_message (GST_ELEMENT (filter), m);
/* first of all, flush current buffer */
GST_DEBUG_OBJECT (filter, "flushing buffer of length %" GST_TIME_FORMAT,
GST_TIME_ARGS (filter->pre_run_length));
while (filter->pre_buffer) {
prebuf = (g_list_first (filter->pre_buffer))->data;
filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
gst_pad_push (filter->srcpad, prebuf);
++count;
}
GST_DEBUG_OBJECT (filter, "flushed %d buffers", count);
filter->pre_run_length = 0 * GST_SECOND;
}
}
/* now check if we have to send the new buffer to the internal buffer cache
* or to the srcpad */
if (filter->silent) {
filter->pre_buffer = g_list_append (filter->pre_buffer, buf);
filter->pre_run_length += gst_guint64_to_gdouble (duration);
while (filter->pre_run_length > filter->pre_length) {
GstClockTime pduration;
gsize psize;
prebuf = (g_list_first (filter->pre_buffer))->data;
g_assert (GST_IS_BUFFER (prebuf));
psize = gst_buffer_get_size (prebuf);
pduration = gst_util_uint64_scale (psize / bpf, GST_SECOND, rate);
filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
filter->pre_run_length -= gst_guint64_to_gdouble (pduration);
/* only pass buffers if we don't leak */
if (!filter->leaky)
ret = gst_pad_push (filter->srcpad, prebuf);
else
gst_buffer_unref (prebuf);
}
} else
ret = gst_pad_push (filter->srcpad, buf);
return ret;
/* ERRORS */
not_negotiated:
{
return GST_FLOW_NOT_NEGOTIATED;
}
}
/*
* Adds buffer sizes together.
*/
static void
buffer_count_size (void *buffer, void *user_data)
{
guint *sum = (guint *) user_data;
*sum += gst_buffer_get_size (buffer);
}
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 gboolean
gst_mpegv_parse_process_config (GstMpegvParse * mpvparse, GstBuffer * buf,
guint size)
{
guint8 *data;
guint8 *data_with_prefix;
GstMapInfo map;
gint i, offset;
if (mpvparse->seq_offset < 4) {
/* This shouldn't happen, but just in case... */
GST_WARNING_OBJECT (mpvparse, "Sequence header start code missing.");
return FALSE;
}
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data + mpvparse->seq_offset;
g_assert (size <= map.size);
/* pointer to sequence header data including the start code prefix -
used for codec private data */
data_with_prefix = data - 4;
/* only do stuff if something new; only compare first 11 bytes, changes in
quantiser matrix doesn't matter here. Also changing the matrices in
codec_data seems to cause problem with decoders */
if (mpvparse->config && size == gst_buffer_get_size (mpvparse->config) &&
gst_buffer_memcmp (mpvparse->config, 0, data_with_prefix, MIN (size,
11)) == 0) {
gst_buffer_unmap (buf, &map);
return TRUE;
}
if (!gst_mpeg_video_parse_sequence_header (&mpvparse->sequencehdr, data,
size - mpvparse->seq_offset, 0)) {
GST_DEBUG_OBJECT (mpvparse,
"failed to parse config data (size %d) at offset %d",
size, mpvparse->seq_offset);
gst_buffer_unmap (buf, &map);
return FALSE;
}
GST_LOG_OBJECT (mpvparse, "accepting parsed config size %d", size);
/* Set mpeg version, and parse sequence extension */
mpvparse->config_flags = FLAG_NONE;
for (i = 0; i < mpvparse->ext_count; ++i) {
offset = mpvparse->ext_offsets[i];
mpvparse->config_flags |= FLAG_MPEG2;
if (offset < size) {
if (gst_mpeg_video_parse_sequence_extension (&mpvparse->sequenceext,
map.data, size, offset)) {
GST_LOG_OBJECT (mpvparse, "Read Sequence Extension");
mpvparse->config_flags |= FLAG_SEQUENCE_EXT;
} else
if (gst_mpeg_video_parse_sequence_display_extension
(&mpvparse->sequencedispext, map.data, size, offset)) {
GST_LOG_OBJECT (mpvparse, "Read Sequence Display Extension");
mpvparse->config_flags |= FLAG_SEQUENCE_DISPLAY_EXT;
}
}
}
if (mpvparse->config_flags & FLAG_MPEG2) {
/* Update the sequence header based on extensions */
GstMpegVideoSequenceExt *seqext = NULL;
GstMpegVideoSequenceDisplayExt *seqdispext = NULL;
if (mpvparse->config_flags & FLAG_SEQUENCE_EXT)
seqext = &mpvparse->sequenceext;
if (mpvparse->config_flags & FLAG_SEQUENCE_DISPLAY_EXT)
seqdispext = &mpvparse->sequencedispext;
gst_mpeg_video_finalise_mpeg2_sequence_header (&mpvparse->sequencehdr,
seqext, seqdispext);
}
if (mpvparse->fps_num == 0 || mpvparse->fps_den == 0) {
mpvparse->fps_num = mpvparse->sequencehdr.fps_n;
mpvparse->fps_den = mpvparse->sequencehdr.fps_d;
}
/* parsing ok, so accept it as new config */
if (mpvparse->config != NULL)
gst_buffer_unref (mpvparse->config);
mpvparse->config = gst_buffer_new_and_alloc (size);
gst_buffer_fill (mpvparse->config, 0, data_with_prefix, size);
/* trigger src caps update */
mpvparse->update_caps = TRUE;
gst_buffer_unmap (buf, &map);
return TRUE;
}
/* Fill */
static GstFlowReturn gst_android_video_source_fill(GstPushSrc * p_pushsrc, GstBuffer * p_buf)
{
GstAndroidVideoSource *p_src;
int vcd_ret;
static struct timeval time_of_day;
static struct timeval window_time;
static struct timeval start_time;
int timeDiffUsec;
static gint frame_count = 0;
static gint frame_count_window = 0;
GstBuffer *p_outbuf;
GstMapInfo mem_info;
gboolean ok;
GA_LOGTRACE("ENTER %s --xx--> thread(%ld)", __FUNCTION__, pthread_self());
p_src = GST_ANDROIDVIDEOSOURCE(p_pushsrc);
if (gst_buffer_get_size(p_buf) != p_src->m_bufSize) {
GA_LOGWARN("%s: WARNING: gst_buffer_get_size(p_buf)==%d != p_src->m_bufSize==%d", __FUNCTION__, gst_buffer_get_size(p_buf), p_src->m_bufSize);
goto fill_error_negotiation;
}
VCD_checkChangeCamera(p_src->m_devHandle);
if (!p_src->vcdStarted) {
AV_CHECK_ERR(VCD_start(p_src->m_devHandle), fill_error_vcd_start);
p_src->vcdStarted = TRUE;
}
if (!frame_count) { // Only first time
gettimeofday(&start_time, NULL);
gettimeofday(&window_time, NULL);
}
frame_count++;
frame_count_window++;
gettimeofday(&time_of_day, NULL);
timeDiffUsec = time_diff_usec(&time_of_day, &window_time);
if (timeDiffUsec > p_src->log_interval || !p_src->log_interval) {
int framerate;
int framerateWindow;
int timeDiffSec;
timeDiffSec = time_diff_sec(&time_of_day, &start_time);
framerate = frame_count / (timeDiffSec > 0 ? timeDiffSec : 1);
framerateWindow = frame_count_window * 1000000 / timeDiffUsec;
GA_LOGVERB("%s ------> has now been called %d times --Create--> framerate since start: %d fps, framerate last %d usec: %d fps", __FUNCTION__, frame_count, framerate, timeDiffUsec, framerateWindow);
gettimeofday(&window_time, NULL);
frame_count_window = 0;
}
g_warn_if_fail(gst_buffer_is_writable(p_buf)); /* g_warn_if_fail() used for internal error (exception to our rules for this special "buffer copying case") */
g_assert(gst_buffer_is_writable(p_buf)); /* this buf should be allocated in the base class and should always be writable */
p_outbuf = gst_buffer_make_writable(p_buf); /* do this cause we never wanna crash in release even if somebody makes a mistake somewhere... */
ok = gst_buffer_map(p_outbuf, &mem_info, GST_MAP_WRITE);
if (!ok) {
goto fill_error_gst_buffer_map;
}
vcd_ret = VCD_read(p_src->m_devHandle, &(mem_info.data), mem_info.size);
if (vcd_ret == VCD_ERR_NO_DATA) {
// This should never happen. There should always be more data from the device.
// In any case, if it happens we don't want to end or lock or anything, we just
// want to go on as if there actually were data. Specifically, we do not want
// to block the streaming thread...
memset(mem_info.data, 0, mem_info.size);
gst_buffer_unmap(p_outbuf, &mem_info);
GST_BUFFER_OFFSET(p_outbuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END(p_outbuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_PTS(p_outbuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DTS(p_outbuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION(p_outbuf) = GST_CLOCK_TIME_NONE;
GA_LOGWARN("%s: WARNING: Returning GST_FLOW_OK with a buffer with zeros...", __FUNCTION__);
return GST_FLOW_OK;
}
if (vcd_ret != VCD_NO_ERROR) {
gst_buffer_unmap(p_outbuf, &mem_info);
goto fill_error_read;
}
gst_buffer_unmap(p_outbuf, &mem_info);
set_gstbuf_time_and_offset(p_src, p_outbuf);
GA_LOGTRACE("EXIT %s", __FUNCTION__);
return GST_FLOW_OK;
/*
* propagate unhandled errors
*/
fill_error_read:
{
GA_LOGERROR("%s: Error when reading data from device!", __FUNCTION__);
return GST_FLOW_ERROR;
}
fill_error_negotiation:
{
GA_LOGERROR("%s: ERROR: Strange buffer size. Negotiation not done? Disallowed renegotiation done?", __FUNCTION__);
return GST_FLOW_ERROR;
}
fill_error_gst_buffer_map:
{
GA_LOGERROR("%s: gst_buffer_map() failed!", __FUNCTION__);
return GST_FLOW_ERROR;
}
fill_error_vcd_start:
{
GA_LOGERROR("%s: FATAL ERROR: Could not start the video device!", __FUNCTION__);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_fake_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstFakeSink *sink = GST_FAKE_SINK_CAST (bsink);
if (sink->num_buffers_left == 0)
goto eos;
if (sink->num_buffers_left != -1)
sink->num_buffers_left--;
if (!sink->silent) {
gchar dts_str[64], pts_str[64], dur_str[64];
gchar *flag_str, *meta_str;
GST_OBJECT_LOCK (sink);
g_free (sink->last_message);
if (GST_BUFFER_DTS (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (dts_str, sizeof (dts_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DTS (buf)));
} else {
g_strlcpy (dts_str, "none", sizeof (dts_str));
}
if (GST_BUFFER_PTS (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (pts_str, sizeof (pts_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
} else {
g_strlcpy (pts_str, "none", sizeof (pts_str));
}
if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (dur_str, sizeof (dur_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
} else {
g_strlcpy (dur_str, "none", sizeof (dur_str));
}
flag_str = gst_buffer_get_flags_string (buf);
meta_str = gst_buffer_get_meta_string (buf);
sink->last_message =
g_strdup_printf ("chain ******* (%s:%s) (%u bytes, dts: %s, pts: %s"
", duration: %s, offset: %" G_GINT64_FORMAT ", offset_end: %"
G_GINT64_FORMAT ", flags: %08x %s, meta: %s) %p",
GST_DEBUG_PAD_NAME (GST_BASE_SINK_CAST (sink)->sinkpad),
(guint) gst_buffer_get_size (buf), dts_str, pts_str,
dur_str, GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf),
GST_MINI_OBJECT_CAST (buf)->flags, flag_str,
meta_str ? meta_str : "none", buf);
g_free (flag_str);
g_free (meta_str);
GST_OBJECT_UNLOCK (sink);
gst_fake_sink_notify_last_message (sink);
}
if (sink->signal_handoffs)
g_signal_emit (sink, gst_fake_sink_signals[SIGNAL_HANDOFF], 0, buf,
bsink->sinkpad);
if (sink->dump) {
GstMapInfo info;
if (gst_buffer_map (buf, &info, GST_MAP_READ)) {
gst_util_dump_mem (info.data, info.size);
gst_buffer_unmap (buf, &info);
}
}
if (sink->num_buffers_left == 0)
goto eos;
return GST_FLOW_OK;
/* ERRORS */
eos:
{
GST_DEBUG_OBJECT (sink, "we are EOS");
return GST_FLOW_EOS;
}
}
static GstFlowReturn
gst_wavenc_write_toc (GstWavEnc * wavenc)
{
GList *list;
GstToc *toc;
GstTocEntry *entry, *subentry;
GstBuffer *buf;
GstMapInfo map;
guint8 *data;
guint32 ncues, size, cues_size, labls_size, notes_size;
if (!wavenc->toc) {
GST_DEBUG_OBJECT (wavenc, "have no toc, checking toc_setter");
wavenc->toc = gst_toc_setter_get_toc (GST_TOC_SETTER (wavenc));
}
if (!wavenc->toc) {
GST_WARNING_OBJECT (wavenc, "have no toc");
return GST_FLOW_OK;
}
toc = gst_toc_ref (wavenc->toc);
size = 0;
cues_size = 0;
labls_size = 0;
notes_size = 0;
/* check if the TOC entries is valid */
list = gst_toc_get_entries (toc);
entry = list->data;
if (gst_toc_entry_is_alternative (entry)) {
list = gst_toc_entry_get_sub_entries (entry);
while (list) {
subentry = list->data;
if (!gst_toc_entry_is_sequence (subentry))
return FALSE;
list = g_list_next (list);
}
list = gst_toc_entry_get_sub_entries (entry);
}
if (gst_toc_entry_is_sequence (entry)) {
while (list) {
entry = list->data;
if (!gst_toc_entry_is_sequence (entry))
return FALSE;
list = g_list_next (list);
}
list = gst_toc_get_entries (toc);
}
ncues = g_list_length (list);
GST_DEBUG_OBJECT (wavenc, "number of cue entries: %d", ncues);
while (list) {
guint32 id = 0;
gint64 id64;
const gchar *uid;
entry = list->data;
uid = gst_toc_entry_get_uid (entry);
id64 = g_ascii_strtoll (uid, NULL, 0);
/* check if id unique compatible with guint32 else generate random */
if (id64 >= 0 && gst_wavenc_is_cue_id_unique (id64, wavenc->cues)) {
id = (guint32) id64;
} else {
do {
id = g_random_int ();
} while (!gst_wavenc_is_cue_id_unique (id, wavenc->cues));
}
gst_wavenc_parse_cue (wavenc, id, entry);
gst_wavenc_parse_labl (wavenc, id, entry);
gst_wavenc_parse_note (wavenc, id, entry);
list = g_list_next (list);
}
/* count cues size */
if (wavenc->cues) {
cues_size = 24 * g_list_length (wavenc->cues);
size += 12 + cues_size;
} else {
GST_WARNING_OBJECT (wavenc, "cue's not found");
return FALSE;
}
/* count labls size */
if (wavenc->labls) {
list = wavenc->labls;
while (list) {
GstWavEncLabl *labl;
labl = list->data;
labls_size += 8 + GST_ROUND_UP_2 (labl->chunk_data_size);
list = g_list_next (list);
}
size += labls_size;
}
/* count notes size */
if (wavenc->notes) {
list = wavenc->notes;
while (list) {
GstWavEncNote *note;
note = list->data;
notes_size += 8 + GST_ROUND_UP_2 (note->chunk_data_size);
list = g_list_next (list);
}
size += notes_size;
}
if (wavenc->labls || wavenc->notes) {
size += 12;
}
buf = gst_buffer_new_and_alloc (size);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
data = map.data;
memset (data, 0, size);
/* write Cue Chunk */
if (wavenc->cues) {
memcpy (data, (gchar *) "cue ", 4);
GST_WRITE_UINT32_LE (data + 4, 4 + cues_size);
GST_WRITE_UINT32_LE (data + 8, ncues);
data += 12;
gst_wavenc_write_cues (&data, wavenc->cues);
/* write Associated Data List Chunk */
if (wavenc->labls || wavenc->notes) {
memcpy (data, (gchar *) "LIST", 4);
GST_WRITE_UINT32_LE (data + 4, 4 + labls_size + notes_size);
memcpy (data + 8, (gchar *) "adtl", 4);
data += 12;
if (wavenc->labls)
gst_wavenc_write_labls (&data, wavenc->labls);
if (wavenc->notes)
gst_wavenc_write_notes (&data, wavenc->notes);
}
}
/* free resources */
if (toc)
gst_toc_unref (toc);
if (wavenc->cues)
g_list_free_full (wavenc->cues, g_free);
if (wavenc->labls)
g_list_free_full (wavenc->labls, g_free);
if (wavenc->notes)
g_list_free_full (wavenc->notes, g_free);
gst_buffer_unmap (buf, &map);
wavenc->meta_length += gst_buffer_get_size (buf);
return gst_pad_push (wavenc->srcpad, buf);
}
static GstFlowReturn
gst_rtp_vraw_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
{
GstRtpVRawPay *rtpvrawpay;
GstFlowReturn ret = GST_FLOW_OK;
gfloat packets_per_packline;
guint pgroups_per_packet;
guint packlines_per_list, buffers_per_list;
guint lines_delay; /* after how many packed lines we push out a buffer list */
guint last_line; /* last pack line number we pushed out a buffer list */
guint line, offset;
guint8 *p0, *yp, *up, *vp;
guint ystride, uvstride;
guint xinc, yinc;
guint pgroup;
guint mtu;
guint width, height;
gint field, fields;
GstVideoFormat format;
GstVideoFrame frame;
gint interlaced;
gboolean use_buffer_lists;
GstBufferList *list = NULL;
GstRTPBuffer rtp = { NULL, };
rtpvrawpay = GST_RTP_VRAW_PAY (payload);
if (!gst_video_frame_map (&frame, &rtpvrawpay->vinfo, buffer, GST_MAP_READ)) {
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
GST_LOG_OBJECT (rtpvrawpay, "new frame of %" G_GSIZE_FORMAT " bytes",
gst_buffer_get_size (buffer));
/* get pointer and strides of the planes */
p0 = GST_VIDEO_FRAME_PLANE_DATA (&frame, 0);
yp = GST_VIDEO_FRAME_COMP_DATA (&frame, 0);
up = GST_VIDEO_FRAME_COMP_DATA (&frame, 1);
vp = GST_VIDEO_FRAME_COMP_DATA (&frame, 2);
ystride = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 0);
uvstride = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 1);
mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);
/* amount of bytes for one pixel */
pgroup = rtpvrawpay->pgroup;
width = GST_VIDEO_INFO_WIDTH (&rtpvrawpay->vinfo);
height = GST_VIDEO_INFO_HEIGHT (&rtpvrawpay->vinfo);
interlaced = GST_VIDEO_INFO_IS_INTERLACED (&rtpvrawpay->vinfo);
format = GST_VIDEO_INFO_FORMAT (&rtpvrawpay->vinfo);
yinc = rtpvrawpay->yinc;
xinc = rtpvrawpay->xinc;
/* after how many packed lines we push out a buffer list */
lines_delay = GST_ROUND_UP_4 (height / rtpvrawpay->chunks_per_frame);
/* calculate how many buffers we expect to store in a single buffer list */
pgroups_per_packet = (mtu - (12 + 14)) / pgroup;
packets_per_packline = width / (xinc * pgroups_per_packet * 1.0);
packlines_per_list = height / (yinc * rtpvrawpay->chunks_per_frame);
buffers_per_list = packlines_per_list * packets_per_packline;
buffers_per_list = GST_ROUND_UP_8 (buffers_per_list);
use_buffer_lists = (rtpvrawpay->chunks_per_frame < (height / yinc));
fields = 1 + interlaced;
/* start with line 0, offset 0 */
for (field = 0; field < fields; field++) {
line = field;
offset = 0;
last_line = 0;
if (use_buffer_lists)
list = gst_buffer_list_new_sized (buffers_per_list);
/* write all lines */
while (line < height) {
guint left, pack_line;
GstBuffer *out;
guint8 *outdata, *headers;
gboolean next_line, complete = FALSE;
guint length, cont, pixels;
/* get the max allowed payload length size, we try to fill the complete MTU */
left = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
out = gst_rtp_buffer_new_allocate (left, 0, 0);
if (field == 0) {
GST_BUFFER_PTS (out) = GST_BUFFER_PTS (buffer);
} else {
GST_BUFFER_PTS (out) = GST_BUFFER_PTS (buffer) +
GST_BUFFER_DURATION (buffer) / 2;
}
gst_rtp_buffer_map (out, GST_MAP_WRITE, &rtp);
outdata = gst_rtp_buffer_get_payload (&rtp);
GST_LOG_OBJECT (rtpvrawpay, "created buffer of size %u for MTU %u", left,
mtu);
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Extended Sequence Number | Length |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* |F| Line No |C| Offset |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Length |F| Line No |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* |C| Offset | .
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ .
* . .
* . Two (partial) lines of video data .
* . .
* +---------------------------------------------------------------+
*/
/* need 2 bytes for the extended sequence number */
*outdata++ = 0;
*outdata++ = 0;
left -= 2;
/* the headers start here */
headers = outdata;
/* make sure we can fit at least *one* header and pixel */
if (!(left > (6 + pgroup))) {
gst_rtp_buffer_unmap (&rtp);
gst_buffer_unref (out);
goto too_small;
}
/* while we can fit at least one header and one pixel */
while (left > (6 + pgroup)) {
/* we need a 6 bytes header */
left -= 6;
/* get how may bytes we need for the remaining pixels */
pixels = width - offset;
length = (pixels * pgroup) / xinc;
if (left >= length) {
/* pixels and header fit completely, we will write them and skip to the
* next line. */
next_line = TRUE;
} else {
/* line does not fit completely, see how many pixels fit */
pixels = (left / pgroup) * xinc;
length = (pixels * pgroup) / xinc;
next_line = FALSE;
}
GST_LOG_OBJECT (rtpvrawpay, "filling %u bytes in %u pixels", length,
pixels);
left -= length;
/* write length */
*outdata++ = (length >> 8) & 0xff;
*outdata++ = length & 0xff;
/* write line no */
*outdata++ = ((line >> 8) & 0x7f) | ((field << 7) & 0x80);
*outdata++ = line & 0xff;
if (next_line) {
/* go to next line we do this here to make the check below easier */
line += yinc;
}
/* calculate continuation marker */
cont = (left > (6 + pgroup) && line < height) ? 0x80 : 0x00;
/* write offset and continuation marker */
*outdata++ = ((offset >> 8) & 0x7f) | cont;
*outdata++ = offset & 0xff;
if (next_line) {
/* reset offset */
offset = 0;
GST_LOG_OBJECT (rtpvrawpay, "go to next line %u", line);
} else {
offset += pixels;
GST_LOG_OBJECT (rtpvrawpay, "next offset %u", offset);
}
if (!cont)
break;
}
GST_LOG_OBJECT (rtpvrawpay, "consumed %u bytes",
(guint) (outdata - headers));
/* second pass, read headers and write the data */
while (TRUE) {
guint offs, lin;
/* read length and cont */
length = (headers[0] << 8) | headers[1];
lin = ((headers[2] & 0x7f) << 8) | headers[3];
offs = ((headers[4] & 0x7f) << 8) | headers[5];
cont = headers[4] & 0x80;
pixels = length / pgroup;
headers += 6;
GST_LOG_OBJECT (payload,
"writing length %u, line %u, offset %u, cont %d", length, lin, offs,
cont);
switch (format) {
case GST_VIDEO_FORMAT_RGB:
case GST_VIDEO_FORMAT_RGBA:
case GST_VIDEO_FORMAT_BGR:
case GST_VIDEO_FORMAT_BGRA:
case GST_VIDEO_FORMAT_UYVY:
case GST_VIDEO_FORMAT_UYVP:
offs /= xinc;
memcpy (outdata, p0 + (lin * ystride) + (offs * pgroup), length);
outdata += length;
break;
case GST_VIDEO_FORMAT_AYUV:
{
gint i;
guint8 *datap;
datap = p0 + (lin * ystride) + (offs * 4);
for (i = 0; i < pixels; i++) {
*outdata++ = datap[2];
*outdata++ = datap[1];
*outdata++ = datap[3];
datap += 4;
}
break;
}
case GST_VIDEO_FORMAT_I420:
{
gint i;
guint uvoff;
guint8 *yd1p, *yd2p, *udp, *vdp;
yd1p = yp + (lin * ystride) + (offs);
yd2p = yd1p + ystride;
uvoff = (lin / yinc * uvstride) + (offs / xinc);
udp = up + uvoff;
vdp = vp + uvoff;
for (i = 0; i < pixels; i++) {
*outdata++ = *yd1p++;
*outdata++ = *yd1p++;
*outdata++ = *yd2p++;
*outdata++ = *yd2p++;
*outdata++ = *udp++;
*outdata++ = *vdp++;
}
break;
}
case GST_VIDEO_FORMAT_Y41B:
{
gint i;
guint uvoff;
guint8 *ydp, *udp, *vdp;
ydp = yp + (lin * ystride) + offs;
uvoff = (lin / yinc * uvstride) + (offs / xinc);
udp = up + uvoff;
vdp = vp + uvoff;
for (i = 0; i < pixels; i++) {
*outdata++ = *udp++;
*outdata++ = *ydp++;
*outdata++ = *ydp++;
*outdata++ = *vdp++;
*outdata++ = *ydp++;
*outdata++ = *ydp++;
}
break;
}
default:
gst_rtp_buffer_unmap (&rtp);
gst_buffer_unref (out);
goto unknown_sampling;
}
if (!cont)
break;
}
if (line >= height) {
GST_LOG_OBJECT (rtpvrawpay, "field/frame complete, set marker");
gst_rtp_buffer_set_marker (&rtp, TRUE);
complete = TRUE;
}
gst_rtp_buffer_unmap (&rtp);
if (left > 0) {
GST_LOG_OBJECT (rtpvrawpay, "we have %u bytes left", left);
gst_buffer_resize (out, 0, gst_buffer_get_size (out) - left);
}
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpvrawpay), out, buffer,
g_quark_from_static_string (GST_META_TAG_VIDEO_STR));
/* Now either push out the buffer directly */
if (!use_buffer_lists) {
ret = gst_rtp_base_payload_push (payload, out);
continue;
}
/* or add the buffer to buffer list ... */
gst_buffer_list_add (list, out);
/* .. and check if we need to push out the list */
pack_line = (line - field) / fields;
if (complete || (pack_line > last_line && pack_line % lines_delay == 0)) {
GST_LOG_OBJECT (rtpvrawpay, "pushing list of %u buffers up to pack "
"line %u", gst_buffer_list_length (list), pack_line);
ret = gst_rtp_base_payload_push_list (payload, list);
list = NULL;
if (!complete)
list = gst_buffer_list_new_sized (buffers_per_list);
last_line = pack_line;
}
}
}
gst_video_frame_unmap (&frame);
gst_buffer_unref (buffer);
return ret;
/* ERRORS */
unknown_sampling:
{
GST_ELEMENT_ERROR (payload, STREAM, FORMAT,
(NULL), ("unimplemented sampling"));
gst_video_frame_unmap (&frame);
gst_buffer_unref (buffer);
return GST_FLOW_NOT_SUPPORTED;
}
too_small:
{
GST_ELEMENT_ERROR (payload, RESOURCE, NO_SPACE_LEFT,
(NULL), ("not enough space to send at least one pixel"));
gst_video_frame_unmap (&frame);
gst_buffer_unref (buffer);
return GST_FLOW_NOT_SUPPORTED;
}
}
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_bz2enc_chain (GstPad * pad, GstObject * parent, GstBuffer * in)
{
GstFlowReturn flow = GST_FLOW_OK;
GstBuffer *out;
GstBz2enc *b;
guint n;
int bz2_ret;
GstMapInfo map, omap;
b = GST_BZ2ENC (parent);
if (!b->ready)
goto not_ready;
gst_buffer_map (in, &map, GST_MAP_READ);
b->stream.next_in = (char *) map.data;
b->stream.avail_in = map.size;
while (b->stream.avail_in) {
out = gst_buffer_new_and_alloc (b->buffer_size);
gst_buffer_map (out, &omap, GST_MAP_WRITE);
b->stream.next_out = (char *) omap.data;
b->stream.avail_out = omap.size;
bz2_ret = BZ2_bzCompress (&b->stream, BZ_RUN);
gst_buffer_unmap (out, &omap);
if (bz2_ret != BZ_RUN_OK)
goto compress_error;
n = gst_buffer_get_size (out);
if (b->stream.avail_out >= n) {
gst_buffer_unref (out);
break;
}
gst_buffer_resize (out, 0, n - b->stream.avail_out);
n = gst_buffer_get_size (out);
GST_BUFFER_OFFSET (out) = b->stream.total_out_lo32 - n;
flow = gst_pad_push (b->src, out);
if (flow != GST_FLOW_OK)
break;
b->offset += n;
}
done:
gst_buffer_unmap (in, &map);
gst_buffer_unref (in);
return flow;
/* ERRORS */
not_ready:
{
GST_ELEMENT_ERROR (b, LIBRARY, FAILED, (NULL), ("Compressor not ready."));
flow = GST_FLOW_FLUSHING;
goto done;
}
compress_error:
{
GST_ELEMENT_ERROR (b, STREAM, ENCODE, (NULL),
("Failed to compress data (error code %i)", bz2_ret));
gst_bz2enc_compress_init (b);
gst_buffer_unref (out);
flow = GST_FLOW_ERROR;
goto done;
}
}
/**
* gst_audio_buffer_clip:
* @buffer: (transfer full): The buffer to clip.
* @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
* the buffer should be clipped.
* @rate: sample rate.
* @bpf: size of one audio frame in bytes. This is the size of one sample
* * channels.
*
* Clip the buffer to the given %GstSegment.
*
* After calling this function the caller does not own a reference to
* @buffer anymore.
*
* Returns: (transfer full): %NULL if the buffer is completely outside the configured segment,
* otherwise the clipped buffer is returned.
*
* If the buffer has no timestamp, it is assumed to be inside the segment and
* is not clipped
*/
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
gint bpf)
{
GstBuffer *ret;
GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;
gsize trim, size, osize;
gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end =
TRUE;
g_return_val_if_fail (segment->format == GST_FORMAT_TIME ||
segment->format == GST_FORMAT_DEFAULT, buffer);
g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
/* No timestamp - assume the buffer is completely in the segment */
return buffer;
/* Get copies of the buffer metadata to change later.
* Calculate the missing values for the calculations,
* they won't be changed later though. */
trim = 0;
osize = size = gst_buffer_get_size (buffer);
/* no data, nothing to clip */
if (!size)
return buffer;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
duration = GST_BUFFER_DURATION (buffer);
} else {
change_duration = FALSE;
duration = gst_util_uint64_scale (size / bpf, GST_SECOND, rate);
}
if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
offset = GST_BUFFER_OFFSET (buffer);
} else {
change_offset = FALSE;
offset = 0;
}
if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) {
offset_end = GST_BUFFER_OFFSET_END (buffer);
} else {
change_offset_end = FALSE;
offset_end = offset + size / bpf;
}
if (segment->format == GST_FORMAT_TIME) {
/* Handle clipping for GST_FORMAT_TIME */
guint64 start, stop, cstart, cstop, diff;
start = timestamp;
stop = timestamp + duration;
if (gst_segment_clip (segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
timestamp = cstart;
if (change_duration)
duration -= diff;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset)
offset += diff;
trim += diff * bpf;
size -= diff * bpf;
}
diff = stop - cstop;
if (diff > 0) {
/* duration is always valid if stop is valid */
duration -= diff;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset_end)
offset_end -= diff;
size -= diff * bpf;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
} else {
/* Handle clipping for GST_FORMAT_DEFAULT */
guint64 start, stop, cstart, cstop, diff;
g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer);
start = offset;
stop = offset_end;
if (gst_segment_clip (segment, GST_FORMAT_DEFAULT,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
offset = cstart;
timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate);
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
trim += diff * bpf;
size -= diff * bpf;
}
diff = stop - cstop;
if (diff > 0) {
offset_end = cstop;
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
size -= diff * bpf;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
}
if (trim == 0 && size == osize) {
ret = buffer;
if (GST_BUFFER_TIMESTAMP (ret) != timestamp) {
ret = gst_buffer_make_writable (ret);
GST_BUFFER_TIMESTAMP (ret) = timestamp;
}
if (GST_BUFFER_DURATION (ret) != duration) {
ret = gst_buffer_make_writable (ret);
GST_BUFFER_DURATION (ret) = duration;
}
} else {
/* Get a writable buffer and apply all changes */
GST_DEBUG ("trim %" G_GSIZE_FORMAT " size %" G_GSIZE_FORMAT, trim, size);
ret = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, trim, size);
gst_buffer_unref (buffer);
GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
GST_BUFFER_TIMESTAMP (ret) = timestamp;
if (change_duration)
GST_BUFFER_DURATION (ret) = duration;
if (change_offset)
GST_BUFFER_OFFSET (ret) = offset;
if (change_offset_end)
GST_BUFFER_OFFSET_END (ret) = offset_end;
}
return ret;
}