static GstFlowReturn
gst_jpeg_parse_push_buffer (GstJpegParse * parse, guint len)
{
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
gboolean header_ok;
/* reset the offset (only when we flushed) */
parse->priv->last_offset = 0;
parse->priv->last_entropy_len = 0;
outbuf = gst_adapter_take_buffer (parse->priv->adapter, len);
if (outbuf == NULL) {
GST_ELEMENT_ERROR (parse, STREAM, DECODE,
("Failed to take buffer of size %u", len),
("Failed to take buffer of size %u", len));
return GST_FLOW_ERROR;
}
header_ok = gst_jpeg_parse_read_header (parse, outbuf);
if (parse->priv->new_segment == TRUE
|| parse->priv->width != parse->priv->caps_width
|| parse->priv->height != parse->priv->caps_height
|| parse->priv->framerate_numerator !=
parse->priv->caps_framerate_numerator
|| parse->priv->framerate_denominator !=
parse->priv->caps_framerate_denominator) {
if (!gst_jpeg_parse_set_new_caps (parse, header_ok)) {
GST_ELEMENT_ERROR (parse, CORE, NEGOTIATION,
("Can't set caps to the src pad"), ("Can't set caps to the src pad"));
return GST_FLOW_ERROR;
}
if (parse->priv->tags) {
GST_DEBUG_OBJECT (parse, "Pushing tags: %" GST_PTR_FORMAT,
parse->priv->tags);
gst_element_found_tags_for_pad (GST_ELEMENT_CAST (parse),
parse->priv->srcpad, parse->priv->tags);
parse->priv->tags = NULL;
}
parse->priv->new_segment = FALSE;
parse->priv->caps_width = parse->priv->width;
parse->priv->caps_height = parse->priv->height;
parse->priv->caps_framerate_numerator = parse->priv->framerate_numerator;
parse->priv->caps_framerate_denominator =
parse->priv->framerate_denominator;
}
GST_BUFFER_TIMESTAMP (outbuf) = parse->priv->next_ts;
if (parse->priv->has_fps && GST_CLOCK_TIME_IS_VALID (parse->priv->next_ts)
&& GST_CLOCK_TIME_IS_VALID (parse->priv->duration)) {
parse->priv->next_ts += parse->priv->duration;
} else {
parse->priv->duration = GST_CLOCK_TIME_NONE;
parse->priv->next_ts = GST_CLOCK_TIME_NONE;
}
GST_BUFFER_DURATION (outbuf) = parse->priv->duration;
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (parse->priv->srcpad));
GST_LOG_OBJECT (parse, "pushing buffer (ts=%" GST_TIME_FORMAT ", len=%u)",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), len);
ret = gst_pad_push (parse->priv->srcpad, outbuf);
return ret;
}
/* chain function
* this function does the actual processing
*/
static GstFlowReturn
gst_ivf_parse_chain (GstPad * pad, GstBuffer * buf)
{
GstIvfParse *ivf = GST_IVF_PARSE (GST_OBJECT_PARENT (pad));
gboolean res;
/* lazy creation of the adapter */
if (G_UNLIKELY (ivf->adapter == NULL)) {
ivf->adapter = gst_adapter_new ();
}
GST_LOG_OBJECT (ivf, "Pushing buffer of size %u to adapter",
GST_BUFFER_SIZE (buf));
gst_adapter_push (ivf->adapter, buf); /* adapter takes ownership of buf */
res = GST_FLOW_OK;
switch (ivf->state) {
case GST_IVF_PARSE_START:
if (gst_adapter_available (ivf->adapter) >= 32) {
GstCaps *caps;
const guint8 *data = gst_adapter_peek (ivf->adapter, 32);
guint32 magic = GST_READ_UINT32_LE (data);
guint16 version = GST_READ_UINT16_LE (data + 4);
guint16 header_size = GST_READ_UINT16_LE (data + 6);
guint32 fourcc = GST_READ_UINT32_LE (data + 8);
guint16 width = GST_READ_UINT16_LE (data + 12);
guint16 height = GST_READ_UINT16_LE (data + 14);
guint32 rate_num = GST_READ_UINT32_LE (data + 16);
guint32 rate_den = GST_READ_UINT32_LE (data + 20);
#ifndef GST_DISABLE_GST_DEBUG
guint32 num_frames = GST_READ_UINT32_LE (data + 24);
#endif
/* last 4 bytes unused */
gst_adapter_flush (ivf->adapter, 32);
if (magic != GST_MAKE_FOURCC ('D', 'K', 'I', 'F') ||
version != 0 || header_size != 32 ||
fourcc != GST_MAKE_FOURCC ('V', 'P', '8', '0')) {
GST_ELEMENT_ERROR (ivf, STREAM, WRONG_TYPE, (NULL), (NULL));
return GST_FLOW_ERROR;
}
/* create src pad caps */
caps = gst_caps_new_simple ("video/x-vp8",
"width", G_TYPE_INT, width, "height", G_TYPE_INT, height,
"framerate", GST_TYPE_FRACTION, rate_num, rate_den, NULL);
GST_INFO_OBJECT (ivf, "Found stream: %" GST_PTR_FORMAT, caps);
GST_LOG_OBJECT (ivf, "Stream has %d frames", num_frames);
gst_pad_set_caps (ivf->srcpad, caps);
gst_caps_unref (caps);
/* keep framerate in instance for convenience */
ivf->rate_num = rate_num;
ivf->rate_den = rate_den;
gst_pad_push_event (ivf->srcpad, gst_event_new_new_segment (FALSE, 1.0,
GST_FORMAT_TIME, 0, -1, 0));
/* move along */
ivf->state = GST_IVF_PARSE_DATA;
} else {
GST_LOG_OBJECT (ivf, "Header data not yet available.");
break;
}
/* fall through */
case GST_IVF_PARSE_DATA:
while (gst_adapter_available (ivf->adapter) > 12) {
const guint8 *data = gst_adapter_peek (ivf->adapter, 12);
guint32 frame_size = GST_READ_UINT32_LE (data);
guint64 frame_pts = GST_READ_UINT64_LE (data + 4);
GST_LOG_OBJECT (ivf,
"Read frame header: size %u, pts %" G_GUINT64_FORMAT, frame_size,
frame_pts);
if (gst_adapter_available (ivf->adapter) >= 12 + frame_size) {
GstBuffer *frame;
gst_adapter_flush (ivf->adapter, 12);
frame = gst_adapter_take_buffer (ivf->adapter, frame_size);
gst_buffer_set_caps (frame, GST_PAD_CAPS (ivf->srcpad));
GST_BUFFER_TIMESTAMP (frame) =
gst_util_uint64_scale_int (GST_SECOND * frame_pts, ivf->rate_den,
ivf->rate_num);
GST_BUFFER_DURATION (frame) =
gst_util_uint64_scale_int (GST_SECOND, ivf->rate_den,
ivf->rate_num);
GST_DEBUG_OBJECT (ivf, "Pushing frame of size %u, ts %"
GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT ", off %"
G_GUINT64_FORMAT ", off_end %" G_GUINT64_FORMAT,
GST_BUFFER_SIZE (frame),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (frame)),
GST_TIME_ARGS (GST_BUFFER_DURATION (frame)),
GST_BUFFER_OFFSET (frame), GST_BUFFER_OFFSET_END (frame));
res = gst_pad_push (ivf->srcpad, frame);
if (res != GST_FLOW_OK)
break;
} else {
GST_LOG_OBJECT (ivf, "Frame data not yet available.");
break;
}
}
break;
default:
g_return_val_if_reached (GST_FLOW_ERROR);
}
return res;
}
static GstFlowReturn
gst_rtp_h263p_pay_flush (GstRtpH263PPay * rtph263ppay)
{
guint avail;
GstBufferList *list = NULL;
GstBuffer *outbuf = NULL;
GstFlowReturn ret;
gboolean fragmented = FALSE;
avail = gst_adapter_available (rtph263ppay->adapter);
if (avail == 0)
return GST_FLOW_OK;
fragmented = FALSE;
/* This algorithm assumes the H263/+/++ encoder sends complete frames in each
* buffer */
/* With Fragmentation Mode at GST_FRAGMENTATION_MODE_NORMAL:
* This algorithm implements the Follow-on packets method for packetization.
* This assumes low packet loss network.
* With Fragmentation Mode at GST_FRAGMENTATION_MODE_SYNC:
* This algorithm separates large frames at synchronisation points (Segments)
* (See RFC 4629 section 6). It would be interesting to have a property such as network
* quality to select between both packetization methods */
/* TODO Add VRC supprt (See RFC 4629 section 5.2) */
while (avail > 0) {
guint towrite;
guint8 *payload;
gint header_len;
guint next_gop = 0;
gboolean found_gob = FALSE;
GstRTPBuffer rtp = { NULL };
GstBuffer *payload_buf;
if (rtph263ppay->fragmentation_mode == GST_FRAGMENTATION_MODE_SYNC) {
/* start after 1st gop possible */
/* Check if we have a gob or eos , eossbs */
/* FIXME EOS and EOSSBS packets should never contain any gobs and vice-versa */
next_gop =
gst_adapter_masked_scan_uint32 (rtph263ppay->adapter, 0xffff8000,
0x00008000, 0, avail);
if (next_gop == 0) {
GST_DEBUG_OBJECT (rtph263ppay, " Found GOB header");
found_gob = TRUE;
}
/* Find next and cut the packet accordingly */
/* TODO we should get as many gobs as possible until MTU is reached, this
* code seems to just get one GOB per packet */
if (next_gop == 0 && avail > 3)
next_gop =
gst_adapter_masked_scan_uint32 (rtph263ppay->adapter, 0xffff8000,
0x00008000, 3, avail - 3);
GST_DEBUG_OBJECT (rtph263ppay, " Next GOB Detected at : %d", next_gop);
if (next_gop == -1)
next_gop = 0;
}
/* for picture start frames (non-fragmented), we need to remove the first
* two 0x00 bytes and set P=1 */
if (!fragmented || found_gob) {
gst_adapter_flush (rtph263ppay->adapter, 2);
avail -= 2;
}
header_len = 2;
towrite = MIN (avail, gst_rtp_buffer_calc_payload_len
(GST_RTP_BASE_PAYLOAD_MTU (rtph263ppay) - header_len, 0, 0));
if (next_gop > 0)
towrite = MIN (next_gop, towrite);
outbuf = gst_rtp_buffer_new_allocate (header_len, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* last fragment gets the marker bit set */
gst_rtp_buffer_set_marker (&rtp, avail > towrite ? 0 : 1);
payload = gst_rtp_buffer_get_payload (&rtp);
/* 0 1
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | RR |P|V| PLEN |PEBIT|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
/* if fragmented or gop header , write p bit =1 */
payload[0] = (fragmented && !found_gob) ? 0x00 : 0x04;
payload[1] = 0;
GST_BUFFER_PTS (outbuf) = rtph263ppay->first_timestamp;
GST_BUFFER_DURATION (outbuf) = rtph263ppay->first_duration;
gst_rtp_buffer_unmap (&rtp);
payload_buf = gst_adapter_take_buffer_fast (rtph263ppay->adapter, towrite);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtph263ppay), outbuf, payload_buf,
g_quark_from_static_string (GST_META_TAG_VIDEO_STR));
outbuf = gst_buffer_append (outbuf, payload_buf);
avail -= towrite;
/* If more data is available and this is our first iteration,
* we create a buffer list and remember that we're fragmented.
*
* If we're fragmented already, add buffers to the previously
* created buffer list.
*
* Otherwise fragmented will be FALSE and we just push the single output
* buffer, and no list is allocated.
*/
if (avail && !fragmented) {
fragmented = TRUE;
list = gst_buffer_list_new ();
gst_buffer_list_add (list, outbuf);
} else if (fragmented) {
gst_buffer_list_add (list, outbuf);
}
}
if (fragmented) {
ret =
gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtph263ppay),
list);
} else {
ret =
gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtph263ppay), outbuf);
}
return ret;
}
static GstFlowReturn
speex_dec_chain_parse_data (GstSpeexDec * dec, GstBuffer * buf,
GstClockTime timestamp, GstClockTime duration)
{
GstFlowReturn res = GST_FLOW_OK;
gint i, fpp;
guint size;
guint8 *data;
SpeexBits *bits;
if (!dec->frame_duration)
goto not_negotiated;
if (timestamp != -1) {
dec->segment.last_stop = timestamp;
} else {
timestamp = dec->segment.last_stop;
}
if (buf) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* send data to the bitstream */
speex_bits_read_from (&dec->bits, (char *) data, size);
fpp = 0;
bits = &dec->bits;
GST_DEBUG_OBJECT (dec, "received buffer of size %u, fpp %d", size, fpp);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
fpp = dec->header->frames_per_packet;
bits = NULL;
}
/* now decode each frame, catering for unknown number of them (e.g. rtp) */
for (i = 0; (!fpp || i < fpp) && (!bits || speex_bits_remaining (bits) > 0);
i++) {
GstBuffer *outbuf;
gint16 *out_data;
gint ret;
GST_LOG_OBJECT (dec, "decoding frame %d/%d", i, fpp);
res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad,
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2,
GST_PAD_CAPS (dec->srcpad), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
ret = speex_decode_int (dec->state, bits, out_data);
if (ret == -1) {
/* uh? end of stream */
GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
gst_buffer_unref (outbuf);
outbuf = NULL;
break;
} else if (ret == -2) {
GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
gst_buffer_unref (outbuf);
outbuf = NULL;
break;
}
if (bits && speex_bits_remaining (bits) < 0) {
GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
gst_buffer_unref (outbuf);
outbuf = NULL;
break;
}
if (dec->header->nb_channels == 2)
speex_decode_stereo_int (out_data, dec->frame_size, dec->stereo);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = dec->frame_duration;
dec->segment.last_stop += dec->frame_duration;
timestamp = dec->segment.last_stop;
GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (dec->frame_duration));
res = gst_pad_push (dec->srcpad, outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
break;
}
}
return res;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
("decoder not initialized"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
static GstFlowReturn
gst_gdiscreencapsrc_create (GstPushSrc * push_src, GstBuffer ** buf)
{
GstGDIScreenCapSrc *src = GST_GDISCREENCAPSRC (push_src);
GstBuffer *new_buf;
gint new_buf_size;
GstClock *clock;
GstClockTime buf_time, buf_dur;
guint64 frame_number;
if (G_UNLIKELY (!src->info.bmiHeader.biWidth ||
!src->info.bmiHeader.biHeight)) {
GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before create function"));
return GST_FLOW_NOT_NEGOTIATED;
}
new_buf_size = GST_ROUND_UP_4 (src->info.bmiHeader.biWidth * 3) *
(-src->info.bmiHeader.biHeight);
GST_LOG_OBJECT (src,
"creating buffer of %d bytes with %dx%d image",
new_buf_size, (gint) src->info.bmiHeader.biWidth,
(gint) (-src->info.bmiHeader.biHeight));
new_buf = gst_buffer_new_and_alloc (new_buf_size);
clock = gst_element_get_clock (GST_ELEMENT (src));
if (clock != NULL) {
GstClockTime time, base_time;
/* Calculate sync time. */
time = gst_clock_get_time (clock);
base_time = gst_element_get_base_time (GST_ELEMENT (src));
buf_time = time - base_time;
if (src->rate_numerator) {
frame_number = gst_util_uint64_scale (buf_time,
src->rate_numerator, GST_SECOND * src->rate_denominator);
} else {
frame_number = -1;
}
} else {
buf_time = GST_CLOCK_TIME_NONE;
frame_number = -1;
}
if (frame_number != -1 && frame_number == src->frame_number) {
GstClockID id;
GstClockReturn ret;
/* Need to wait for the next frame */
frame_number += 1;
/* Figure out what the next frame time is */
buf_time = gst_util_uint64_scale (frame_number,
src->rate_denominator * GST_SECOND, src->rate_numerator);
id = gst_clock_new_single_shot_id (clock,
buf_time + gst_element_get_base_time (GST_ELEMENT (src)));
GST_OBJECT_LOCK (src);
src->clock_id = id;
GST_OBJECT_UNLOCK (src);
GST_DEBUG_OBJECT (src, "Waiting for next frame time %" G_GUINT64_FORMAT,
buf_time);
ret = gst_clock_id_wait (id, NULL);
GST_OBJECT_LOCK (src);
gst_clock_id_unref (id);
src->clock_id = NULL;
if (ret == GST_CLOCK_UNSCHEDULED) {
/* Got woken up by the unlock function */
GST_OBJECT_UNLOCK (src);
return GST_FLOW_FLUSHING;
}
GST_OBJECT_UNLOCK (src);
/* Duration is a complete 1/fps frame duration */
buf_dur =
gst_util_uint64_scale_int (GST_SECOND, src->rate_denominator,
src->rate_numerator);
} else if (frame_number != -1) {
GstClockTime next_buf_time;
GST_DEBUG_OBJECT (src, "No need to wait for next frame time %"
G_GUINT64_FORMAT " next frame = %" G_GINT64_FORMAT " prev = %"
G_GINT64_FORMAT, buf_time, frame_number, src->frame_number);
next_buf_time = gst_util_uint64_scale (frame_number + 1,
src->rate_denominator * GST_SECOND, src->rate_numerator);
/* Frame duration is from now until the next expected capture time */
buf_dur = next_buf_time - buf_time;
} else {
buf_dur = GST_CLOCK_TIME_NONE;
}
src->frame_number = frame_number;
GST_BUFFER_TIMESTAMP (new_buf) = buf_time;
GST_BUFFER_DURATION (new_buf) = buf_dur;
/* Do screen capture and put it into buffer... */
gst_gdiscreencapsrc_screen_capture (src, new_buf);
gst_object_unref (clock);
*buf = new_buf;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
GstFlowReturn ret;
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
* so not valid anymore */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
GST_DEBUG ("got %u raw samples", sample_count);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
/* create raw context */
enc->wp_context =
WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
(enc->correction_mode > 0) ? &enc->wvc_id : NULL);
if (!enc->wp_context) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("error creating Wavpack context"));
gst_object_unref (enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (enc);
/* set the configuration to the context now that we know everything
* and initialize the encoder */
if (!WavpackSetConfiguration (enc->wp_context,
enc->wp_config, (uint32_t) (-1))
|| !WavpackPackInit (enc->wp_context)) {
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
("error setting up wavpack encoding context"));
WavpackCloseFile (enc->wp_context);
gst_object_unref (enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
GST_DEBUG ("setup of encoding context successfull");
}
/* Save the timestamp of the first buffer. This will be later
* used as offset for all following buffers */
if (enc->timestamp_offset == GST_CLOCK_TIME_NONE) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
enc->timestamp_offset = GST_BUFFER_TIMESTAMP (buf);
enc->next_ts = GST_BUFFER_TIMESTAMP (buf);
} else {
enc->timestamp_offset = 0;
enc->next_ts = 0;
}
}
/* Check if we have a continous stream, if not drop some samples or the buffer or
* insert some silence samples */
if (enc->next_ts != GST_CLOCK_TIME_NONE &&
GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
guint64 diff_bytes;
GST_WARNING_OBJECT (enc, "Buffer is older than previous "
"timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
"), cannot handle. Clipping buffer.",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (enc->next_ts));
diff_bytes =
GST_CLOCK_TIME_TO_FRAMES (diff, enc->samplerate) * enc->channels * 2;
if (diff_bytes >= GST_BUFFER_SIZE (buf)) {
gst_buffer_unref (buf);
return GST_FLOW_OK;
}
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_DATA (buf) += diff_bytes;
GST_BUFFER_SIZE (buf) -= diff_bytes;
GST_BUFFER_TIMESTAMP (buf) += diff;
if (GST_BUFFER_DURATION_IS_VALID (buf))
GST_BUFFER_DURATION (buf) -= diff;
}
/* Allow a diff of at most 5 ms */
if (enc->next_ts != GST_CLOCK_TIME_NONE
&& GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > 5 * GST_MSECOND) {
GST_WARNING_OBJECT (enc,
"Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, 5 * GST_MSECOND);
WavpackFlushSamples (enc->wp_context);
enc->timestamp_offset += (GST_BUFFER_TIMESTAMP (buf) - enc->next_ts);
}
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
&& GST_BUFFER_DURATION_IS_VALID (buf))
enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
else
enc->next_ts = GST_CLOCK_TIME_NONE;
if (enc->need_channel_remap) {
buf = gst_buffer_make_writable (buf);
gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf),
sample_count);
}
/* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */
if (enc->md5) {
MD5Update (enc->md5_context, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
}
/* encode and handle return values from encoding */
if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
sample_count / enc->channels)) {
GST_DEBUG ("encoding samples successful");
ret = GST_FLOW_OK;
} else {
if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
(enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
ret = GST_FLOW_RESEND;
} else if ((enc->srcpad_last_return == GST_FLOW_OK) ||
(enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
ret = GST_FLOW_OK;
} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
(enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
ret = GST_FLOW_NOT_LINKED;
} else if ((enc->srcpad_last_return == GST_FLOW_WRONG_STATE) &&
(enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) {
ret = GST_FLOW_WRONG_STATE;
} else {
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
}
}
gst_buffer_unref (buf);
gst_object_unref (enc);
return ret;
}
static GstFlowReturn
gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)
{
GstFlowReturn ret;
GstBuffer *buf, *outbuf;
guint8 *payload, *spayload;
guint payload_len;
GstClockTime duration;
GstRTPBuffer rtp = { NULL, };
payload_len = rtpceltpay->bytes + rtpceltpay->sbytes;
duration = rtpceltpay->qduration;
GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (rtpceltpay->qduration));
/* get a big enough packet for the sizes + payloads */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
GST_BUFFER_DURATION (outbuf) = duration;
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* point to the payload for size headers and data */
spayload = gst_rtp_buffer_get_payload (&rtp);
payload = spayload + rtpceltpay->sbytes;
while ((buf = g_queue_pop_head (rtpceltpay->queue))) {
guint size;
/* copy first timestamp to output */
if (GST_BUFFER_PTS (outbuf) == -1)
GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buf);
/* write the size to the header */
size = gst_buffer_get_size (buf);
while (size > 0xff) {
*spayload++ = 0xff;
size -= 0xff;
}
*spayload++ = size;
/* copy payload */
size = gst_buffer_get_size (buf);
gst_buffer_extract (buf, 0, payload, size);
payload += size;
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpceltpay), outbuf, buf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
gst_buffer_unref (buf);
}
gst_rtp_buffer_unmap (&rtp);
/* we consumed it all */
rtpceltpay->bytes = 0;
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_mux_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstRTPMux *rtp_mux;
GstFlowReturn ret;
GstRTPMuxPadPrivate *padpriv;
gboolean drop;
gboolean changed = FALSE;
GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
rtp_mux = GST_RTP_MUX (parent);
if (gst_pad_check_reconfigure (rtp_mux->srcpad)) {
GstCaps *current_caps = gst_pad_get_current_caps (pad);
if (!gst_rtp_mux_setcaps (pad, rtp_mux, current_caps)) {
ret = GST_FLOW_NOT_NEGOTIATED;
gst_buffer_unref (buffer);
goto out;
}
gst_caps_unref (current_caps);
}
GST_OBJECT_LOCK (rtp_mux);
padpriv = gst_pad_get_element_private (pad);
if (!padpriv) {
GST_OBJECT_UNLOCK (rtp_mux);
gst_buffer_unref (buffer);
return GST_FLOW_NOT_LINKED;
}
buffer = gst_buffer_make_writable (buffer);
if (!gst_rtp_buffer_map (buffer, GST_MAP_READWRITE, &rtpbuffer)) {
GST_OBJECT_UNLOCK (rtp_mux);
gst_buffer_unref (buffer);
GST_ERROR_OBJECT (rtp_mux, "Invalid RTP buffer");
return GST_FLOW_ERROR;
}
drop = !process_buffer_locked (rtp_mux, padpriv, &rtpbuffer);
gst_rtp_buffer_unmap (&rtpbuffer);
if (!drop) {
if (pad != rtp_mux->last_pad) {
changed = TRUE;
g_clear_object (&rtp_mux->last_pad);
rtp_mux->last_pad = g_object_ref (pad);
}
if (GST_BUFFER_DURATION_IS_VALID (buffer) &&
GST_BUFFER_PTS_IS_VALID (buffer))
rtp_mux->last_stop = GST_BUFFER_PTS (buffer) +
GST_BUFFER_DURATION (buffer);
else
rtp_mux->last_stop = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (rtp_mux);
if (changed)
gst_pad_sticky_events_foreach (pad, resend_events, rtp_mux);
if (drop) {
gst_buffer_unref (buffer);
ret = GST_FLOW_OK;
} else {
ret = gst_pad_push (rtp_mux->srcpad, buffer);
}
out:
return ret;
}
bool GStreamerReader::DecodeVideoFrame(bool &aKeyFrameSkip,
int64_t aTimeThreshold)
{
NS_ASSERTION(mDecoder->OnDecodeThread(), "Should be on decode thread.");
GstBuffer *buffer = nullptr;
{
ReentrantMonitorAutoEnter mon(mGstThreadsMonitor);
if (mReachedVideoEos && !mVideoSinkBufferCount) {
return false;
}
/* Wait something to be decoded before return or continue */
if (!mVideoSinkBufferCount) {
if (!mAudioSinkBufferCount) {
/* We have nothing decoded so it makes no sense to return to the state machine
* as it will call us back immediately, we'll return again and so on, wasting
* CPU cycles for no job done. So, block here until there is either video or
* audio data available
*/
mon.Wait();
if (!mVideoSinkBufferCount) {
/* There is still no video data available, so either there is audio data or
* something else has happened (Eos, etc...). Return to the state machine
* to process it
*/
return true;
}
}
else {
return true;
}
}
mDecoder->NotifyDecodedFrames(0, 1);
#if GST_VERSION_MAJOR >= 1
GstSample *sample = gst_app_sink_pull_sample(mVideoAppSink);
buffer = gst_buffer_ref(gst_sample_get_buffer(sample));
gst_sample_unref(sample);
#else
buffer = gst_app_sink_pull_buffer(mVideoAppSink);
#endif
mVideoSinkBufferCount--;
}
bool isKeyframe = !GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT);
if ((aKeyFrameSkip && !isKeyframe)) {
gst_buffer_unref(buffer);
return true;
}
int64_t timestamp = GST_BUFFER_TIMESTAMP(buffer);
{
ReentrantMonitorAutoEnter mon(mGstThreadsMonitor);
timestamp = gst_segment_to_stream_time(&mVideoSegment,
GST_FORMAT_TIME, timestamp);
}
NS_ASSERTION(GST_CLOCK_TIME_IS_VALID(timestamp),
"frame has invalid timestamp");
timestamp = GST_TIME_AS_USECONDS(timestamp);
int64_t duration = 0;
if (GST_CLOCK_TIME_IS_VALID(GST_BUFFER_DURATION(buffer)))
duration = GST_TIME_AS_USECONDS(GST_BUFFER_DURATION(buffer));
else if (fpsNum && fpsDen)
/* add 1-frame duration */
duration = gst_util_uint64_scale(GST_USECOND, fpsDen, fpsNum);
if (timestamp < aTimeThreshold) {
LOG(PR_LOG_DEBUG, ("skipping frame %" GST_TIME_FORMAT
" threshold %" GST_TIME_FORMAT,
GST_TIME_ARGS(timestamp * 1000),
GST_TIME_ARGS(aTimeThreshold * 1000)));
gst_buffer_unref(buffer);
return true;
}
if (!buffer)
/* no more frames */
return true;
#if GST_VERSION_MAJOR >= 1
if (mConfigureAlignment && buffer->pool) {
GstStructure *config = gst_buffer_pool_get_config(buffer->pool);
GstVideoAlignment align;
if (gst_buffer_pool_config_get_video_alignment(config, &align))
gst_video_info_align(&mVideoInfo, &align);
gst_structure_free(config);
mConfigureAlignment = false;
}
#endif
nsRefPtr<PlanarYCbCrImage> image = GetImageFromBuffer(buffer);
if (!image) {
/* Ugh, upstream is not calling gst_pad_alloc_buffer(). Fallback to
* allocating a PlanarYCbCrImage backed GstBuffer here and memcpy.
*/
GstBuffer* tmp = nullptr;
CopyIntoImageBuffer(buffer, &tmp, image);
gst_buffer_unref(buffer);
buffer = tmp;
}
int64_t offset = mDecoder->GetResource()->Tell(); // Estimate location in media.
VideoData* video = VideoData::CreateFromImage(mInfo.mVideo,
mDecoder->GetImageContainer(),
offset, timestamp, duration,
static_cast<Image*>(image.get()),
isKeyframe, -1, mPicture);
mVideoQueue.Push(video);
gst_buffer_unref(buffer);
return true;
}
/* we expect buffers starting on startcodes.
*/
static GstFlowReturn
gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4VPay *rtpmp4vpay;
GstFlowReturn ret;
guint size, avail;
guint packet_len;
guint8 *data;
gboolean flush;
gint strip;
GstClockTime timestamp, duration;
ret = GST_FLOW_OK;
rtpmp4vpay = GST_RTP_MP4V_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
avail = gst_adapter_available (rtpmp4vpay->adapter);
if (duration == -1)
duration = 0;
/* empty buffer, take timestamp */
if (avail == 0) {
rtpmp4vpay->first_timestamp = timestamp;
rtpmp4vpay->duration = 0;
}
/* depay incomming data and see if we need to start a new RTP
* packet */
flush = gst_rtp_mp4v_pay_depay_data (rtpmp4vpay, data, size, &strip);
if (strip) {
/* strip off config if requested */
if (!rtpmp4vpay->send_config) {
GstBuffer *subbuf;
/* strip off header */
subbuf = gst_buffer_create_sub (buffer, strip, size - strip);
GST_BUFFER_TIMESTAMP (subbuf) = timestamp;
gst_buffer_unref (buffer);
buffer = subbuf;
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
}
}
/* if we need to flush, do so now */
if (flush) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_timestamp = timestamp;
rtpmp4vpay->duration = 0;
avail = 0;
}
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpmp4vpay->duration + duration)) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_timestamp = timestamp;
rtpmp4vpay->duration = 0;
}
/* push new data */
gst_adapter_push (rtpmp4vpay->adapter, buffer);
rtpmp4vpay->duration += duration;
return ret;
}
static GstFlowReturn
gst_audio_segment_clip_clip_buffer (GstSegmentClip * base, GstBuffer * buffer,
GstBuffer ** outbuf)
{
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
GstSegment *segment = &base->segment;
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
GstClockTime duration = GST_BUFFER_DURATION (buffer);
guint64 offset = GST_BUFFER_OFFSET (buffer);
guint64 offset_end = GST_BUFFER_OFFSET_END (buffer);
guint size = gst_buffer_get_size (buffer);
if (!self->rate || !self->framesize) {
GST_ERROR_OBJECT (self, "Not negotiated yet");
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
if (segment->format != GST_FORMAT_DEFAULT &&
segment->format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (self, "Unsupported segment format %s",
gst_format_get_name (segment->format));
*outbuf = buffer;
return GST_FLOW_OK;
}
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
GST_WARNING_OBJECT (self, "Buffer without valid timestamp");
*outbuf = buffer;
return GST_FLOW_OK;
}
*outbuf =
gst_audio_buffer_clip (buffer, segment, self->rate, self->framesize);
if (!*outbuf) {
GST_DEBUG_OBJECT (self, "Buffer outside the configured segment");
/* Now return unexpected if we're before/after the end */
if (segment->format == GST_FORMAT_TIME) {
if (segment->rate >= 0) {
if (segment->stop != -1 && timestamp >= segment->stop)
return GST_FLOW_EOS;
} else {
if (!GST_CLOCK_TIME_IS_VALID (duration))
duration =
gst_util_uint64_scale_int (size, GST_SECOND,
self->framesize * self->rate);
if (segment->start != -1 && timestamp + duration <= segment->start)
return GST_FLOW_EOS;
}
} else {
if (segment->rate >= 0) {
if (segment->stop != -1 && offset != -1 && offset >= segment->stop)
return GST_FLOW_EOS;
} else if (offset != -1 || offset_end != -1) {
if (offset_end == -1)
offset_end = offset + size / self->framesize;
if (segment->start != -1 && offset_end <= segment->start)
return GST_FLOW_EOS;
}
}
}
return GST_FLOW_OK;
}
static void
gst_musepackdec_loop (GstPad * sinkpad)
{
GstMusepackDec *musepackdec;
GstFlowReturn flow;
GstBuffer *out;
#ifdef MPC_IS_OLD_API
guint32 update_acc, update_bits;
#else
mpc_frame_info frame;
mpc_status err;
#endif
gint num_samples, samplerate, bitspersample;
musepackdec = GST_MUSEPACK_DEC (GST_PAD_PARENT (sinkpad));
samplerate = g_atomic_int_get (&musepackdec->rate);
if (samplerate == 0) {
if (!gst_musepack_stream_init (musepackdec))
goto pause_task;
gst_musepackdec_send_newsegment (musepackdec);
samplerate = g_atomic_int_get (&musepackdec->rate);
}
bitspersample = g_atomic_int_get (&musepackdec->bps);
flow = gst_pad_alloc_buffer_and_set_caps (musepackdec->srcpad, -1,
MPC_DECODER_BUFFER_LENGTH * 4, GST_PAD_CAPS (musepackdec->srcpad), &out);
if (flow != GST_FLOW_OK) {
GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow));
goto pause_task;
}
#ifdef MPC_IS_OLD_API
num_samples = mpc_decoder_decode (musepackdec->d,
(MPC_SAMPLE_FORMAT *) GST_BUFFER_DATA (out), &update_acc, &update_bits);
if (num_samples < 0) {
GST_ERROR_OBJECT (musepackdec, "Failed to decode sample");
GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL));
goto pause_task;
} else if (num_samples == 0) {
goto eos_and_pause;
}
#else
frame.buffer = (MPC_SAMPLE_FORMAT *) GST_BUFFER_DATA (out);
err = mpc_demux_decode (musepackdec->d, &frame);
if (err != MPC_STATUS_OK) {
GST_ERROR_OBJECT (musepackdec, "Failed to decode sample");
GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL));
goto pause_task;
} else if (frame.bits == -1) {
goto eos_and_pause;
}
num_samples = frame.samples;
#endif
GST_BUFFER_SIZE (out) = num_samples * bitspersample;
GST_BUFFER_OFFSET (out) = musepackdec->segment.last_stop;
GST_BUFFER_TIMESTAMP (out) =
gst_util_uint64_scale_int (musepackdec->segment.last_stop,
GST_SECOND, samplerate);
GST_BUFFER_DURATION (out) =
gst_util_uint64_scale_int (num_samples, GST_SECOND, samplerate);
musepackdec->segment.last_stop += num_samples;
GST_LOG_OBJECT (musepackdec, "Pushing buffer, timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)));
flow = gst_pad_push (musepackdec->srcpad, out);
if (flow != GST_FLOW_OK) {
GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow));
goto pause_task;
}
/* check if we're at the end of a configured segment */
if (musepackdec->segment.stop != -1 &&
musepackdec->segment.last_stop >= musepackdec->segment.stop) {
gint64 stop_time;
GST_DEBUG_OBJECT (musepackdec, "Reached end of configured segment");
if ((musepackdec->segment.flags & GST_SEEK_FLAG_SEGMENT) == 0)
goto eos_and_pause;
GST_DEBUG_OBJECT (musepackdec, "Posting SEGMENT_DONE message");
stop_time = gst_util_uint64_scale_int (musepackdec->segment.stop,
GST_SECOND, samplerate);
gst_element_post_message (GST_ELEMENT (musepackdec),
gst_message_new_segment_done (GST_OBJECT (musepackdec),
GST_FORMAT_TIME, stop_time));
goto pause_task;
}
return;
eos_and_pause:
{
GST_DEBUG_OBJECT (musepackdec, "sending EOS event");
gst_pad_push_event (musepackdec->srcpad, gst_event_new_eos ());
/* fall through to pause */
}
pause_task:
{
GST_DEBUG_OBJECT (musepackdec, "Pausing task");
gst_pad_pause_task (sinkpad);
return;
}
}
static GstFlowReturn
gst_ac3_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
GstAc3Parse *ac3parse = GST_AC3_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader reader;
gint off;
gboolean lost_sync, draining, eac, more = FALSE;
guint frmsiz, blocks, sid;
guint rate, chans;
gboolean update_rate = FALSE;
gint framesize = 0;
gint have_blocks = 0;
GstMapInfo map;
gboolean ret = FALSE;
GstFlowReturn res = GST_FLOW_OK;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < 8)) {
*skipsize = 1;
goto cleanup;
}
gst_byte_reader_init (&reader, map.data, map.size);
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffff0000, 0x0b770000,
0, map.size);
GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
/* didn't find anything that looks like a sync word, skip */
if (off < 0) {
*skipsize = map.size - 3;
goto cleanup;
}
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
goto cleanup;
}
/* make sure the values in the frame header look sane */
if (!gst_ac3_parse_frame_header (ac3parse, buf, 0, &frmsiz, &rate, &chans,
&blocks, &sid, &eac)) {
*skipsize = off + 2;
goto cleanup;
}
GST_LOG_OBJECT (parse, "size: %u, blocks: %u, rate: %u, chans: %u", frmsiz,
blocks, rate, chans);
framesize = frmsiz;
if (G_UNLIKELY (g_atomic_int_get (&ac3parse->align) ==
GST_AC3_PARSE_ALIGN_NONE))
gst_ac3_parse_set_alignment (ac3parse, eac);
GST_LOG_OBJECT (parse, "got frame");
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
draining = GST_BASE_PARSE_DRAINING (parse);
if (g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_IEC61937) {
/* We need 6 audio blocks from each substream, so we keep going forwards
* till we have it */
g_assert (blocks > 0);
GST_LOG_OBJECT (ac3parse, "Need %d frames before pushing", 6 / blocks);
if (sid != 0) {
/* We need the first substream to be the one with id 0 */
GST_LOG_OBJECT (ac3parse, "Skipping till we find sid 0");
*skipsize = off + 2;
goto cleanup;
}
framesize = 0;
/* Loop till we have 6 blocks per substream */
for (have_blocks = 0; !more && have_blocks < 6; have_blocks += blocks) {
/* Loop till we get one frame from each substream */
do {
framesize += frmsiz;
if (!gst_byte_reader_skip (&reader, frmsiz)
|| map.size < (framesize + 6)) {
more = TRUE;
break;
}
if (!gst_ac3_parse_frame_header (ac3parse, buf, framesize, &frmsiz,
NULL, NULL, NULL, &sid, &eac)) {
*skipsize = off + 2;
goto cleanup;
}
} while (sid);
}
/* We're now at the next frame, so no need to skip if resyncing */
frmsiz = 0;
}
if (lost_sync && !draining) {
guint16 word = 0;
GST_DEBUG_OBJECT (ac3parse, "resyncing; checking next frame syncword");
if (more || !gst_byte_reader_skip (&reader, frmsiz) ||
!gst_byte_reader_get_uint16_be (&reader, &word)) {
GST_DEBUG_OBJECT (ac3parse, "... but not sufficient data");
gst_base_parse_set_min_frame_size (parse, framesize + 8);
*skipsize = 0;
goto cleanup;
} else {
if (word != 0x0b77) {
GST_DEBUG_OBJECT (ac3parse, "0x%x not OK", word);
*skipsize = off + 2;
goto cleanup;
} else {
/* ok, got sync now, let's assume constant frame size */
gst_base_parse_set_min_frame_size (parse, framesize);
}
}
}
/* expect to have found a frame here */
g_assert (framesize);
ret = TRUE;
/* arrange for metadata setup */
if (G_UNLIKELY (sid)) {
/* dependent frame, no need to (ac)count for or consider further */
GST_LOG_OBJECT (parse, "sid: %d", sid);
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_NO_FRAME;
/* TODO maybe also mark as DELTA_UNIT,
* if that does not surprise baseparse elsewhere */
/* occupies same time space as previous base frame */
if (G_LIKELY (GST_BUFFER_TIMESTAMP (buf) >= GST_BUFFER_DURATION (buf)))
GST_BUFFER_TIMESTAMP (buf) -= GST_BUFFER_DURATION (buf);
/* only shortcut if we already arranged for caps */
if (G_LIKELY (ac3parse->sample_rate > 0))
goto cleanup;
}
if (G_UNLIKELY (ac3parse->sample_rate != rate || ac3parse->channels != chans
|| ac3parse->eac != eac)) {
GstCaps *caps = gst_caps_new_simple (eac ? "audio/x-eac3" : "audio/x-ac3",
"framed", G_TYPE_BOOLEAN, TRUE, "rate", G_TYPE_INT, rate,
"channels", G_TYPE_INT, chans, NULL);
gst_caps_set_simple (caps, "alignment", G_TYPE_STRING,
g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_IEC61937 ?
"iec61937" : "frame", NULL);
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
ac3parse->sample_rate = rate;
ac3parse->channels = chans;
ac3parse->eac = eac;
update_rate = TRUE;
}
if (G_UNLIKELY (ac3parse->blocks != blocks)) {
ac3parse->blocks = blocks;
update_rate = TRUE;
}
if (G_UNLIKELY (update_rate))
gst_base_parse_set_frame_rate (parse, rate, 256 * blocks, 2, 2);
cleanup:
gst_buffer_unmap (buf, &map);
if (ret && framesize <= map.size) {
res = gst_base_parse_finish_frame (parse, frame, framesize);
}
return res;
}
GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad*,
GstPadProbeInfo* info,
gpointer self) {
GstEnginePipeline* instance = reinterpret_cast<GstEnginePipeline*>(self);
GstBuffer* buf = gst_pad_probe_info_get_buffer(info);
QList<BufferConsumer*> consumers;
{
QMutexLocker l(&instance->buffer_consumers_mutex_);
consumers = instance->buffer_consumers_;
}
for (BufferConsumer* consumer : consumers) {
gst_buffer_ref(buf);
consumer->ConsumeBuffer(buf, instance->id());
}
// Calculate the end time of this buffer so we can stop playback if it's
// after the end time of this song.
if (instance->end_offset_nanosec_ > 0) {
quint64 start_time = GST_BUFFER_TIMESTAMP(buf) - instance->segment_start_;
quint64 duration = GST_BUFFER_DURATION(buf);
quint64 end_time = start_time + duration;
if (end_time > instance->end_offset_nanosec_) {
if (instance->has_next_valid_url()) {
if (instance->next_url_ == instance->url_ &&
instance->next_beginning_offset_nanosec_ ==
instance->end_offset_nanosec_) {
// The "next" song is actually the next segment of this file - so
// cheat and keep on playing, but just tell the Engine we've moved on.
instance->end_offset_nanosec_ = instance->next_end_offset_nanosec_;
instance->next_url_ = QUrl();
instance->next_beginning_offset_nanosec_ = 0;
instance->next_end_offset_nanosec_ = 0;
// GstEngine will try to seek to the start of the new section, but
// we're already there so ignore it.
instance->ignore_next_seek_ = true;
emit instance->EndOfStreamReached(instance->id(), true);
} else {
// We have a next song but we can't cheat, so move to it normally.
instance->TransitionToNext();
}
} else {
// There's no next song
emit instance->EndOfStreamReached(instance->id(), false);
}
}
}
if (instance->emit_track_ended_on_time_discontinuity_) {
if (GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLAG_DISCONT) ||
GST_BUFFER_OFFSET(buf) < instance->last_buffer_offset_) {
qLog(Debug) << "Buffer discontinuity - emitting EOS";
instance->emit_track_ended_on_time_discontinuity_ = false;
emit instance->EndOfStreamReached(instance->id(), true);
}
}
instance->last_buffer_offset_ = GST_BUFFER_OFFSET(buf);
return GST_PAD_PROBE_OK;
}
static GstFlowReturn
gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpAMRPay *rtpamrpay;
GstFlowReturn ret;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data, *payload_amr;
GstClockTime timestamp, duration;
guint packet_len, mtu;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
gint *frame_size;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* setup frame size pointer */
if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
frame_size = nb_frame_size;
else
frame_size = wb_frame_size;
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
/* FIXME, only
* octet aligned, no interleaving, single channel, no CRC,
* no robust-sorting. To fix this you need to implement the downstream
* negotiation function. */
/* first count number of packets and total amr frame size */
amr_len = num_packets = num_nonempty_packets = 0;
for (i = 0; i < size; i++) {
guint8 FT;
gint fr_size;
FT = (data[i] & 0x78) >> 3;
fr_size = frame_size[FT];
GST_DEBUG_OBJECT (basepayload, "frame size %d", fr_size);
/* FIXME, we don't handle this yet.. */
if (fr_size <= 0)
goto wrong_size;
amr_len += fr_size;
num_nonempty_packets++;
num_packets++;
i += fr_size;
}
if (amr_len > size)
goto incomplete_frame;
/* we need one extra byte for the CMR, the ToC is in the input
* data */
payload_len = size + 1;
/* get packet len to check against MTU */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
if (packet_len > mtu)
goto too_big;
/* now alloc output buffer */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* FIXME: when we do more than one AMR frame per packet, fix this */
if (duration != GST_CLOCK_TIME_NONE)
GST_BUFFER_DURATION (outbuf) = duration;
else {
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
}
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
gst_rtp_buffer_set_marker (outbuf, TRUE);
}
/* get payload, this is now writable */
payload = gst_rtp_buffer_get_payload (outbuf);
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* | CMR |R|R|R|R|
* +-+-+-+-+-+-+-+-+
*/
payload[0] = 0xF0; /* CMR, no specific mode requested */
/* this is where we copy the AMR data, after num_packets FTs and the
* CMR. */
payload_amr = payload + num_packets + 1;
/* copy data in payload, first we copy all the FTs then all
* the AMR data. The last FT has to have the F flag cleared. */
for (i = 1; i <= num_packets; i++) {
guint8 FT;
gint fr_size;
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |F| FT |Q|P|P| more FT...
* +-+-+-+-+-+-+-+-+
*/
FT = (*data & 0x78) >> 3;
fr_size = frame_size[FT];
if (i == num_packets)
/* last packet, clear F flag */
payload[i] = *data & 0x7f;
else
/* set F flag */
payload[i] = *data | 0x80;
memcpy (payload_amr, &data[1], fr_size);
/* all sizes are > 0 since we checked for that above */
data += fr_size + 1;
payload_amr += fr_size;
}
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
/* ERRORS */
wrong_size:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received AMR frame with size <= 0"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
incomplete_frame:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received incomplete AMR frames"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
too_big:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received too many AMR frames for MTU"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstAmcAudioDec *self;
gint idx;
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
guint offset = 0;
GstClockTime timestamp, duration, timestamp_offset = 0;
GstMapInfo minfo;
memset (&minfo, 0, sizeof (minfo));
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Handling frame");
/* Make sure to keep a reference to the input here,
* it can be unreffed from the other thread if
* finish_frame() is called */
if (inbuf)
inbuf = gst_buffer_ref (inbuf);
if (!self->started) {
GST_ERROR_OBJECT (self, "Codec not started yet");
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_NOT_NEGOTIATED;
}
if (self->eos) {
GST_WARNING_OBJECT (self, "Got frame after EOS");
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_EOS;
}
if (self->flushing)
goto flushing;
if (self->downstream_flow_ret != GST_FLOW_OK)
goto downstream_error;
if (!inbuf)
return gst_amc_audio_dec_drain (self);
timestamp = GST_BUFFER_PTS (inbuf);
duration = GST_BUFFER_DURATION (inbuf);
gst_buffer_map (inbuf, &minfo, GST_MAP_READ);
while (offset < minfo.size) {
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx = gst_amc_codec_dequeue_input_buffer (self->codec, 100000);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (idx < 0) {
if (self->flushing)
goto flushing;
switch (idx) {
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing input buffer timed out");
continue; /* next try */
break;
case G_MININT:
GST_ERROR_OBJECT (self, "Failed to dequeue input buffer");
goto dequeue_error;
default:
g_assert_not_reached ();
break;
}
continue;
}
if (idx >= self->n_input_buffers)
goto invalid_buffer_index;
if (self->flushing)
goto flushing;
if (self->downstream_flow_ret != GST_FLOW_OK) {
memset (&buffer_info, 0, sizeof (buffer_info));
gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info);
goto downstream_error;
}
/* Now handle the frame */
/* Copy the buffer content in chunks of size as requested
* by the port */
buf = &self->input_buffers[idx];
memset (&buffer_info, 0, sizeof (buffer_info));
buffer_info.offset = 0;
buffer_info.size = MIN (minfo.size - offset, buf->size);
orc_memcpy (buf->data, minfo.data + offset, buffer_info.size);
/* Interpolate timestamps if we're passing the buffer
* in multiple chunks */
if (offset != 0 && duration != GST_CLOCK_TIME_NONE) {
timestamp_offset = gst_util_uint64_scale (offset, duration, minfo.size);
}
if (timestamp != GST_CLOCK_TIME_NONE) {
buffer_info.presentation_time_us =
gst_util_uint64_scale (timestamp + timestamp_offset, 1, GST_USECOND);
self->last_upstream_ts = timestamp + timestamp_offset;
}
if (duration != GST_CLOCK_TIME_NONE)
self->last_upstream_ts += duration;
if (offset == 0) {
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DELTA_UNIT))
buffer_info.flags |= BUFFER_FLAG_SYNC_FRAME;
}
offset += buffer_info.size;
GST_DEBUG_OBJECT (self,
"Queueing buffer %d: size %d time %" G_GINT64_FORMAT " flags 0x%08x",
idx, buffer_info.size, buffer_info.presentation_time_us,
buffer_info.flags);
if (!gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info))
goto queue_error;
}
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
downstream_error:
{
GST_ERROR_OBJECT (self, "Downstream returned %s",
gst_flow_get_name (self->downstream_flow_ret));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
}
invalid_buffer_index:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Invalid input buffer index %d of %d", idx, self->n_input_buffers));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
dequeue_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to dequeue input buffer"));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
queue_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to queue input buffer"));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_FLUSHING;
}
}
static int
gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
{
GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id;
GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc);
GstFlowReturn *flow;
GstBuffer *buffer;
GstPad *pad;
guchar *block = (guchar *) data;
pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
flow =
(wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
srcpad_last_return;
*flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
count, GST_PAD_CAPS (pad), &buffer);
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
return FALSE;
}
g_memmove (GST_BUFFER_DATA (buffer), block, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
WavpackHeader wph;
GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata",
count, (wid->correction) ? "correction " : "");
gst_wavpack_read_header (&wph, block);
/* Only set when pushing the first buffer again, in that case
* we don't want to delay the buffer or push newsegment events
*/
if (!wid->passthrough) {
/* Only push complete blocks */
if (enc->pending_buffer == NULL) {
enc->pending_buffer = buffer;
enc->pending_offset = wph.block_index;
} else if (enc->pending_offset == wph.block_index) {
enc->pending_buffer = gst_buffer_join (enc->pending_buffer, buffer);
} else {
GST_ERROR ("Got incomplete block, dropping");
gst_buffer_unref (enc->pending_buffer);
enc->pending_buffer = buffer;
enc->pending_offset = wph.block_index;
}
if (!(wph.flags & FINAL_BLOCK))
return TRUE;
buffer = enc->pending_buffer;
enc->pending_buffer = NULL;
enc->pending_offset = 0;
/* if it's the first wavpack block, send a NEW_SEGMENT event */
if (wph.block_index == 0) {
gst_pad_push_event (pad,
gst_event_new_new_segment (FALSE,
1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
/* save header for later reference, so we can re-send it later on
* EOS with fixed up values for total sample count etc. */
if (enc->first_block == NULL && !wid->correction) {
enc->first_block =
g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
enc->first_block_size = GST_BUFFER_SIZE (buffer);
}
}
}
/* set buffer timestamp, duration, offset, offset_end from
* the wavpack header */
GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset +
gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
enc->samplerate);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
enc->samplerate);
GST_BUFFER_OFFSET (buffer) = wph.block_index;
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
} else {
/* if it's something else set no timestamp and duration on the buffer */
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
}
/* push the buffer and forward errors */
GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
GST_BUFFER_SIZE (buffer));
*flow = gst_pad_push (pad, buffer);
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
return FALSE;
}
return TRUE;
}
static GstFlowReturn
pad_chain (GstPad *pad,
GstBuffer *buf)
{
GOmxCore *gomx;
GOmxPort *in_port;
GstOmxBaseFilter *self;
GstFlowReturn ret = GST_FLOW_OK;
self = GST_OMX_BASE_FILTER (GST_OBJECT_PARENT (pad));
gomx = self->gomx;
GST_LOG_OBJECT (self, "begin");
GST_LOG_OBJECT (self, "gst_buffer: size=%u", GST_BUFFER_SIZE (buf));
GST_LOG_OBJECT (self, "state: %d", gomx->omx_state);
if (G_UNLIKELY (gomx->omx_state == OMX_StateLoaded))
{
g_mutex_lock (self->ready_lock);
GST_INFO_OBJECT (self, "omx: prepare");
/** @todo this should probably go after doing preparations. */
if (self->omx_setup)
{
self->omx_setup (self);
}
setup_ports (self);
g_omx_core_prepare (self->gomx);
if (gomx->omx_state == OMX_StateIdle)
{
self->ready = TRUE;
gst_pad_start_task (self->srcpad, output_loop, self->srcpad);
}
g_mutex_unlock (self->ready_lock);
if (gomx->omx_state != OMX_StateIdle)
goto out_flushing;
}
in_port = self->in_port;
if (G_LIKELY (in_port->enabled))
{
guint buffer_offset = 0;
if (G_UNLIKELY (gomx->omx_state == OMX_StateIdle))
{
GST_INFO_OBJECT (self, "omx: play");
g_omx_core_start (gomx);
if (gomx->omx_state != OMX_StateExecuting)
goto out_flushing;
/* send buffer with codec data flag */
/** @todo move to util */
if (self->codec_data)
{
OMX_BUFFERHEADERTYPE *omx_buffer;
GST_LOG_OBJECT (self, "request buffer");
omx_buffer = g_omx_port_request_buffer (in_port);
if (G_LIKELY (omx_buffer))
{
omx_buffer->nFlags |= 0x00000080; /* codec data flag */
omx_buffer->nFilledLen = GST_BUFFER_SIZE (self->codec_data);
memcpy (omx_buffer->pBuffer + omx_buffer->nOffset, GST_BUFFER_DATA (self->codec_data), omx_buffer->nFilledLen);
GST_LOG_OBJECT (self, "release_buffer");
g_omx_port_release_buffer (in_port, omx_buffer);
}
}
}
if (G_UNLIKELY (gomx->omx_state != OMX_StateExecuting))
{
GST_ERROR_OBJECT (self, "Whoa! very wrong");
}
while (G_LIKELY (buffer_offset < GST_BUFFER_SIZE (buf)))
{
OMX_BUFFERHEADERTYPE *omx_buffer;
if (self->last_pad_push_return != GST_FLOW_OK ||
!(gomx->omx_state == OMX_StateExecuting ||
gomx->omx_state == OMX_StatePause))
{
goto out_flushing;
}
GST_LOG_OBJECT (self, "request buffer");
omx_buffer = g_omx_port_request_buffer (in_port);
GST_LOG_OBJECT (self, "omx_buffer: %p", omx_buffer);
if (G_LIKELY (omx_buffer))
{
GST_DEBUG_OBJECT (self, "omx_buffer: size=%lu, len=%lu, flags=%lu, offset=%lu, timestamp=%lld",
omx_buffer->nAllocLen, omx_buffer->nFilledLen, omx_buffer->nFlags,
omx_buffer->nOffset, omx_buffer->nTimeStamp);
if (omx_buffer->nOffset == 0 &&
self->share_input_buffer)
{
{
GstBuffer *old_buf;
old_buf = omx_buffer->pAppPrivate;
if (old_buf)
{
gst_buffer_unref (old_buf);
}
else if (omx_buffer->pBuffer)
{
g_free (omx_buffer->pBuffer);
}
}
omx_buffer->pBuffer = GST_BUFFER_DATA (buf);
omx_buffer->nAllocLen = GST_BUFFER_SIZE (buf);
omx_buffer->nFilledLen = GST_BUFFER_SIZE (buf);
omx_buffer->pAppPrivate = buf;
}
else
{
omx_buffer->nFilledLen = MIN (GST_BUFFER_SIZE (buf) - buffer_offset,
omx_buffer->nAllocLen - omx_buffer->nOffset);
memcpy (omx_buffer->pBuffer + omx_buffer->nOffset, GST_BUFFER_DATA (buf) + buffer_offset, omx_buffer->nFilledLen);
}
if (self->use_timestamps)
{
GstClockTime timestamp_offset = 0;
if (buffer_offset && GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE)
{
timestamp_offset = gst_util_uint64_scale_int (buffer_offset,
GST_BUFFER_DURATION (buf),
GST_BUFFER_SIZE (buf));
}
omx_buffer->nTimeStamp = gst_util_uint64_scale_int (GST_BUFFER_TIMESTAMP (buf) + timestamp_offset,
OMX_TICKS_PER_SECOND,
GST_SECOND);
}
buffer_offset += omx_buffer->nFilledLen;
GST_LOG_OBJECT (self, "release_buffer");
/** @todo untaint buffer */
g_omx_port_release_buffer (in_port, omx_buffer);
}
else
{
GST_WARNING_OBJECT (self, "null buffer");
ret = GST_FLOW_WRONG_STATE;
goto out_flushing;
}
}
}
else
{
GST_WARNING_OBJECT (self, "done");
ret = GST_FLOW_UNEXPECTED;
}
if (!self->share_input_buffer)
{
gst_buffer_unref (buf);
}
leave:
GST_LOG_OBJECT (self, "end");
return ret;
/* special conditions */
out_flushing:
{
const gchar *error_msg = NULL;
if (gomx->omx_error)
{
error_msg = "Error from OpenMAX component";
}
else if (gomx->omx_state != OMX_StateExecuting &&
gomx->omx_state != OMX_StatePause)
{
error_msg = "OpenMAX component in wrong state";
}
if (error_msg)
{
GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL), (error_msg));
ret = GST_FLOW_ERROR;
}
gst_buffer_unref (buf);
goto leave;
}
}
/* Pipeline Callbacks */
static gboolean
probe_cb (InsanityGstTest * ptest, GstPad * pad, GstMiniObject * object,
gpointer userdata)
{
InsanityTest *test = INSANITY_TEST (ptest);
global_last_probe = g_get_monotonic_time ();
DECODER_TEST_LOCK ();
if (GST_IS_BUFFER (object)) {
GstBuffer *buf;
GstClockTime ts;
buf = GST_BUFFER (object);
ts = GST_BUFFER_PTS (buf);
/* First check clipping */
if (glob_testing_parser == FALSE && GST_CLOCK_TIME_IS_VALID (ts) &&
glob_waiting_segment == FALSE) {
GstClockTime ts_end, cstart, cstop;
/* Check if buffer is completely outside the segment */
ts_end = ts;
if (GST_BUFFER_DURATION_IS_VALID (buf))
ts_end += GST_BUFFER_DURATION (buf);
/* Check if buffer is completely outside the segment */
ts_end = ts;
if (!gst_segment_clip (&glob_last_segment,
glob_last_segment.format, ts, ts_end, &cstart, &cstop)) {
char *msg = g_strdup_printf ("Got timestamp %" GST_TIME_FORMAT " -- %"
GST_TIME_FORMAT ", outside configured segment (%" GST_TIME_FORMAT
" -- %" GST_TIME_FORMAT "), method %s",
GST_TIME_ARGS (ts), GST_TIME_ARGS (ts_end),
GST_TIME_ARGS (glob_last_segment.start),
GST_TIME_ARGS (glob_last_segment.stop),
test_get_name (glob_in_progress));
insanity_test_validate_checklist_item (INSANITY_TEST (ptest),
"segment-clipping", FALSE, msg);
g_free (msg);
glob_bad_segment_clipping = TRUE;
}
}
switch (glob_in_progress) {
case TEST_NONE:
if (glob_waiting_first_segment == TRUE)
insanity_test_validate_checklist_item (test, "first-segment",
FALSE, "Got a buffer before the first segment");
/* Got the first buffer, starting testing dance */
next_test (test);
break;
case TEST_POSITION:
test_position (test, buf);
break;
case TEST_FAST_FORWARD:
case TEST_BACKWARD_PLAYBACK:
case TEST_FAST_BACKWARD:
{
gint64 stime_ts;
if (GST_CLOCK_TIME_IS_VALID (ts) == FALSE ||
glob_waiting_segment == TRUE) {
break;
}
stime_ts = gst_segment_to_stream_time (&glob_last_segment,
glob_last_segment.format, ts);
if (GST_CLOCK_TIME_IS_VALID (glob_seek_first_buf_ts) == FALSE) {
GstClockTime expected_ts =
gst_segment_to_stream_time (&glob_last_segment,
glob_last_segment.format,
glob_seek_rate <
0 ? glob_seek_stop_ts : glob_seek_segment_seektime);
GstClockTimeDiff diff = ABS (GST_CLOCK_DIFF (stime_ts, expected_ts));
if (diff > SEEK_THRESHOLD) {
gchar *valmsg =
g_strdup_printf ("Received buffer timestamp %" GST_TIME_FORMAT
" Seeek wanted %" GST_TIME_FORMAT "",
GST_TIME_ARGS (stime_ts),
GST_TIME_ARGS (expected_ts));
validate_current_test (test, FALSE, valmsg);
next_test (test);
g_free (valmsg);
} else
glob_seek_first_buf_ts = stime_ts;
} else {
GstClockTimeDiff diff =
GST_CLOCK_DIFF (stime_ts, glob_seek_first_buf_ts);
if (diff < 0)
diff = -diff;
if (diff >= glob_playback_duration * GST_SECOND) {
validate_current_test (test, TRUE, NULL);
next_test (test);
}
}
break;
}
default:
break;
}
} else if (GST_IS_EVENT (object)) {
GstEvent *event = GST_EVENT (object);
guint seqnum = gst_event_get_seqnum (event);
if (G_LIKELY (glob_seqnum_found == FALSE) && seqnum == glob_seqnum)
glob_seqnum_found = TRUE;
if (glob_seqnum_found == TRUE && seqnum != glob_seqnum) {
gchar *message = g_strdup_printf ("Current seqnum %i != "
"received %i", glob_seqnum, seqnum);
insanity_test_validate_checklist_item (test, "seqnum-management",
FALSE, message);
glob_wrong_seqnum = TRUE;
g_free (message);
}
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
{
gst_event_copy_segment (event, &glob_last_segment);
if (glob_waiting_segment == FALSE)
/* Cache the segment as it will be our reference but don't look
* further */
goto done;
glob_last_segment_start_time = glob_last_segment.start;
if (glob_waiting_first_segment == TRUE) {
insanity_test_validate_checklist_item (test, "first-segment", TRUE,
NULL);
glob_waiting_first_segment = FALSE;
} else if (glob_in_progress >= TEST_FAST_FORWARD &&
glob_in_progress <= TEST_FAST_BACKWARD) {
GstClockTimeDiff diff;
gboolean valid_stop = TRUE;
GstClockTimeDiff wdiff, rdiff;
rdiff =
ABS (GST_CLOCK_DIFF (glob_last_segment.stop,
glob_last_segment.start)) * ABS (glob_last_segment.rate *
glob_last_segment.applied_rate);
wdiff =
ABS (GST_CLOCK_DIFF (glob_seek_stop_ts,
glob_seek_segment_seektime));
diff =
GST_CLOCK_DIFF (glob_last_segment.position,
glob_seek_segment_seektime);
if (diff < 0)
diff = -diff;
/* Now compare with the expected segment */
if ((glob_last_segment.rate * glob_last_segment.applied_rate) ==
glob_seek_rate && diff <= SEEK_THRESHOLD && valid_stop) {
glob_seek_got_segment = TRUE;
} else {
GstClockTime stopdiff = ABS (GST_CLOCK_DIFF (rdiff, wdiff));
gchar *validate_msg =
g_strdup_printf ("Wrong segment received, Rate %f expected "
"%f, start time diff %" GST_TIME_FORMAT " stop diff %"
GST_TIME_FORMAT,
(glob_last_segment.rate * glob_last_segment.applied_rate),
glob_seek_rate,
GST_TIME_ARGS (diff), GST_TIME_ARGS (stopdiff));
validate_current_test (test, FALSE, validate_msg);
next_test (test);
g_free (validate_msg);
}
}
glob_waiting_segment = FALSE;
break;
}
default:
break;
}
}
done:
DECODER_TEST_UNLOCK ();
return TRUE;
}
static GstFlowReturn
gst_vaapidecode_step(GstVaapiDecode *decode)
{
GstVaapiSurfaceProxy *proxy;
GstVaapiDecoderStatus status;
GstBuffer *buffer;
GstFlowReturn ret;
GstClockTime timestamp;
gint64 end_time;
for (;;) {
end_time = decode->render_time_base;
if (!end_time)
end_time = g_get_monotonic_time();
end_time += GST_TIME_AS_USECONDS(decode->last_buffer_time);
end_time += G_TIME_SPAN_SECOND;
proxy = gst_vaapi_decoder_get_surface(decode->decoder, &status);
if (!proxy) {
if (status == GST_VAAPI_DECODER_STATUS_ERROR_NO_SURFACE) {
gboolean was_signalled;
g_mutex_lock(decode->decoder_mutex);
was_signalled = g_cond_wait_until(
decode->decoder_ready,
decode->decoder_mutex,
end_time
);
g_mutex_unlock(decode->decoder_mutex);
if (was_signalled)
continue;
goto error_decode_timeout;
}
if (status != GST_VAAPI_DECODER_STATUS_ERROR_NO_DATA)
goto error_decode;
/* More data is needed */
break;
}
g_object_weak_ref(
G_OBJECT(proxy),
(GWeakNotify)gst_vaapidecode_release,
decode
);
buffer = gst_vaapi_video_buffer_new(decode->display);
if (!buffer)
goto error_create_buffer;
timestamp = GST_VAAPI_SURFACE_PROXY_TIMESTAMP(proxy);
if (!decode->render_time_base)
decode->render_time_base = g_get_monotonic_time();
decode->last_buffer_time = timestamp;
GST_BUFFER_TIMESTAMP(buffer) = timestamp;
GST_BUFFER_DURATION(buffer) = GST_VAAPI_SURFACE_PROXY_DURATION(proxy);
gst_buffer_set_caps(buffer, GST_PAD_CAPS(decode->srcpad));
if (GST_VAAPI_SURFACE_PROXY_TFF(proxy))
GST_BUFFER_FLAG_SET(buffer, GST_VIDEO_BUFFER_TFF);
gst_vaapi_video_buffer_set_surface_proxy(
GST_VAAPI_VIDEO_BUFFER(buffer),
proxy
);
ret = gst_pad_push(decode->srcpad, buffer);
if (ret != GST_FLOW_OK)
goto error_commit_buffer;
g_object_unref(proxy);
}
return GST_FLOW_OK;
/* ERRORS */
error_decode_timeout:
{
GST_DEBUG("decode timeout. Decoder required a VA surface but none "
"got available within one second");
return GST_FLOW_UNEXPECTED;
}
error_decode:
{
GST_DEBUG("decode error %d", status);
switch (status) {
case GST_VAAPI_DECODER_STATUS_ERROR_UNSUPPORTED_CODEC:
case GST_VAAPI_DECODER_STATUS_ERROR_UNSUPPORTED_PROFILE:
case GST_VAAPI_DECODER_STATUS_ERROR_UNSUPPORTED_CHROMA_FORMAT:
ret = GST_FLOW_NOT_SUPPORTED;
break;
default:
ret = GST_FLOW_UNEXPECTED;
break;
}
return ret;
}
error_create_buffer:
{
const GstVaapiID surface_id =
gst_vaapi_surface_get_id(GST_VAAPI_SURFACE_PROXY_SURFACE(proxy));
GST_DEBUG("video sink failed to create video buffer for proxy'ed "
"surface %" GST_VAAPI_ID_FORMAT " (error %d)",
GST_VAAPI_ID_ARGS(surface_id), ret);
g_object_unref(proxy);
return GST_FLOW_UNEXPECTED;
}
error_commit_buffer:
{
GST_DEBUG("video sink rejected the video buffer (error %d)", ret);
g_object_unref(proxy);
return GST_FLOW_UNEXPECTED;
}
}
static GstFlowReturn
gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstFlowReturn ret;
GstRtpCELTPay *rtpceltpay;
gsize payload_len;
GstMapInfo map;
GstClockTime duration, packet_dur;
guint i, ssize, packet_len;
rtpceltpay = GST_RTP_CELT_PAY (basepayload);
ret = GST_FLOW_OK;
gst_buffer_map (buffer, &map, GST_MAP_READ);
switch (rtpceltpay->packet) {
case 0:
/* ident packet. We need to parse the headers to construct the RTP
* properties. */
if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, map.data, map.size))
goto parse_error;
goto cleanup;
case 1:
/* comment packet, we ignore it */
goto cleanup;
default:
/* other packets go in the payload */
break;
}
gst_buffer_unmap (buffer, &map);
duration = GST_BUFFER_DURATION (buffer);
GST_LOG_OBJECT (rtpceltpay,
"got buffer of duration %" GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT,
GST_TIME_ARGS (duration), map.size);
/* calculate the size of the size field and the payload */
ssize = 1;
for (i = map.size; i > 0xff; i -= 0xff)
ssize++;
GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize);
/* calculate what the new size and duration would be of the packet */
payload_len = ssize + map.size + rtpceltpay->bytes + rtpceltpay->sbytes;
if (rtpceltpay->qduration != -1 && duration != -1)
packet_dur = rtpceltpay->qduration + duration;
else
packet_dur = 0;
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
if (gst_rtp_base_payload_is_filled (basepayload, packet_len, packet_dur)) {
/* size or duration would overflow the packet, flush the queued data */
ret = gst_rtp_celt_pay_flush_queued (rtpceltpay);
}
/* queue the packet */
gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, map.size, duration);
done:
rtpceltpay->packet++;
return ret;
/* ERRORS */
cleanup:
{
gst_buffer_unmap (buffer, &map);
goto done;
}
parse_error:
{
GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL),
("Error parsing first identification packet."));
gst_buffer_unmap (buffer, &map);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_visual_chain (GstPad * pad, GstBuffer * buffer)
{
GstBuffer *outbuf = NULL;
guint i;
GstVisual *visual = GST_VISUAL (gst_pad_get_parent (pad));
GstFlowReturn ret = GST_FLOW_OK;
guint avail;
GST_DEBUG_OBJECT (visual, "chain function called");
/* If we don't have an output format yet, preallocate a buffer to try and
* set one */
if (GST_PAD_CAPS (visual->srcpad) == NULL) {
ret = get_buffer (visual, &outbuf);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buffer);
goto beach;
}
}
/* resync on DISCONT */
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (visual->adapter);
}
GST_DEBUG_OBJECT (visual,
"Input buffer has %d samples, time=%" G_GUINT64_FORMAT,
GST_BUFFER_SIZE (buffer) / visual->bps, GST_BUFFER_TIMESTAMP (buffer));
gst_adapter_push (visual->adapter, buffer);
while (TRUE) {
gboolean need_skip;
const guint16 *data;
guint64 dist, timestamp;
GST_DEBUG_OBJECT (visual, "processing buffer");
avail = gst_adapter_available (visual->adapter);
GST_DEBUG_OBJECT (visual, "avail now %u", avail);
/* we need at least 512 samples */
if (avail < 512 * visual->bps)
break;
/* we need at least enough samples to make one frame */
if (avail < visual->spf * visual->bps)
break;
/* get timestamp of the current adapter byte */
timestamp = gst_adapter_prev_timestamp (visual->adapter, &dist);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert bytes to time */
dist /= visual->bps;
timestamp += gst_util_uint64_scale_int (dist, GST_SECOND, visual->rate);
}
if (timestamp != -1) {
gint64 qostime;
/* QoS is done on running time */
qostime = gst_segment_to_running_time (&visual->segment, GST_FORMAT_TIME,
timestamp);
GST_OBJECT_LOCK (visual);
/* check for QoS, don't compute buffers that are known to be late */
need_skip = visual->earliest_time != -1 &&
qostime <= visual->earliest_time;
GST_OBJECT_UNLOCK (visual);
if (need_skip) {
GST_WARNING_OBJECT (visual,
"QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT,
GST_TIME_ARGS (qostime), GST_TIME_ARGS (visual->earliest_time));
goto skip;
}
}
/* Read 512 samples per channel */
data =
(const guint16 *) gst_adapter_peek (visual->adapter, 512 * visual->bps);
#if defined(VISUAL_API_VERSION) && VISUAL_API_VERSION >= 4000 && VISUAL_API_VERSION < 5000
{
VisBuffer *lbuf, *rbuf;
guint16 ldata[512], rdata[512];
VisAudioSampleRateType rate;
lbuf = visual_buffer_new_with_buffer (ldata, sizeof (ldata), NULL);
rbuf = visual_buffer_new_with_buffer (rdata, sizeof (rdata), NULL);
if (visual->channels == 2) {
for (i = 0; i < 512; i++) {
ldata[i] = *data++;
rdata[i] = *data++;
}
} else {
for (i = 0; i < 512; i++) {
ldata[i] = *data;
rdata[i] = *data++;
}
}
switch (visual->rate) {
case 8000:
rate = VISUAL_AUDIO_SAMPLE_RATE_8000;
break;
case 11250:
rate = VISUAL_AUDIO_SAMPLE_RATE_11250;
break;
case 22500:
rate = VISUAL_AUDIO_SAMPLE_RATE_22500;
break;
case 32000:
rate = VISUAL_AUDIO_SAMPLE_RATE_32000;
break;
case 44100:
rate = VISUAL_AUDIO_SAMPLE_RATE_44100;
break;
case 48000:
rate = VISUAL_AUDIO_SAMPLE_RATE_48000;
break;
case 96000:
rate = VISUAL_AUDIO_SAMPLE_RATE_96000;
break;
default:
visual_object_unref (VISUAL_OBJECT (lbuf));
visual_object_unref (VISUAL_OBJECT (rbuf));
GST_ERROR_OBJECT (visual, "unsupported rate %d", visual->rate);
ret = GST_FLOW_ERROR;
goto beach;
break;
}
visual_audio_samplepool_input_channel (visual->audio->samplepool,
lbuf,
rate, VISUAL_AUDIO_SAMPLE_FORMAT_S16,
(char *) VISUAL_AUDIO_CHANNEL_LEFT);
visual_audio_samplepool_input_channel (visual->audio->samplepool, rbuf,
rate, VISUAL_AUDIO_SAMPLE_FORMAT_S16,
(char *) VISUAL_AUDIO_CHANNEL_RIGHT);
visual_object_unref (VISUAL_OBJECT (lbuf));
visual_object_unref (VISUAL_OBJECT (rbuf));
}
#else
if (visual->channels == 2) {
for (i = 0; i < 512; i++) {
visual->audio->plugpcm[0][i] = *data++;
visual->audio->plugpcm[1][i] = *data++;
}
} else {
for (i = 0; i < 512; i++) {
visual->audio->plugpcm[0][i] = *data;
visual->audio->plugpcm[1][i] = *data++;
}
}
#endif
/* alloc a buffer if we don't have one yet, this happens
* when we pushed a buffer in this while loop before */
if (outbuf == NULL) {
ret = get_buffer (visual, &outbuf);
if (ret != GST_FLOW_OK) {
goto beach;
}
}
visual_video_set_buffer (visual->video, GST_BUFFER_DATA (outbuf));
visual_audio_analyze (visual->audio);
visual_actor_run (visual->actor, visual->audio);
visual_video_set_buffer (visual->video, NULL);
GST_DEBUG_OBJECT (visual, "rendered one frame");
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = visual->duration;
ret = gst_pad_push (visual->srcpad, outbuf);
outbuf = NULL;
skip:
GST_DEBUG_OBJECT (visual, "finished frame, flushing %u samples from input",
visual->spf);
/* Flush out the number of samples per frame */
gst_adapter_flush (visual->adapter, visual->spf * visual->bps);
/* quit the loop if something was wrong */
if (ret != GST_FLOW_OK)
break;
}
beach:
if (outbuf != NULL)
gst_buffer_unref (outbuf);
gst_object_unref (visual);
return ret;
}
static void
gst_frame_store_task (GstPad *pad)
{
GstFrameStore *fs;
GstBuffer *buffer;
GstEvent *event = NULL;
fs = GST_FRAME_STORE (gst_pad_get_parent (pad));
GST_DEBUG("task");
g_mutex_lock (fs->lock);
while(1) {
if (fs->stepping == FALSE || (fs->frame_number != fs->pushed_frame_number)) {
buffer = gst_frame_store_get_frame (fs, fs->frame_number);
}
if (buffer) break;
g_cond_wait (fs->cond, fs->lock);
}
if (fs->need_newsegment) {
GstClock *clock;
GstClockTime now;
GstClockTime stream_time;
clock = GST_ELEMENT_CLOCK (fs);
if (clock == NULL) {
now = 0;
stream_time = 0;
} else {
now = gst_clock_get_time (GST_ELEMENT_CLOCK (fs));
stream_time = now - GST_ELEMENT(fs)->base_time;
}
GST_ERROR("now %lld buffer %lld stream_time %lld",
now, GST_BUFFER_TIMESTAMP(buffer), stream_time);
stream_time = GST_SECOND*10;
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP(buffer), -1, stream_time);
fs->need_newsegment = FALSE;
}
if (fs->stepping) {
buffer = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_TIMESTAMP(buffer) = -1;
GST_BUFFER_DURATION(buffer) = -1;
}
fs->pushed_frame_number = fs->frame_number;
if (!fs->stepping) {
fs->frame_number++;
}
if (fs->frame_number + 1 >= fs->range_offset + fs->range_size) {
gst_frame_store_advance (fs);
}
g_mutex_unlock (fs->lock);
if (event) {
gst_pad_push_event (fs->srcpad, event);
}
gst_pad_push (fs->srcpad, buffer);
GST_DEBUG("task done");
gst_object_unref (fs);
}
static GstFlowReturn
gst_ladspa_source_type_fill (GstBaseSrc * base, guint64 offset,
guint length, GstBuffer * buffer)
{
GstLADSPASource *ladspa = GST_LADSPA_SOURCE (base);
GstClockTime next_time;
gint64 next_sample, next_byte;
gint bytes, samples;
GstElementClass *eclass;
GstMapInfo map;
gint samplerate, bpf;
/* example for tagging generated data */
if (!ladspa->tags_pushed) {
GstTagList *taglist;
taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "ladspa wave", NULL);
eclass = GST_ELEMENT_CLASS (gst_ladspa_source_type_parent_class);
if (eclass->send_event)
eclass->send_event (GST_ELEMENT (base), gst_event_new_tag (taglist));
else
gst_tag_list_unref (taglist);
ladspa->tags_pushed = TRUE;
}
if (ladspa->eos_reached) {
GST_INFO_OBJECT (ladspa, "eos");
return GST_FLOW_EOS;
}
samplerate = GST_AUDIO_INFO_RATE (&ladspa->info);
bpf = GST_AUDIO_INFO_BPF (&ladspa->info);
/* if no length was given, use our default length in samples otherwise convert
* the length in bytes to samples. */
if (length == -1)
samples = ladspa->samples_per_buffer;
else
samples = length / bpf;
/* if no offset was given, use our next logical byte */
if (offset == -1)
offset = ladspa->next_byte;
/* now see if we are at the byteoffset we think we are */
if (offset != ladspa->next_byte) {
GST_DEBUG_OBJECT (ladspa, "seek to new offset %" G_GUINT64_FORMAT, offset);
/* we have a discont in the expected sample offset, do a 'seek' */
ladspa->next_sample = offset / bpf;
ladspa->next_time =
gst_util_uint64_scale_int (ladspa->next_sample, GST_SECOND, samplerate);
ladspa->next_byte = offset;
}
/* check for eos */
if (ladspa->check_seek_stop &&
(ladspa->sample_stop > ladspa->next_sample) &&
(ladspa->sample_stop < ladspa->next_sample + samples)
) {
/* calculate only partial buffer */
ladspa->generate_samples_per_buffer =
ladspa->sample_stop - ladspa->next_sample;
next_sample = ladspa->sample_stop;
ladspa->eos_reached = TRUE;
} else {
/* calculate full buffer */
ladspa->generate_samples_per_buffer = samples;
next_sample =
ladspa->next_sample + (ladspa->reverse ? (-samples) : samples);
}
bytes = ladspa->generate_samples_per_buffer * bpf;
next_byte = ladspa->next_byte + (ladspa->reverse ? (-bytes) : bytes);
next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
GST_LOG_OBJECT (ladspa, "samplerate %d", samplerate);
GST_LOG_OBJECT (ladspa,
"next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (next_time));
gst_buffer_set_size (buffer, bytes);
GST_BUFFER_OFFSET (buffer) = ladspa->next_sample;
GST_BUFFER_OFFSET_END (buffer) = next_sample;
if (!ladspa->reverse) {
GST_BUFFER_TIMESTAMP (buffer) =
ladspa->timestamp_offset + ladspa->next_time;
GST_BUFFER_DURATION (buffer) = next_time - ladspa->next_time;
} else {
GST_BUFFER_TIMESTAMP (buffer) = ladspa->timestamp_offset + next_time;
GST_BUFFER_DURATION (buffer) = ladspa->next_time - next_time;
}
gst_object_sync_values (GST_OBJECT (ladspa), GST_BUFFER_TIMESTAMP (buffer));
ladspa->next_time = next_time;
ladspa->next_sample = next_sample;
ladspa->next_byte = next_byte;
GST_LOG_OBJECT (ladspa, "generating %u samples at ts %" GST_TIME_FORMAT,
ladspa->generate_samples_per_buffer,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
gst_ladspa_transform (&ladspa->ladspa, map.data,
ladspa->generate_samples_per_buffer, NULL);
gst_buffer_unmap (buffer, &map);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_decklink_video_sink_prepare (GstBaseSink * bsink, GstBuffer * buffer)
{
GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (bsink);
GstVideoFrame vframe;
IDeckLinkMutableVideoFrame *frame;
guint8 *outdata, *indata;
GstFlowReturn flow_ret;
HRESULT ret;
GstClockTime timestamp, duration;
GstClockTime running_time, running_time_duration;
gint i;
GstClock *clock;
GST_DEBUG_OBJECT (self, "Preparing buffer %p", buffer);
// FIXME: Handle no timestamps
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
return GST_FLOW_ERROR;
}
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
if (duration == GST_CLOCK_TIME_NONE) {
duration =
gst_util_uint64_scale_int (GST_SECOND, self->info.fps_d,
self->info.fps_n);
}
running_time =
gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment,
GST_FORMAT_TIME, timestamp);
running_time_duration =
gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment,
GST_FORMAT_TIME, timestamp + duration) - running_time;
// FIXME: https://bugzilla.gnome.org/show_bug.cgi?id=742916
// We need to drop late buffers here immediately instead of
// potentially overflowing the internal queue of the hardware
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
if (clock) {
GstClockTime clock_running_time, base_time, clock_time, latency,
max_lateness;
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (self));
clock_time = gst_clock_get_time (clock);
if (base_time != GST_CLOCK_TIME_NONE && clock_time != GST_CLOCK_TIME_NONE) {
clock_running_time = clock_time - base_time;
latency = gst_base_sink_get_latency (GST_BASE_SINK_CAST (self));
max_lateness = gst_base_sink_get_max_lateness (GST_BASE_SINK_CAST (self));
if (clock_running_time >
running_time + running_time_duration + latency + max_lateness) {
GST_DEBUG_OBJECT (self,
"Late buffer: %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
GST_TIME_ARGS (clock_running_time),
GST_TIME_ARGS (running_time + running_time_duration));
if (self->last_render_time == GST_CLOCK_TIME_NONE
|| (self->last_render_time < clock_running_time
&& clock_running_time - self->last_render_time >= GST_SECOND)) {
GST_DEBUG_OBJECT (self,
"Rendering frame nonetheless because we had none for more than 1s");
running_time = clock_running_time;
running_time_duration = 0;
} else {
GST_WARNING_OBJECT (self, "Dropping frame");
gst_object_unref (clock);
return GST_FLOW_OK;
}
}
}
gst_object_unref (clock);
}
self->last_render_time = running_time;
ret = self->output->output->CreateVideoFrame (self->info.width,
self->info.height, self->info.stride[0], bmdFormat8BitYUV,
bmdFrameFlagDefault, &frame);
if (ret != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to create video frame: 0x%08x", ret));
return GST_FLOW_ERROR;
}
if (!gst_video_frame_map (&vframe, &self->info, buffer, GST_MAP_READ)) {
GST_ERROR_OBJECT (self, "Failed to map video frame");
flow_ret = GST_FLOW_ERROR;
goto out;
}
frame->GetBytes ((void **) &outdata);
indata = (guint8 *) GST_VIDEO_FRAME_PLANE_DATA (&vframe, 0);
for (i = 0; i < self->info.height; i++) {
memcpy (outdata, indata, GST_VIDEO_FRAME_WIDTH (&vframe) * 2);
indata += GST_VIDEO_FRAME_PLANE_STRIDE (&vframe, 0);
outdata += frame->GetRowBytes ();
}
gst_video_frame_unmap (&vframe);
convert_to_internal_clock (self, &running_time, &running_time_duration);
GST_LOG_OBJECT (self, "Scheduling video frame %p at %" GST_TIME_FORMAT
" with duration %" GST_TIME_FORMAT, frame, GST_TIME_ARGS (running_time),
GST_TIME_ARGS (running_time_duration));
ret = self->output->output->ScheduleVideoFrame (frame,
running_time, running_time_duration, GST_SECOND);
if (ret != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to schedule frame: 0x%08x", ret));
flow_ret = GST_FLOW_ERROR;
goto out;
}
flow_ret = GST_FLOW_OK;
out:
frame->Release ();
return flow_ret;
}
static GstFlowReturn
celt_dec_chain_parse_data (GstCeltDec * dec, GstBuffer * buf,
GstClockTime timestamp, GstClockTime duration)
{
GstFlowReturn res = GST_FLOW_OK;
gint size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
gint error = CELT_OK;
if (timestamp != -1) {
dec->segment.last_stop = timestamp;
dec->granulepos = -1;
}
if (buf) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG_OBJECT (dec, "received buffer of size %u", size);
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buf)
&& GST_BUFFER_OFFSET_END_IS_VALID (buf)) {
dec->granulepos = GST_BUFFER_OFFSET_END (buf);
GST_DEBUG_OBJECT (dec,
"Taking granulepos from upstream: %" G_GUINT64_FORMAT,
dec->granulepos);
}
/* copy timestamp */
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
data = NULL;
size = 0;
}
res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad,
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header.nb_channels * 2,
GST_PAD_CAPS (dec->srcpad), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
GST_LOG_OBJECT (dec, "decoding frame");
error = celt_decode (dec->state, data, size, out_data);
if (error != CELT_OK) {
GST_WARNING_OBJECT (dec, "Decoding error: %d", error);
return GST_FLOW_ERROR;
}
if (dec->granulepos == -1) {
if (dec->segment.format != GST_FORMAT_TIME) {
GST_WARNING_OBJECT (dec, "segment not initialized or not TIME format");
dec->granulepos = dec->frame_size;
} else {
dec->granulepos = gst_util_uint64_scale_int (dec->segment.last_stop,
dec->header.sample_rate, GST_SECOND) + dec->frame_size;
}
GST_DEBUG_OBJECT (dec, "granulepos=%" G_GINT64_FORMAT, dec->granulepos);
}
GST_BUFFER_OFFSET (outbuf) = dec->granulepos - dec->frame_size;
GST_BUFFER_OFFSET_END (outbuf) = dec->granulepos;
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale_int (dec->granulepos - dec->frame_size, GST_SECOND,
dec->header.sample_rate);
GST_BUFFER_DURATION (outbuf) = dec->frame_duration;
dec->granulepos += dec->frame_size;
dec->segment.last_stop += dec->frame_duration;
GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (dec->frame_duration));
res = gst_pad_push (dec->srcpad, outbuf);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
return res;
}
static GstFlowReturn
gst_dvbsub_overlay_chain_video (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstDVBSubOverlay *overlay = GST_DVBSUB_OVERLAY (parent);
GstFlowReturn ret = GST_FLOW_OK;
gint64 start, stop;
guint64 cstart, cstop;
gboolean in_seg;
GstClockTime vid_running_time, vid_running_time_end;
if (GST_VIDEO_INFO_FORMAT (&overlay->info) == GST_VIDEO_FORMAT_UNKNOWN)
return GST_FLOW_NOT_NEGOTIATED;
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
goto missing_timestamp;
start = GST_BUFFER_TIMESTAMP (buffer);
GST_LOG_OBJECT (overlay,
"Video segment: %" GST_SEGMENT_FORMAT " --- Subtitle position: %"
GST_TIME_FORMAT " --- BUFFER: ts=%" GST_TIME_FORMAT,
&overlay->video_segment,
GST_TIME_ARGS (overlay->subtitle_segment.position),
GST_TIME_ARGS (start));
/* ignore buffers that are outside of the current segment */
if (!GST_BUFFER_DURATION_IS_VALID (buffer)) {
stop = GST_CLOCK_TIME_NONE;
} else {
stop = start + GST_BUFFER_DURATION (buffer);
}
in_seg = gst_segment_clip (&overlay->video_segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop);
if (!in_seg) {
GST_DEBUG_OBJECT (overlay, "Buffer outside configured segment -- dropping");
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
buffer = gst_buffer_make_writable (buffer);
GST_BUFFER_TIMESTAMP (buffer) = cstart;
if (GST_BUFFER_DURATION_IS_VALID (buffer))
GST_BUFFER_DURATION (buffer) = cstop - cstart;
vid_running_time =
gst_segment_to_running_time (&overlay->video_segment, GST_FORMAT_TIME,
cstart);
if (GST_BUFFER_DURATION_IS_VALID (buffer))
vid_running_time_end =
gst_segment_to_running_time (&overlay->video_segment, GST_FORMAT_TIME,
cstop);
else
vid_running_time_end = vid_running_time;
GST_DEBUG_OBJECT (overlay, "Video running time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (vid_running_time));
overlay->video_segment.position = GST_BUFFER_TIMESTAMP (buffer);
g_mutex_lock (&overlay->dvbsub_mutex);
if (!g_queue_is_empty (overlay->pending_subtitles)) {
DVBSubtitles *tmp, *candidate = NULL;
while (!g_queue_is_empty (overlay->pending_subtitles)) {
tmp = g_queue_peek_head (overlay->pending_subtitles);
if (tmp->pts > vid_running_time_end) {
/* For a future video frame */
break;
} else if (tmp->num_rects == 0) {
/* Clear screen */
if (overlay->current_subtitle)
dvb_subtitles_free (overlay->current_subtitle);
overlay->current_subtitle = NULL;
if (candidate)
dvb_subtitles_free (candidate);
candidate = NULL;
g_queue_pop_head (overlay->pending_subtitles);
dvb_subtitles_free (tmp);
tmp = NULL;
} else if (tmp->pts + tmp->page_time_out * GST_SECOND *
ABS (overlay->subtitle_segment.rate) >= vid_running_time) {
if (candidate)
dvb_subtitles_free (candidate);
candidate = tmp;
g_queue_pop_head (overlay->pending_subtitles);
} else {
/* Too late */
dvb_subtitles_free (tmp);
tmp = NULL;
g_queue_pop_head (overlay->pending_subtitles);
}
}
if (candidate) {
GST_DEBUG_OBJECT (overlay,
"Time to show the next subtitle page (%" GST_TIME_FORMAT " >= %"
GST_TIME_FORMAT ") - it has %u regions",
GST_TIME_ARGS (vid_running_time), GST_TIME_ARGS (candidate->pts),
candidate->num_rects);
dvb_subtitles_free (overlay->current_subtitle);
overlay->current_subtitle = candidate;
if (overlay->current_comp)
gst_video_overlay_composition_unref (overlay->current_comp);
overlay->current_comp =
gst_dvbsub_overlay_subs_to_comp (overlay, overlay->current_subtitle);
}
}
/* Check that we haven't hit the fallback timeout for current subtitle page */
if (overlay->current_subtitle
&& vid_running_time >
(overlay->current_subtitle->pts +
overlay->current_subtitle->page_time_out * GST_SECOND *
ABS (overlay->subtitle_segment.rate))) {
GST_INFO_OBJECT (overlay,
"Subtitle page not redefined before fallback page_time_out of %u seconds (missed data?) - deleting current page",
overlay->current_subtitle->page_time_out);
dvb_subtitles_free (overlay->current_subtitle);
overlay->current_subtitle = NULL;
}
/* Now render it */
if (g_atomic_int_get (&overlay->enable) && overlay->current_subtitle) {
GstVideoFrame frame;
g_assert (overlay->current_comp);
if (overlay->attach_compo_to_buffer) {
GST_DEBUG_OBJECT (overlay, "Attaching overlay image to video buffer");
gst_buffer_add_video_overlay_composition_meta (buffer,
overlay->current_comp);
} else {
GST_DEBUG_OBJECT (overlay, "Blending overlay image to video buffer");
gst_video_frame_map (&frame, &overlay->info, buffer, GST_MAP_READWRITE);
gst_video_overlay_composition_blend (overlay->current_comp, &frame);
gst_video_frame_unmap (&frame);
}
}
g_mutex_unlock (&overlay->dvbsub_mutex);
ret = gst_pad_push (overlay->srcpad, buffer);
return ret;
missing_timestamp:
{
GST_WARNING_OBJECT (overlay, "video buffer without timestamp, discarding");
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
}
static GstFlowReturn
gst_decklink_video_sink_prepare (GstBaseSink * bsink, GstBuffer * buffer)
{
GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (bsink);
GstVideoFrame vframe;
IDeckLinkMutableVideoFrame *frame;
guint8 *outdata, *indata;
GstFlowReturn flow_ret;
HRESULT ret;
GstClockTime timestamp, duration;
GstClockTime running_time, running_time_duration;
GstClockTime latency, render_delay;
GstClockTimeDiff ts_offset;
gint i;
GstDecklinkVideoFormat caps_format;
BMDPixelFormat format;
gint stride;
GstVideoTimeCodeMeta *tc_meta;
GST_DEBUG_OBJECT (self, "Preparing buffer %p", buffer);
// FIXME: Handle no timestamps
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
return GST_FLOW_ERROR;
}
caps_format = gst_decklink_type_from_video_format (self->info.finfo->format);
format = gst_decklink_pixel_format_from_type (caps_format);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
if (duration == GST_CLOCK_TIME_NONE) {
duration =
gst_util_uint64_scale_int (GST_SECOND, self->info.fps_d,
self->info.fps_n);
}
running_time =
gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment,
GST_FORMAT_TIME, timestamp);
running_time_duration =
gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment,
GST_FORMAT_TIME, timestamp + duration) - running_time;
/* See gst_base_sink_adjust_time() */
latency = gst_base_sink_get_latency (bsink);
render_delay = gst_base_sink_get_render_delay (bsink);
ts_offset = gst_base_sink_get_ts_offset (bsink);
running_time += latency;
if (ts_offset < 0) {
ts_offset = -ts_offset;
if ((GstClockTime) ts_offset < running_time)
running_time -= ts_offset;
else
running_time = 0;
} else {
running_time += ts_offset;
}
if (running_time > render_delay)
running_time -= render_delay;
else
running_time = 0;
ret = self->output->output->CreateVideoFrame (self->info.width,
self->info.height, self->info.stride[0], format, bmdFrameFlagDefault,
&frame);
if (ret != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to create video frame: 0x%08x", ret));
return GST_FLOW_ERROR;
}
if (!gst_video_frame_map (&vframe, &self->info, buffer, GST_MAP_READ)) {
GST_ERROR_OBJECT (self, "Failed to map video frame");
flow_ret = GST_FLOW_ERROR;
goto out;
}
frame->GetBytes ((void **) &outdata);
indata = (guint8 *) GST_VIDEO_FRAME_PLANE_DATA (&vframe, 0);
stride = MIN (GST_VIDEO_FRAME_PLANE_STRIDE (&vframe, 0), frame->GetRowBytes());
for (i = 0; i < self->info.height; i++) {
memcpy (outdata, indata, stride);
indata += GST_VIDEO_FRAME_PLANE_STRIDE (&vframe, 0);
outdata += frame->GetRowBytes ();
}
gst_video_frame_unmap (&vframe);
tc_meta = gst_buffer_get_video_time_code_meta (buffer);
if (tc_meta) {
BMDTimecodeFlags bflags = (BMDTimecodeFlags) 0;
gchar *tc_str;
if (((GstVideoTimeCodeFlags) (tc_meta->tc.
config.flags)) & GST_VIDEO_TIME_CODE_FLAGS_DROP_FRAME)
bflags = (BMDTimecodeFlags) (bflags | bmdTimecodeIsDropFrame);
else
bflags = (BMDTimecodeFlags) (bflags | bmdTimecodeFlagDefault);
if (tc_meta->tc.field_count == 2)
bflags = (BMDTimecodeFlags) (bflags | bmdTimecodeFieldMark);
tc_str = gst_video_time_code_to_string (&tc_meta->tc);
ret = frame->SetTimecodeFromComponents (self->timecode_format,
(uint8_t) tc_meta->tc.hours,
(uint8_t) tc_meta->tc.minutes,
(uint8_t) tc_meta->tc.seconds, (uint8_t) tc_meta->tc.frames, bflags);
if (ret != S_OK) {
GST_ERROR_OBJECT (self,
"Failed to set timecode %s to video frame: 0x%08x", tc_str, ret);
flow_ret = GST_FLOW_ERROR;
g_free (tc_str);
goto out;
}
GST_DEBUG_OBJECT (self, "Set frame timecode to %s", tc_str);
g_free (tc_str);
}
convert_to_internal_clock (self, &running_time, &running_time_duration);
if (!self->output->started) {
GST_LOG_OBJECT (self, "Showing video frame synchronously because PAUSED");
ret = self->output->output->DisplayVideoFrameSync (frame);
if (ret != S_OK) {
GST_ELEMENT_WARNING (self, STREAM, FAILED,
(NULL), ("Failed to show video frame synchronously: 0x%08x", ret));
ret = S_OK;
}
}
GST_LOG_OBJECT (self, "Scheduling video frame %p at %" GST_TIME_FORMAT
" with duration %" GST_TIME_FORMAT, frame, GST_TIME_ARGS (running_time),
GST_TIME_ARGS (running_time_duration));
ret = self->output->output->ScheduleVideoFrame (frame,
running_time, running_time_duration, GST_SECOND);
if (ret != S_OK) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Failed to schedule frame: 0x%08x", ret));
flow_ret = GST_FLOW_ERROR;
goto out;
}
flow_ret = GST_FLOW_OK;
out:
frame->Release ();
return flow_ret;
}
static GstFlowReturn
gst_gdiscreencapsrc_create (GstPushSrc * push_src, GstBuffer ** buf)
{
GstGDIScreenCapSrc *src = GST_GDISCREENCAPSRC (push_src);
GstBuffer *new_buf;
GstFlowReturn res;
gint new_buf_size;
GstClock *clock;
GstClockTime time;
GstClockTime base_time;
if (G_UNLIKELY (!src->info.bmiHeader.biWidth ||
!src->info.bmiHeader.biHeight)) {
GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before create function"));
return GST_FLOW_NOT_NEGOTIATED;
} else if (G_UNLIKELY (src->rate_numerator == 0 && src->frames == 1)) {
GST_DEBUG_OBJECT (src, "eos: 0 framerate, frame %d", (gint) src->frames);
return GST_FLOW_UNEXPECTED;
}
new_buf_size = GST_ROUND_UP_4 (src->info.bmiHeader.biWidth * 3) *
(-src->info.bmiHeader.biHeight);
GST_LOG_OBJECT (src,
"creating buffer of %lu bytes with %dx%d image for frame %d",
new_buf_size, src->info.bmiHeader.biWidth,
-src->info.bmiHeader.biHeight, (gint) src->frames);
res =
gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (src),
GST_BUFFER_OFFSET_NONE, new_buf_size,
GST_PAD_CAPS (GST_BASE_SRC_PAD (push_src)), &new_buf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (src, "could not allocate buffer, reason %s",
gst_flow_get_name (res));
return res;
}
clock = gst_element_get_clock (GST_ELEMENT (src));
if (clock) {
/* Calculate sync time. */
GstClockTime frame_time =
gst_util_uint64_scale_int (src->frames * GST_SECOND,
src->rate_denominator, src->rate_numerator);
time = gst_clock_get_time (clock);
base_time = gst_element_get_base_time (GST_ELEMENT (src));
GST_BUFFER_TIMESTAMP (new_buf) = MAX (time - base_time, frame_time);
} else {
GST_BUFFER_TIMESTAMP (new_buf) = GST_CLOCK_TIME_NONE;
}
/* Do screen capture and put it into buffer... */
gst_gdiscreencapsrc_screen_capture (src, new_buf);
if (src->rate_numerator) {
GST_BUFFER_DURATION (new_buf) =
gst_util_uint64_scale_int (GST_SECOND,
src->rate_denominator, src->rate_numerator);
if (clock) {
GST_BUFFER_DURATION (new_buf) =
MAX (GST_BUFFER_DURATION (new_buf),
gst_clock_get_time (clock) - time);
}
} else {
/* NONE means forever */
GST_BUFFER_DURATION (new_buf) = GST_CLOCK_TIME_NONE;
}
GST_BUFFER_OFFSET (new_buf) = src->frames;
src->frames++;
GST_BUFFER_OFFSET_END (new_buf) = src->frames;
gst_object_unref (clock);
*buf = new_buf;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_rsvg_dec_chain (GstPad * pad, GstBuffer * buffer)
{
GstRsvgDec *rsvg = GST_RSVG_DEC (GST_PAD_PARENT (pad));
gboolean completed = FALSE;
const guint8 *data;
guint size;
gboolean ret = GST_FLOW_OK;
/* first_timestamp is used slightly differently where a framerate
is given or not.
If there is a frame rate, it will be used as a base.
If there is not, it will be used to keep track of the timestamp
of the first buffer, to be used as the timestamp of the output
buffer. When a buffer is output, first timestamp will resync to
the next buffer's timestamp. */
if (rsvg->first_timestamp == GST_CLOCK_TIME_NONE) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
rsvg->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
else if (rsvg->fps_n != 0)
rsvg->first_timestamp = 0;
}
gst_adapter_push (rsvg->adapter, buffer);
size = gst_adapter_available (rsvg->adapter);
/* "<svg></svg>" */
while (size >= 5 + 6 && ret == GST_FLOW_OK) {
guint i;
data = gst_adapter_peek (rsvg->adapter, size);
for (i = size - 6; i >= 5; i--) {
if (memcmp (data + i, "</svg>", 6) == 0) {
completed = TRUE;
size = i + 6;
break;
}
}
if (completed) {
GstBuffer *outbuf = NULL;
GST_LOG_OBJECT (rsvg, "have complete svg of %u bytes", size);
data = gst_adapter_peek (rsvg->adapter, size);
ret = gst_rsvg_decode_image (rsvg, data, size, &outbuf);
if (ret != GST_FLOW_OK)
break;
if (rsvg->first_timestamp != GST_CLOCK_TIME_NONE) {
GST_BUFFER_TIMESTAMP (outbuf) = rsvg->first_timestamp;
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
GstClockTime end =
GST_BUFFER_TIMESTAMP_IS_VALID (buffer) ?
GST_BUFFER_TIMESTAMP (buffer) : rsvg->first_timestamp;
end += GST_BUFFER_DURATION (buffer);
GST_BUFFER_DURATION (outbuf) = end - GST_BUFFER_TIMESTAMP (outbuf);
}
if (rsvg->fps_n == 0) {
rsvg->first_timestamp = GST_CLOCK_TIME_NONE;
} else {
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (rsvg->frame_count, rsvg->fps_d,
rsvg->fps_n * GST_SECOND);
}
} else if (rsvg->fps_n != 0) {
GST_BUFFER_TIMESTAMP (outbuf) =
rsvg->first_timestamp + gst_util_uint64_scale (rsvg->frame_count,
rsvg->fps_d, rsvg->fps_n * GST_SECOND);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (rsvg->frame_count, rsvg->fps_d,
rsvg->fps_n * GST_SECOND);
} else {
GST_BUFFER_TIMESTAMP (outbuf) = rsvg->first_timestamp;
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
}
rsvg->frame_count++;
if (rsvg->need_newsegment) {
gst_pad_push_event (rsvg->srcpad,
gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0));
rsvg->need_newsegment = FALSE;
}
if (rsvg->pending_events) {
GList *l;
for (l = rsvg->pending_events; l; l = l->next)
gst_pad_push_event (rsvg->srcpad, l->data);
g_list_free (rsvg->pending_events);
rsvg->pending_events = NULL;
}
if (rsvg->pending_tags) {
gst_element_found_tags (GST_ELEMENT_CAST (rsvg), rsvg->pending_tags);
rsvg->pending_tags = NULL;
}
GST_LOG_OBJECT (rsvg, "image rendered okay");
ret = gst_pad_push (rsvg->srcpad, outbuf);
if (ret != GST_FLOW_OK)
break;
gst_adapter_flush (rsvg->adapter, size);
size = gst_adapter_available (rsvg->adapter);
continue;
} else {
break;
}
}
return GST_FLOW_OK;
}
