PassRefPtr<AudioBus> AudioFileReader::createBus(float sampleRate, bool mixToMono)
{
m_sampleRate = sampleRate;
m_channels = mixToMono ? 1 : 2;
m_frontLeftBuffers = gst_buffer_list_new();
m_frontRightBuffers = gst_buffer_list_new();
GRefPtr<GMainContext> context = adoptGRef(g_main_context_new());
g_main_context_push_thread_default(context.get());
m_loop = adoptGRef(g_main_loop_new(context.get(), FALSE));
// Start the pipeline processing just after the loop is started.
GThreadSafeMainLoopSource source;
source.schedule("[WebKit] AudioFileReader::decodeAudioForBusCreation", std::function<void()>(std::bind(&AudioFileReader::decodeAudioForBusCreation, this)), G_PRIORITY_DEFAULT, nullptr, context.get());
g_main_loop_run(m_loop.get());
g_main_context_pop_thread_default(context.get());
// Set pipeline to GST_STATE_NULL state here already ASAP to
// release any resources that might still be used.
gst_element_set_state(m_pipeline, GST_STATE_NULL);
if (m_errorOccurred)
return 0;
RefPtr<AudioBus> audioBus = AudioBus::create(m_channels, m_channelSize, true);
audioBus->setSampleRate(m_sampleRate);
copyGstreamerBuffersToAudioChannel(m_frontLeftBuffers, audioBus->channel(0));
if (!mixToMono)
copyGstreamerBuffersToAudioChannel(m_frontRightBuffers, audioBus->channel(1));
return audioBus;
}
PassRefPtr<AudioBus> AudioFileReader::createBus(float sampleRate, bool mixToMono)
{
m_sampleRate = sampleRate;
m_frontLeftBuffers = gst_buffer_list_new();
m_frontRightBuffers = gst_buffer_list_new();
GRefPtr<GMainContext> context = adoptGRef(g_main_context_new());
g_main_context_push_thread_default(context.get());
m_loop = adoptGRef(g_main_loop_new(context.get(), FALSE));
// Start the pipeline processing just after the loop is started.
GRefPtr<GSource> timeoutSource = adoptGRef(g_timeout_source_new(0));
g_source_attach(timeoutSource.get(), context.get());
g_source_set_callback(timeoutSource.get(), reinterpret_cast<GSourceFunc>(enteredMainLoopCallback), this, 0);
g_main_loop_run(m_loop.get());
g_main_context_pop_thread_default(context.get());
if (m_errorOccurred)
return 0;
unsigned channels = mixToMono ? 1 : 2;
RefPtr<AudioBus> audioBus = AudioBus::create(channels, m_channelSize, true);
audioBus->setSampleRate(m_sampleRate);
copyGstreamerBuffersToAudioChannel(m_frontLeftBuffers, audioBus->channel(0));
if (!mixToMono)
copyGstreamerBuffersToAudioChannel(m_frontRightBuffers, audioBus->channel(1));
return audioBus;
}
static gboolean
new_packet_cb (guint8 * data, guint len, void *user_data)
{
/* Called when the PsMux has prepared a packet for output. Return FALSE
* on error */
MpegPsMux *mux = (MpegPsMux *) user_data;
GstBuffer *buf;
GstFlowReturn ret;
GST_LOG_OBJECT (mux, "Outputting a packet of length %d", len);
data = g_memdup (data, len);
buf = gst_buffer_new_wrapped (data, len);
GST_BUFFER_TIMESTAMP (buf) = mux->last_ts;
if (mux->aggregate_gops) {
if (mux->gop_list == NULL)
mux->gop_list = gst_buffer_list_new ();
gst_buffer_list_add (mux->gop_list, buf);
return TRUE;
}
ret = gst_pad_push (mux->srcpad, buf);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
mux->last_flow_ret = ret;
return FALSE;
}
return TRUE;
}
static GstBufferList *
_create_buffer_list (guint * data_size)
{
GstBufferList *list;
GstBufferListIterator *it;
GstBuffer *rtp_buffer;
GstBuffer *data_buffer;
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
/*** First group, i.e. first packet. **/
/* Create the RTP header buffer */
rtp_buffer = gst_buffer_new ();
GST_BUFFER_MALLOCDATA (rtp_buffer) = g_malloc (RTP_HEADER_SIZE);
GST_BUFFER_DATA (rtp_buffer) = GST_BUFFER_MALLOCDATA (rtp_buffer);
GST_BUFFER_SIZE (rtp_buffer) = RTP_HEADER_SIZE;
memset (GST_BUFFER_DATA (rtp_buffer), 0, RTP_HEADER_SIZE);
/* Create the buffer that holds the payload */
data_buffer = gst_buffer_new ();
GST_BUFFER_MALLOCDATA (data_buffer) = g_malloc (RTP_PAYLOAD_SIZE);
GST_BUFFER_DATA (data_buffer) = GST_BUFFER_MALLOCDATA (data_buffer);
GST_BUFFER_SIZE (data_buffer) = RTP_PAYLOAD_SIZE;
memset (GST_BUFFER_DATA (data_buffer), 0, RTP_PAYLOAD_SIZE);
/* Create a new group to hold the rtp header and the payload */
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, rtp_buffer);
gst_buffer_list_iterator_add (it, data_buffer);
/*** Second group, i.e. second packet. ***/
/* Create the RTP header buffer */
rtp_buffer = gst_buffer_new ();
GST_BUFFER_MALLOCDATA (rtp_buffer) = g_malloc (RTP_HEADER_SIZE);
GST_BUFFER_DATA (rtp_buffer) = GST_BUFFER_MALLOCDATA (rtp_buffer);
GST_BUFFER_SIZE (rtp_buffer) = RTP_HEADER_SIZE;
memset (GST_BUFFER_DATA (rtp_buffer), 0, RTP_HEADER_SIZE);
/* Create the buffer that holds the payload */
data_buffer = gst_buffer_new ();
GST_BUFFER_MALLOCDATA (data_buffer) = g_malloc (RTP_PAYLOAD_SIZE);
GST_BUFFER_DATA (data_buffer) = GST_BUFFER_MALLOCDATA (data_buffer);
GST_BUFFER_SIZE (data_buffer) = RTP_PAYLOAD_SIZE;
memset (GST_BUFFER_DATA (data_buffer), 0, RTP_PAYLOAD_SIZE);
/* Create a new group to hold the rtp header and the payload */
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, rtp_buffer);
gst_buffer_list_iterator_add (it, data_buffer);
/* Calculate the size of the data */
*data_size = 2 * RTP_HEADER_SIZE + 2 * RTP_PAYLOAD_SIZE;
gst_buffer_list_iterator_free (it);
return list;
}
static GstBufferList *
create_buffer_list (void)
{
guint len;
GstBuffer *buffer;
mylist = gst_buffer_list_new ();
fail_if (mylist == NULL);
mycaps = gst_caps_from_string ("application/x-gst-check");
fail_if (mycaps == NULL);
len = gst_buffer_list_length (mylist);
fail_if (len != 0);
buffer = gst_buffer_new_and_alloc (sizeof (gint));
gst_buffer_fill (buffer, 0, &values[0], sizeof (gint));
gst_buffer_list_add (mylist, buffer);
buffer = gst_buffer_new_and_alloc (sizeof (gint));
gst_buffer_fill (buffer, 0, &values[1], sizeof (gint));
gst_buffer_list_add (mylist, buffer);
buffer = gst_buffer_new_and_alloc (sizeof (gint));
gst_buffer_fill (buffer, 0, &values[2], sizeof (gint));
gst_buffer_list_add (mylist, buffer);
gst_pad_set_caps (mysrcpad, mycaps);
gst_caps_unref (mycaps);
return mylist;
}
static GstFlowReturn
gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
payload_len = gst_buffer_get_size (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create just the RTP header buffer */
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
guint i, len;
list = gst_buffer_list_new ();
len = gst_buffer_list_length (list);
for (i = 0; i < len; i++) {
/* FIXME */
g_warning ("bufferlist not implemented");
gst_buffer_list_add (list, outbuf);
gst_buffer_list_add (list, buffer);
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_rtp_base_payload_push_list (basepayload, list);
} else {
CopyMetaData data;
/* copy payload */
data.pay = baseaudiopayload;
data.outbuf = outbuf;
gst_buffer_foreach_meta (buffer, foreach_metadata, &data);
outbuf = gst_buffer_append (outbuf, buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}
PassOwnPtr<AudioBus> AudioFileReader::createBus(float sampleRate, bool mixToMono)
{
m_sampleRate = sampleRate;
m_frontLeftBuffers = gst_buffer_list_new();
m_frontRightBuffers = gst_buffer_list_new();
#ifndef GST_API_VERSION_1
m_frontLeftBuffersIterator = gst_buffer_list_iterate(m_frontLeftBuffers);
gst_buffer_list_iterator_add_group(m_frontLeftBuffersIterator);
m_frontRightBuffersIterator = gst_buffer_list_iterate(m_frontRightBuffers);
gst_buffer_list_iterator_add_group(m_frontRightBuffersIterator);
#endif
GRefPtr<GMainContext> context = g_main_context_new();
g_main_context_push_thread_default(context.get());
m_loop = adoptGRef(g_main_loop_new(context.get(), FALSE));
// Start the pipeline processing just after the loop is started.
GSource* timeoutSource = g_timeout_source_new(0);
g_source_attach(timeoutSource, context.get());
g_source_set_callback(timeoutSource, reinterpret_cast<GSourceFunc>(enteredMainLoopCallback), this, 0);
g_main_loop_run(m_loop.get());
g_main_context_pop_thread_default(context.get());
if (m_errorOccurred)
return nullptr;
unsigned channels = mixToMono ? 1 : 2;
OwnPtr<AudioBus> audioBus = adoptPtr(new AudioBus(channels, m_channelSize, true));
audioBus->setSampleRate(m_sampleRate);
copyGstreamerBuffersToAudioChannel(m_frontLeftBuffers, audioBus->channel(0));
if (!mixToMono)
copyGstreamerBuffersToAudioChannel(m_frontRightBuffers, audioBus->channel(1));
return audioBus.release();
}
static void
add_buffer_to_goplist (MpegPsMux * mux, GstBuffer * buf)
{
GstBufferListIterator *it;
if (mux->gop_list == NULL)
mux->gop_list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (mux->gop_list);
/* move iterator to end */
while (gst_buffer_list_iterator_next_group (it)) {
/* .. */
}
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, buf);
gst_buffer_list_iterator_free (it);
}
static void
gst_fragment_init (GstFragment * fragment)
{
GstFragmentPrivate *priv;
fragment->priv = priv = GST_FRAGMENT_GET_PRIVATE (fragment);
g_mutex_init (&fragment->priv->lock);
priv->buffer_list = gst_buffer_list_new ();
priv->size = 0;
priv->buffer_iterator = gst_buffer_list_iterate (priv->buffer_list);
gst_buffer_list_iterator_add_group (priv->buffer_iterator);
fragment->download_start_time = g_get_real_time ();
fragment->start_time = 0;
fragment->stop_time = 0;
fragment->index = 0;
fragment->name = g_strdup ("");
fragment->completed = FALSE;
fragment->discontinuous = FALSE;
}
static GstBufferList *
create_buffer_list (guint * data_size)
{
GstBufferList *list;
GstBuffer *rtp_buffer;
GstBuffer *data_buffer;
list = gst_buffer_list_new ();
/*** First group, i.e. first packet. **/
/* Create the RTP header buffer */
rtp_buffer = gst_buffer_new_allocate (NULL, RTP_HEADER_SIZE, NULL);
gst_buffer_memset (rtp_buffer, 0, 0, RTP_HEADER_SIZE);
/* Create the buffer that holds the payload */
data_buffer = gst_buffer_new_allocate (NULL, RTP_PAYLOAD_SIZE, NULL);
gst_buffer_memset (data_buffer, 0, 0, RTP_PAYLOAD_SIZE);
/* Create a new group to hold the rtp header and the payload */
gst_buffer_list_add (list, gst_buffer_append (rtp_buffer, data_buffer));
/*** Second group, i.e. second packet. ***/
/* Create the RTP header buffer */
rtp_buffer = gst_buffer_new_allocate (NULL, RTP_HEADER_SIZE, NULL);
gst_buffer_memset (rtp_buffer, 0, 0, RTP_HEADER_SIZE);
/* Create the buffer that holds the payload */
data_buffer = gst_buffer_new_allocate (NULL, RTP_PAYLOAD_SIZE, NULL);
gst_buffer_memset (data_buffer, 0, 0, RTP_PAYLOAD_SIZE);
/* Create a new group to hold the rtp header and the payload */
gst_buffer_list_add (list, gst_buffer_append (rtp_buffer, data_buffer));
/* Calculate the size of the data */
*data_size = 2 * RTP_HEADER_SIZE + 2 * RTP_PAYLOAD_SIZE;
return list;
}
static GstBufferList *
_gst_buffer_list_copy (GstBufferList * list)
{
GstBufferList *list_copy;
GList *tmp;
g_return_val_if_fail (list != NULL, NULL);
list_copy = gst_buffer_list_new ();
/* shallow copy of list and pointers */
list_copy->buffers = g_list_copy (list->buffers);
/* ref all buffers in the list */
tmp = list_copy->buffers;
while (tmp) {
if (tmp->data != GROUP_START && tmp->data != STOLEN) {
tmp->data = gst_buffer_ref (GST_BUFFER_CAST (tmp->data));
}
tmp = g_list_next (tmp);
}
return list_copy;
}
static GstFlowReturn
gst_rtp_j2k_depay_flush_frame (GstBaseRTPDepayload * depayload)
{
GstRtpJ2KDepay *rtpj2kdepay;
guint8 end[2];
guint8 *data;
guint avail;
GstFlowReturn ret = GST_FLOW_OK;
rtpj2kdepay = GST_RTP_J2K_DEPAY (depayload);
/* flush pending tile */
gst_rtp_j2k_depay_flush_tile (depayload);
/* last buffer take all data out of the adapter */
avail = gst_adapter_available (rtpj2kdepay->f_adapter);
if (avail == 0)
goto done;
if (avail > 2) {
GstBuffer *outbuf;
/* take the last bytes of the JPEG 2000 data to see if there is an EOC
* marker */
gst_adapter_copy (rtpj2kdepay->f_adapter, end, avail - 2, 2);
if (end[0] != 0xff && end[1] != 0xd9) {
GST_DEBUG_OBJECT (rtpj2kdepay, "no EOC marker, adding one");
/* no EOI marker, add one */
outbuf = gst_buffer_new_and_alloc (2);
data = GST_BUFFER_DATA (outbuf);
data[0] = 0xff;
data[1] = 0xd9;
gst_adapter_push (rtpj2kdepay->f_adapter, outbuf);
avail += 2;
}
if (rtpj2kdepay->buffer_list) {
GList *list;
GstBufferList *buflist;
GstBufferListIterator *it;
GST_DEBUG_OBJECT (rtpj2kdepay, "pushing buffer list of %u bytes", avail);
list = gst_adapter_take_list (rtpj2kdepay->f_adapter, avail);
buflist = gst_buffer_list_new ();
it = gst_buffer_list_iterate (buflist);
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add_list (it, list);
gst_buffer_list_iterator_free (it);
ret = gst_base_rtp_depayload_push_list (depayload, buflist);
} else {
GST_DEBUG_OBJECT (rtpj2kdepay, "pushing buffer of %u bytes", avail);
outbuf = gst_adapter_take_buffer (rtpj2kdepay->f_adapter, avail);
ret = gst_base_rtp_depayload_push (depayload, outbuf);
}
} else {
GST_WARNING_OBJECT (rtpj2kdepay, "empty packet");
gst_adapter_clear (rtpj2kdepay->f_adapter);
}
/* we accept any mh_id now */
rtpj2kdepay->last_mh_id = -1;
/* reset state */
rtpj2kdepay->next_frag = 0;
rtpj2kdepay->have_sync = FALSE;
done:
/* we can't keep headers with mh_id of 0 */
store_mheader (rtpj2kdepay, 0, NULL);
return ret;
}
static void
setup (void)
{
list = gst_buffer_list_new ();
caps = gst_caps_new_simple ("text/plain", NULL);
}
static GstFlowReturn
gst_rtp_klv_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstBufferList *list = NULL;
GstRtpKlvPay *pay;
GstMapInfo map;
GstBuffer *outbuf = NULL;
gsize offset;
guint mtu, rtp_header_size, max_payload_size;
pay = GST_RTP_KLV_PAY (basepayload);
mtu = GST_RTP_BASE_PAYLOAD_MTU (basepayload);
rtp_header_size = gst_rtp_buffer_calc_header_len (0);
max_payload_size = mtu - rtp_header_size;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (map.size == 0)
goto done;
/* KLV coding shall use and only use a fixed 16-byte SMPTE-administered
* Universal Label, according to SMPTE 298M as Key (Rec. ITU R-BT.1653-1) */
if (map.size < 16 || GST_READ_UINT32_BE (map.data) != 0x060E2B34)
goto bad_input;
if (map.size > max_payload_size)
list = gst_buffer_list_new ();
GST_LOG_OBJECT (pay, "%" G_GSIZE_FORMAT " bytes of data to payload",
map.size);
offset = 0;
while (offset < map.size) {
GstBuffer *payloadbuf;
GstRTPBuffer rtp = { NULL };
guint payload_size;
guint bytes_left;
bytes_left = map.size - offset;
payload_size = MIN (bytes_left, max_payload_size);
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
if (payload_size == bytes_left) {
GST_LOG_OBJECT (pay, "last packet of KLV unit");
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
gst_rtp_buffer_set_marker (&rtp, 1);
gst_rtp_buffer_unmap (&rtp);
}
GST_LOG_OBJECT (pay, "packet with payload size %u", payload_size);
gst_rtp_copy_meta (GST_ELEMENT_CAST (pay), outbuf, buf, 0);
payloadbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_MEMORY,
offset, payload_size);
/* join rtp header + payload memory parts */
outbuf = gst_buffer_append (outbuf, payloadbuf);
GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buf);
GST_BUFFER_DTS (outbuf) = GST_BUFFER_DTS (buf);
/* and add to list */
if (list != NULL)
gst_buffer_list_insert (list, -1, outbuf);
offset += payload_size;
}
done:
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
if (list != NULL)
ret = gst_rtp_base_payload_push_list (basepayload, list);
else if (outbuf != NULL)
ret = gst_rtp_base_payload_push (basepayload, outbuf);
return ret;
/* ERRORS */
bad_input:
{
GST_ERROR_OBJECT (pay, "Input doesn't look like a KLV packet, ignoring");
goto done;
}
}
static GstFlowReturn
gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload *
baseaudiopayload, GstBuffer * buffer)
{
GstBaseRTPPayload *basepayload;
GstBaseRTPAudioPayloadPrivate *priv;
GstBuffer *outbuf;
GstClockTime timestamp;
guint8 *payload;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
payload_len = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list) {
/* create just the RTP header buffer */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
} else {
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
}
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
GstBufferListIterator *it;
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
/* add both buffers to the buffer list */
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, outbuf);
gst_buffer_list_iterator_add (it, buffer);
gst_buffer_list_iterator_free (it);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_basertppayload_push_list (basepayload, list);
} else {
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, GST_BUFFER_DATA (buffer), payload_len);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_basertppayload_push (basepayload, outbuf);
}
return ret;
}
BufferListPtr BufferList::create()
{
return BufferListPtr::wrap(gst_buffer_list_new(), false);
}
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
gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint8 *payload;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
payload_len = gst_buffer_get_size (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list) {
/* create just the RTP header buffer */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
} else {
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
}
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
guint i, len;
list = gst_buffer_list_new ();
len = gst_buffer_list_length (list);
for (i = 0; i < len; i++) {
/* FIXME */
g_warning ("bufferlist not implemented");
gst_buffer_list_add (list, outbuf);
gst_buffer_list_add (list, buffer);
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_rtp_base_payload_push_list (basepayload, list);
} else {
GstRTPBuffer rtp = { NULL };
/* copy payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
gst_buffer_extract (buffer, 0, payload, payload_len);
gst_rtp_buffer_unmap (&rtp);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}
