static GstFlowReturn
gst_rnd_buffer_size_drain_adapter (GstRndBufferSize * self, gboolean eos)
{
GstFlowReturn flow;
GstBuffer *buf;
guint num_bytes, avail;
flow = GST_FLOW_OK;
if (G_UNLIKELY (self->min > self->max))
goto bogus_minmax;
do {
if (self->min != self->max) {
num_bytes = g_rand_int_range (self->rand, self->min, self->max);
} else {
num_bytes = self->min;
}
GST_LOG_OBJECT (self, "pulling %u bytes out of adapter", num_bytes);
buf = gst_adapter_take_buffer (self->adapter, num_bytes);
if (buf == NULL) {
if (!eos) {
GST_LOG_OBJECT (self, "not enough bytes in adapter");
break;
}
avail = gst_adapter_available (self->adapter);
if (avail == 0)
break;
if (avail < self->min) {
GST_WARNING_OBJECT (self, "discarding %u bytes at end (min=%u)",
avail, self->min);
gst_adapter_clear (self->adapter);
break;
}
buf = gst_adapter_take_buffer (self->adapter, avail);
g_assert (buf != NULL);
}
flow = gst_pad_push (self->srcpad, buf);
}
while (flow == GST_FLOW_OK);
return flow;
/* ERRORS */
bogus_minmax:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS,
("The minimum buffer size is smaller than the maximum buffer size."),
("buffer sizes: max=%d, min=%d", self->min, self->max));
return GST_FLOW_ERROR;
}
}
static gboolean
gst_hls_demux_fetch_location (GstHLSDemux * demux, const gchar * uri)
{
GstStateChangeReturn ret;
gboolean bret = FALSE;
g_mutex_lock (demux->fetcher_lock);
while (demux->fetcher)
g_cond_wait (demux->fetcher_cond, demux->fetcher_lock);
if (demux->cancelled)
goto quit;
if (!gst_hls_demux_make_fetcher_locked (demux, uri)) {
goto uri_error;
}
ret = gst_element_set_state (demux->fetcher, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE)
goto state_change_error;
/* wait until we have fetched the uri */
GST_DEBUG_OBJECT (demux, "Waiting to fetch the URI");
g_cond_wait (demux->fetcher_cond, demux->fetcher_lock);
gst_hls_demux_stop_fetcher_locked (demux, FALSE);
if (!demux->fetcher_error && gst_adapter_available (demux->download)) {
GST_INFO_OBJECT (demux, "URI fetched successfully");
bret = TRUE;
}
goto quit;
uri_error:
{
GST_ELEMENT_ERROR (demux, RESOURCE, OPEN_READ,
("Could not create an element to fetch the given URI."), ("URI: \"%s\"",
uri));
bret = FALSE;
goto quit;
}
state_change_error:
{
GST_ELEMENT_ERROR (demux, CORE, STATE_CHANGE,
("Error changing state of the fetcher element."), (NULL));
bret = FALSE;
goto quit;
}
quit:
{
/* Unlock any other fetcher that might be waiting */
g_cond_broadcast (demux->fetcher_cond);
g_mutex_unlock (demux->fetcher_lock);
return bret;
}
}
static GstFlowReturn
gst_raw_parse_chain (GstPad * pad, GstBuffer * buffer)
{
GstRawParse *rp = GST_RAW_PARSE (gst_pad_get_parent (pad));
GstFlowReturn ret = GST_FLOW_OK;
GstRawParseClass *rp_class = GST_RAW_PARSE_GET_CLASS (rp);
guint buffersize;
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT))) {
GST_DEBUG_OBJECT (rp, "received DISCONT buffer");
gst_adapter_clear (rp->adapter);
rp->discont = TRUE;
}
if (!gst_raw_parse_set_src_caps (rp))
goto no_caps;
gst_adapter_push (rp->adapter, buffer);
if (rp_class->multiple_frames_per_buffer) {
buffersize = gst_adapter_available (rp->adapter);
buffersize -= buffersize % rp->framesize;
} else {
buffersize = rp->framesize;
}
while (gst_adapter_available (rp->adapter) >= buffersize) {
buffer = gst_adapter_take_buffer (rp->adapter, buffersize);
ret = gst_raw_parse_push_buffer (rp, buffer);
if (ret != GST_FLOW_OK)
break;
}
done:
gst_object_unref (rp);
return ret;
/* ERRORS */
no_caps:
{
GST_ERROR_OBJECT (rp, "could not set caps");
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
}
static gint
gst_vdp_h264_dec_scan_for_sync (GstBaseVideoDecoder * base_video_decoder,
GstAdapter * adapter)
{
GstVdpH264Dec *h264_dec = GST_VDP_H264_DEC (base_video_decoder);
gint m;
if (h264_dec->packetized)
return 0;
m = gst_adapter_masked_scan_uint32 (adapter, 0xffffff00, 0x00000100,
0, gst_adapter_available (adapter));
if (m == -1)
return gst_adapter_available (adapter) - SYNC_CODE_SIZE;
return m;
}
static GstFlowReturn
gst_rtp_mpv_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRTPMPVPay *rtpmpvpay;
guint avail, packet_len;
GstClockTime timestamp, duration;
GstFlowReturn ret = GST_FLOW_OK;
rtpmpvpay = GST_RTP_MPV_PAY (basepayload);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
avail = gst_adapter_available (rtpmpvpay->adapter);
if (duration == -1)
duration = 0;
if (rtpmpvpay->first_ts == GST_CLOCK_TIME_NONE || avail == 0)
rtpmpvpay->first_ts = timestamp;
if (avail == 0) {
rtpmpvpay->duration = duration;
} else {
rtpmpvpay->duration += duration;
}
gst_adapter_push (rtpmpvpay->adapter, buffer);
avail = gst_adapter_available (rtpmpvpay->adapter);
/* get packet length of previous data and this new data,
* payload length includes a 4 byte MPEG video-specific header */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 4, 0);
GST_LOG_OBJECT (rtpmpvpay, "available %d, rtp packet length %d", avail,
packet_len);
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpmpvpay->duration)) {
ret = gst_rtp_mpv_pay_flush (rtpmpvpay);
} else {
rtpmpvpay->first_ts = timestamp;
}
return ret;
}
static gboolean
gst_jpeg_parse_sink_event (GstPad * pad, GstEvent * event)
{
GstJpegParse *parse;
gboolean res = TRUE;
parse = GST_JPEG_PARSE (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (parse, "event : %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
parse->priv->next_ts = GST_CLOCK_TIME_NONE;
parse->priv->duration = GST_CLOCK_TIME_NONE;
parse->priv->last_offset = 0;
parse->priv->last_entropy_len = 0;
parse->priv->last_resync = FALSE;
gst_adapter_clear (parse->priv->adapter);
break;
case GST_EVENT_EOS:{
/* Push the remaining data, even though it's incomplete */
guint available = gst_adapter_available (parse->priv->adapter);
if (available > 0)
gst_jpeg_parse_push_buffer (parse, available);
res = gst_pad_push_event (parse->priv->srcpad, event);
break;
}
case GST_EVENT_NEWSEGMENT:
/* Discard any data in the adapter. There should have been an EOS before
* to flush it. */
gst_adapter_clear (parse->priv->adapter);
res = gst_pad_push_event (parse->priv->srcpad, event);
parse->priv->new_segment = TRUE;
break;
case GST_EVENT_TAG:{
if (!parse->priv->new_segment)
res = gst_pad_event_default (pad, event);
else {
GstTagList *taglist = NULL;
gst_event_parse_tag (event, &taglist);
/* Hold on to the tags till the srcpad caps are definitely set */
gst_tag_list_insert (get_tag_list (parse), taglist,
GST_TAG_MERGE_REPLACE);
GST_DEBUG ("collected tags: %" GST_PTR_FORMAT, parse->priv->tags);
gst_event_unref (event);
}
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (parse);
return res;
}
static gboolean
gst_aiffparse_peek_data (AIFFParse * aiff, guint32 size, const guint8 ** data)
{
if (gst_adapter_available (aiff->adapter) < size)
return FALSE;
*data = gst_adapter_peek (aiff->adapter, size);
return TRUE;
}
static GstFlowReturn
gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
guint available;
guint max_payload;
GstBuffer *outbuf;
guint8 *payload_data;
guint frame_count;
guint payload_length;
struct rtp_payload *payload;
if (sbcpay->frame_length == 0) {
GST_ERROR_OBJECT (sbcpay, "Frame length is 0");
return GST_FLOW_ERROR;
}
available = gst_adapter_available (sbcpay->adapter);
max_payload =
gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (sbcpay) -
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
max_payload = MIN (max_payload, available);
frame_count = max_payload / sbcpay->frame_length;
payload_length = frame_count * sbcpay->frame_length;
if (payload_length == 0) /* Nothing to send */
return GST_FLOW_OK;
outbuf = gst_rtp_buffer_new_allocate (payload_length +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
/* get payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (sbcpay));
/* write header and copy data into payload */
payload_data = gst_rtp_buffer_get_payload (&rtp);
payload = (struct rtp_payload *) payload_data;
memset (payload, 0, sizeof (struct rtp_payload));
payload->frame_count = frame_count;
gst_adapter_copy (sbcpay->adapter, payload_data +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, payload_length);
gst_rtp_buffer_unmap (&rtp);
gst_adapter_flush (sbcpay->adapter, payload_length);
/* FIXME: what about duration? */
GST_BUFFER_PTS (outbuf) = sbcpay->timestamp;
GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", payload_length);
return gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (sbcpay), outbuf);
}
static GstFlowReturn
gst_wildmidi_parse_song (GstWildmidi * wildmidi)
{
struct _WM_Info *info;
GstCaps *outcaps;
guint8 *data;
guint size;
GST_DEBUG_OBJECT (wildmidi, "Parsing song");
size = gst_adapter_available (wildmidi->adapter);
data = gst_adapter_take (wildmidi->adapter, size);
/* this method takes our memory block */
GST_OBJECT_LOCK (wildmidi);
wildmidi->song = WildMidi_OpenBuffer (data, size);
if (!wildmidi->song)
goto open_failed;
WildMidi_LoadSamples (wildmidi->song);
WildMidi_SetOption (wildmidi->song, WM_MO_LINEAR_VOLUME,
wildmidi->linear_volume);
WildMidi_SetOption (wildmidi->song, WM_MO_EXPENSIVE_INTERPOLATION,
wildmidi->high_quality);
info = WildMidi_GetInfo (wildmidi->song);
GST_OBJECT_UNLOCK (wildmidi);
wildmidi->o_len = info->approx_total_samples;
outcaps = gst_caps_copy (gst_pad_get_pad_template_caps (wildmidi->srcpad));
gst_pad_set_caps (wildmidi->srcpad, outcaps);
gst_caps_unref (outcaps);
/* we keep an internal segment in samples */
gst_segment_set_newsegment (wildmidi->o_segment, FALSE, 1.0,
GST_FORMAT_DEFAULT, 0, GST_CLOCK_TIME_NONE, 0);
gst_pad_push_event (wildmidi->srcpad,
gst_wildmidi_get_new_segment_event (wildmidi, GST_FORMAT_TIME));
GST_DEBUG_OBJECT (wildmidi, "Parsing song done");
return GST_FLOW_OK;
/* ERRORS */
open_failed:
{
GST_OBJECT_UNLOCK (wildmidi);
GST_ELEMENT_ERROR (wildmidi, STREAM, DECODE, (NULL),
("Unable to parse midi data"));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRTPMP2TPay *rtpmp2tpay;
guint size, avail, packet_len;
GstClockTime timestamp, duration;
GstFlowReturn ret;
rtpmp2tpay = GST_RTP_MP2T_PAY (basepayload);
size = gst_buffer_get_size (buffer);
timestamp = GST_BUFFER_PTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
again:
ret = GST_FLOW_OK;
avail = gst_adapter_available (rtpmp2tpay->adapter);
/* Initialize new RTP payload */
if (avail == 0) {
rtpmp2tpay->first_ts = timestamp;
rtpmp2tpay->duration = duration;
}
/* get packet length of previous data and this new data */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we have,
* or if upstream is handing us several packets, to keep latency low */
if (!size || gst_rtp_base_payload_is_filled (basepayload,
packet_len, rtpmp2tpay->duration + duration)) {
ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay);
rtpmp2tpay->first_ts = timestamp;
rtpmp2tpay->duration = duration;
/* keep filling the payload */
} else {
if (GST_CLOCK_TIME_IS_VALID (duration))
rtpmp2tpay->duration += duration;
}
/* copy buffer to adapter */
if (buffer) {
gst_adapter_push (rtpmp2tpay->adapter, buffer);
buffer = NULL;
}
if (size >= (188 * 2)) {
size = 0;
goto again;
}
return ret;
}
static GstFlowReturn
gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
{
GstBuffer *outbuf;
GstFlowReturn ret;
guint avail;
guint8 *payload;
avail = gst_adapter_available (rtpmpvpay->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint packet_len;
guint payload_len;
packet_len = gst_rtp_buffer_calc_packet_len (avail, 4, 0);
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay));
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 4, 0);
outbuf = gst_rtp_buffer_new_allocate (payload_len, 4, 0);
payload = gst_rtp_buffer_get_payload (outbuf);
/* enable MPEG Video-specific header
*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ |T| TR | |N|S|B|E| P | | BFC | | FFC |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* AN FBV FFV
*/
/* fill in the MPEG Video-specific header
* data is set to 0x0 here
*/
memset (payload, 0x0, 4);
gst_adapter_copy (rtpmpvpay->adapter, payload + 4, 0, payload_len);
gst_adapter_flush (rtpmpvpay->adapter, payload_len);
avail -= payload_len;
gst_rtp_buffer_set_marker (outbuf, avail == 0);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_real_audio_demux_parse_data (GstRealAudioDemux * demux)
{
GstFlowReturn ret = GST_FLOW_OK;
guint avail, unit_size;
avail = gst_adapter_available (demux->adapter);
if (demux->packet_size > 0)
unit_size = demux->packet_size;
else
unit_size = avail & 0xfffffff0; /* round down to next multiple of 16 */
GST_LOG_OBJECT (demux, "available = %u, unit_size = %u", avail, unit_size);
while (ret == GST_FLOW_OK && unit_size > 0 && avail >= unit_size) {
GstClockTime ts;
const guint8 *data;
GstBuffer *buf = NULL;
buf = gst_buffer_new_and_alloc (unit_size);
gst_buffer_set_caps (buf, GST_PAD_CAPS (demux->srcpad));
data = gst_adapter_peek (demux->adapter, unit_size);
memcpy (GST_BUFFER_DATA (buf), data, unit_size);
gst_adapter_flush (demux->adapter, unit_size);
avail -= unit_size;
if (demux->need_newsegment) {
gst_pad_push_event (demux->srcpad,
gst_event_new_new_segment_full (FALSE, demux->segment.rate,
demux->segment.applied_rate, GST_FORMAT_TIME,
demux->segment.start, demux->segment.stop, demux->segment.time));
demux->need_newsegment = FALSE;
}
if (demux->pending_tags) {
gst_element_found_tags_for_pad (GST_ELEMENT (demux), demux->srcpad,
demux->pending_tags);
demux->pending_tags = NULL;
}
if (demux->fourcc == GST_RM_AUD_DNET) {
buf = gst_rm_utils_descramble_dnet_buffer (buf);
}
ts = gst_real_demux_get_timestamp_from_offset (demux, demux->offset);
GST_BUFFER_TIMESTAMP (buf) = ts;
gst_segment_set_last_stop (&demux->segment, GST_FORMAT_TIME, ts);
ret = gst_pad_push (demux->srcpad, buf);
}
return ret;
}
void
gst_base_video_decoder_add_to_frame (GstBaseVideoDecoder * base_video_decoder,
int n_bytes)
{
GstBuffer *buf;
GST_DEBUG ("add to frame");
if (n_bytes == 0)
return;
if (gst_adapter_available (base_video_decoder->output_adapter) == 0) {
base_video_decoder->frame_offset = base_video_decoder->input_offset -
gst_adapter_available (base_video_decoder->input_adapter);
}
buf = gst_adapter_take_buffer (base_video_decoder->input_adapter, n_bytes);
gst_adapter_push (base_video_decoder->output_adapter, buf);
}
static gboolean
gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
const GValue *headers;
GstFlacDec *flacdec;
GstStructure *s;
guint i, num;
flacdec = GST_FLAC_DEC (dec);
GST_LOG_OBJECT (dec, "sink caps: %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
headers = gst_structure_get_value (s, "streamheader");
if (headers == NULL || !GST_VALUE_HOLDS_ARRAY (headers)) {
GST_WARNING_OBJECT (dec, "no 'streamheader' field in input caps, try "
"adding a flacparse element upstream");
return FALSE;
}
if (gst_adapter_available (flacdec->adapter) > 0) {
GST_WARNING_OBJECT (dec, "unexpected data left in adapter");
gst_adapter_clear (flacdec->adapter);
}
num = gst_value_array_get_size (headers);
for (i = 0; i < num; ++i) {
const GValue *header_val;
GstBuffer *header_buf;
header_val = gst_value_array_get_value (headers, i);
if (header_val == NULL || !GST_VALUE_HOLDS_BUFFER (header_val))
return FALSE;
header_buf = g_value_dup_boxed (header_val);
GST_INFO_OBJECT (dec, "pushing header buffer of %" G_GSIZE_FORMAT " bytes "
"into adapter", gst_buffer_get_size (header_buf));
gst_adapter_push (flacdec->adapter, header_buf);
}
GST_DEBUG_OBJECT (dec, "Processing headers and metadata");
if (!FLAC__stream_decoder_process_until_end_of_metadata (flacdec->decoder)) {
GST_WARNING_OBJECT (dec, "process_until_end_of_metadata failed");
if (FLAC__stream_decoder_get_state (flacdec->decoder) ==
FLAC__STREAM_DECODER_ABORTED) {
GST_WARNING_OBJECT (flacdec, "Read callback caused internal abort");
/* allow recovery */
gst_adapter_clear (flacdec->adapter);
FLAC__stream_decoder_flush (flacdec->decoder);
gst_flac_dec_handle_decoder_error (flacdec, TRUE);
}
}
GST_INFO_OBJECT (dec, "headers and metadata are now processed");
return TRUE;
}
static GstFlowReturn
gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf)
{
GstFlacDec *dec;
dec = GST_FLAC_DEC (audio_dec);
/* drain remaining data? */
if (G_UNLIKELY (buf == NULL)) {
gst_flac_dec_flush (audio_dec, FALSE);
return GST_FLOW_OK;
}
GST_LOG_OBJECT (dec, "frame: ts %" GST_TIME_FORMAT ", flags 0x%04x, "
"%" G_GSIZE_FORMAT " bytes", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_BUFFER_FLAGS (buf), gst_buffer_get_size (buf));
/* drop any in-stream headers, we've processed those in set_format already */
if (G_UNLIKELY (!dec->got_headers)) {
gboolean got_audio_frame;
GstMapInfo map;
/* check if this is a flac audio frame (rather than a header or junk) */
gst_buffer_map (buf, &map, GST_MAP_READ);
got_audio_frame =
gst_flac_dec_scan_got_frame (dec, map.data, map.size, NULL);
gst_buffer_unmap (buf, &map);
if (!got_audio_frame) {
GST_INFO_OBJECT (dec, "dropping in-stream header, %" G_GSIZE_FORMAT " "
"bytes", map.size);
gst_audio_decoder_finish_frame (audio_dec, NULL, 1);
return GST_FLOW_OK;
}
GST_INFO_OBJECT (dec, "first audio frame, got all in-stream headers now");
dec->got_headers = TRUE;
}
gst_adapter_push (dec->adapter, gst_buffer_ref (buf));
buf = NULL;
dec->last_flow = GST_FLOW_OK;
/* framed - there should always be enough data to decode something */
GST_LOG_OBJECT (dec, "%" G_GSIZE_FORMAT " bytes available",
gst_adapter_available (dec->adapter));
if (!FLAC__stream_decoder_process_single (dec->decoder)) {
GST_INFO_OBJECT (dec, "process_single failed");
}
return dec->last_flow;
}
static void
create_fingerprint (GstOFA * ofa)
{
GstBuffer *buf;
GstAudioFilter *ofa_filter = GST_AUDIO_FILTER (ofa);
gint rate = ofa_filter->format.rate;
gint channels = ofa_filter->format.channels;
gint endianness =
ofa_filter->format.bigend ? OFA_BIG_ENDIAN : OFA_LITTLE_ENDIAN;
GstTagList *tags;
guint available;
available = gst_adapter_available (ofa->adapter);
if (available == 0) {
GST_WARNING_OBJECT (ofa, "No data to take fingerprint from");
ofa->record = FALSE;
return;
}
if (GST_AUDIO_FILTER (ofa)->format.bigend)
endianness = OFA_BIG_ENDIAN;
else
endianness = OFA_LITTLE_ENDIAN;
GST_DEBUG_OBJECT (ofa, "Generating fingerprint for %u samples",
available / 2);
buf = gst_adapter_take_buffer (ofa->adapter, available);
ofa->fingerprint = g_strdup (ofa_create_print (GST_BUFFER_DATA (buf),
endianness, GST_BUFFER_SIZE (buf) / 2, rate,
(channels == 2) ? 1 : 0));
if (ofa->fingerprint) {
GST_INFO_OBJECT (ofa, "Generated fingerprint: %s", ofa->fingerprint);
} else {
GST_WARNING_OBJECT (ofa, "Failed to generate fingerprint");
}
gst_buffer_unref (buf);
if (ofa->fingerprint) {
tags = gst_tag_list_new ();
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
GST_TAG_OFA_FINGERPRINT, ofa->fingerprint, NULL);
gst_element_found_tags (GST_ELEMENT (ofa), tags);
g_object_notify (G_OBJECT (ofa), "fingerprint");
}
ofa->record = FALSE;
}
static GstFlowReturn
gst_chop_my_data_process (GstChopMyData * chopmydata, gboolean flush)
{
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *buffer;
if (chopmydata->next_size == 0) {
get_next_size (chopmydata);
}
while (gst_adapter_available (chopmydata->adapter) >= chopmydata->next_size) {
buffer =
gst_adapter_take_buffer (chopmydata->adapter, chopmydata->next_size);
GST_BUFFER_PTS (buffer) = gst_adapter_prev_pts (chopmydata->adapter, NULL);
GST_BUFFER_DTS (buffer) = gst_adapter_prev_dts (chopmydata->adapter, NULL);
chopmydata->next_size = 0;
ret = gst_pad_push (chopmydata->srcpad, buffer);
if (ret != GST_FLOW_OK) {
return ret;
}
get_next_size (chopmydata);
}
if (flush) {
guint min_size = chopmydata->min_size;
while (gst_adapter_available (chopmydata->adapter) >= min_size) {
buffer = gst_adapter_take_buffer (chopmydata->adapter, min_size);
ret = gst_pad_push (chopmydata->srcpad, buffer);
if (ret != GST_FLOW_OK)
break;
}
gst_adapter_clear (chopmydata->adapter);
}
return ret;
}
static GstFlowReturn
gst_webrtc_echo_probe_transform_ip (GstBaseTransform * btrans,
GstBuffer * buffer)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (btrans);
GstBuffer *newbuf = NULL;
GST_WEBRTC_ECHO_PROBE_LOCK (self);
newbuf = gst_buffer_copy (buffer);
/* Moves the buffer timestamp to be in Running time */
GST_BUFFER_PTS (newbuf) = gst_segment_to_running_time (&btrans->segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (buffer));
gst_adapter_push (self->adapter, newbuf);
if (gst_adapter_available (self->adapter) > MAX_ADAPTER_SIZE)
gst_adapter_flush (self->adapter,
gst_adapter_available (self->adapter) - MAX_ADAPTER_SIZE);
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return GST_FLOW_OK;
}
static gboolean
gst_hls_demux_get_next_fragment (GstHLSDemux * demux, gboolean retry)
{
GstBuffer *buf;
guint avail;
const gchar *next_fragment_uri;
GstClockTime duration;
GstClockTime timestamp;
gboolean discont;
if (!gst_m3u8_client_get_next_fragment (demux->client, &discont,
&next_fragment_uri, &duration, ×tamp)) {
GST_INFO_OBJECT (demux, "This playlist doesn't contain more fragments");
demux->end_of_playlist = TRUE;
gst_task_start (demux->task);
return FALSE;
}
GST_INFO_OBJECT (demux, "Fetching next fragment %s", next_fragment_uri);
if (!gst_hls_demux_fetch_location (demux, next_fragment_uri))
return FALSE;
avail = gst_adapter_available (demux->download);
buf = gst_adapter_take_buffer (demux->download, avail);
GST_BUFFER_DURATION (buf) = duration;
GST_BUFFER_TIMESTAMP (buf) = timestamp;
/* We actually need to do this every time we switch bitrate */
if (G_UNLIKELY (demux->do_typefind)) {
GstCaps *caps = gst_type_find_helper_for_buffer (NULL, buf, NULL);
if (!demux->input_caps || !gst_caps_is_equal (caps, demux->input_caps)) {
gst_caps_replace (&demux->input_caps, caps);
/* gst_pad_set_caps (demux->srcpad, demux->input_caps); */
GST_INFO_OBJECT (demux, "Input source caps: %" GST_PTR_FORMAT,
demux->input_caps);
demux->do_typefind = FALSE;
} else
gst_caps_unref (caps);
}
gst_buffer_set_caps (buf, demux->input_caps);
if (discont) {
GST_DEBUG_OBJECT (demux, "Marking fragment as discontinuous");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
}
g_queue_push_tail (demux->queue, buf);
gst_task_start (demux->task);
gst_adapter_clear (demux->download);
return TRUE;
}
static GstFlowReturn
gst_dvd_sub_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstDvdSubParse *parse = GST_DVD_SUB_PARSE (parent);
GstAdapter *adapter;
GstBuffer *outbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
adapter = parse->adapter;
GST_LOG_OBJECT (parse, "%" G_GSIZE_FORMAT " bytes, ts: %" GST_TIME_FORMAT,
gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
gst_adapter_push (adapter, buf);
if (!parse->needed) {
guint8 data[2];
gst_adapter_copy (adapter, data, 0, 2);
parse->needed = GST_READ_UINT16_BE (data);
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
if (GST_CLOCK_TIME_IS_VALID (parse->stamp))
/* normally, we expect only the first fragment to carry a timestamp */
GST_WARNING_OBJECT (parse, "Received more timestamps than expected.");
else
parse->stamp = GST_BUFFER_TIMESTAMP (buf);
}
if (parse->needed) {
guint av;
av = gst_adapter_available (adapter);
if (av >= parse->needed) {
if (av > parse->needed) {
/* normally, we expect several fragment, boundary aligned */
GST_WARNING_OBJECT (parse, "Unexpected: needed %d, "
"but more (%d) is available.", parse->needed, av);
}
outbuf = gst_adapter_take_buffer (adapter, parse->needed);
/* decorate buffer */
GST_BUFFER_TIMESTAMP (outbuf) = parse->stamp;
/* reset state */
parse->stamp = GST_CLOCK_TIME_NONE;
parse->needed = 0;
/* and send along */
ret = gst_pad_push (parse->srcpad, outbuf);
}
}
return ret;
}
static GstFlowReturn
gst_flite_test_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstFliteTestSrc *src;
int n_bytes;
src = GST_FLITE_TEST_SRC (basesrc);
n_bytes = src->info.channels * sizeof (gint16) * src->samples_per_buffer;
while (gst_adapter_available (src->adapter) < n_bytes) {
GstBuffer *buf;
char *text;
int i;
GstMapInfo map;
gint16 *data;
cst_wave *wave;
gsize size;
text = get_channel_name (src, src->channel);
wave = flite_text_to_wave (text, src->voice);
g_free (text);
cst_wave_resample (wave, src->info.rate);
GST_DEBUG ("type %s, sample_rate %d, num_samples %d, num_channels %d",
wave->type, wave->sample_rate, wave->num_samples, wave->num_channels);
size = src->info.channels * sizeof (gint16) * wave->num_samples;
buf = gst_buffer_new_and_alloc (size);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
data = (gint16 *) map.data;
memset (data, 0, size);
for (i = 0; i < wave->num_samples; i++) {
data[i * src->info.channels + src->channel] = wave->samples[i];
}
gst_buffer_unmap (buf, &map);
src->channel++;
if (src->channel == src->info.channels) {
src->channel = 0;
}
gst_adapter_push (src->adapter, buf);
}
*buffer = gst_adapter_take_buffer (src->adapter, n_bytes);
return GST_FLOW_OK;
}
void
gst_base_video_parse_lost_sync (GstBaseVideoParse * base_video_parse)
{
g_return_if_fail (GST_IS_BASE_VIDEO_PARSE (base_video_parse));
GST_DEBUG ("lost_sync");
if (gst_adapter_available (base_video_parse->input_adapter) >= 1) {
gst_adapter_flush (base_video_parse->input_adapter, 1);
}
base_video_parse->have_sync = FALSE;
}
void
gst_base_video_decoder_lost_sync (GstBaseVideoDecoder * base_video_decoder)
{
g_return_if_fail (GST_IS_BASE_VIDEO_DECODER (base_video_decoder));
GST_DEBUG ("lost_sync");
if (gst_adapter_available (base_video_decoder->input_adapter) >= 1) {
gst_adapter_flush (base_video_decoder->input_adapter, 1);
}
base_video_decoder->have_sync = FALSE;
}
static GstFlowReturn sbc_enc_chain(GstPad *pad, GstBuffer *buffer)
{
GstSbcEnc *enc = GST_SBC_ENC(gst_pad_get_parent(pad));
GstAdapter *adapter = enc->adapter;
GstFlowReturn res = GST_FLOW_OK;
gst_adapter_push(adapter, buffer);
while (gst_adapter_available(adapter) >= enc->codesize &&
res == GST_FLOW_OK) {
GstBuffer *output;
GstCaps *caps;
const guint8 *data;
gint consumed;
caps = GST_PAD_CAPS(enc->srcpad);
res = gst_pad_alloc_buffer_and_set_caps(enc->srcpad,
GST_BUFFER_OFFSET_NONE,
enc->frame_length, caps,
&output);
if (res != GST_FLOW_OK)
goto done;
data = gst_adapter_peek(adapter, enc->codesize);
consumed = sbc_encode(&enc->sbc, (gpointer) data,
enc->codesize,
GST_BUFFER_DATA(output),
GST_BUFFER_SIZE(output), NULL);
if (consumed <= 0) {
GST_DEBUG_OBJECT(enc, "comsumed < 0, codesize: %d",
enc->codesize);
break;
}
gst_adapter_flush(adapter, consumed);
GST_BUFFER_TIMESTAMP(output) = GST_BUFFER_TIMESTAMP(buffer);
/* we have only 1 frame */
GST_BUFFER_DURATION(output) = enc->frame_duration;
res = gst_pad_push(enc->srcpad, output);
if (res != GST_FLOW_OK)
goto done;
}
done:
gst_object_unref(enc);
return res;
}
static GstFlowReturn
gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
{
guint avail, mtu;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
avail = gst_adapter_available (rtpmp2tpay->adapter);
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp2tpay);
while (avail > 0 && (ret == GST_FLOW_OK)) {
guint towrite;
guint payload_len;
guint packet_len;
GstBuffer *paybuf;
/* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, mtu);
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
payload_len -= payload_len % 188;
/* need whole packets */
if (!payload_len)
break;
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
/* get payload */
paybuf = gst_adapter_take_buffer_fast (rtpmp2tpay->adapter, payload_len);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp2tpay), outbuf, paybuf, 0);
outbuf = gst_buffer_append (outbuf, paybuf);
avail -= payload_len;
GST_BUFFER_PTS (outbuf) = rtpmp2tpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %u",
(guint) gst_buffer_get_size (outbuf));
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp2tpay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to split the MP4V data
* over multiple packets. */
avail = gst_adapter_available (rtpmp4vpay->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmp4vpay));
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (rtpmp4vpay->adapter, payload, 0, payload_len);
gst_adapter_flush (rtpmp4vpay->adapter, payload_len);
avail -= payload_len;
gst_rtp_buffer_set_marker (outbuf, avail == 0);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4vpay->first_timestamp;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
}
return ret;
}
long
AudioStream::read_bytes (void *data, long length)
{
gssize len = gst_adapter_available (adapter);
if (len < length)
length = len;
GST_LOG ("Reading %d bytes from adapter", (int) length);
gst_adapter_copy (adapter, data, 0, length);
gst_adapter_flush (adapter, length);
return length;
}
static GstFlowReturn
gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpMPAPay *rtpmpapay;
GstFlowReturn ret;
guint size, avail;
guint packet_len;
GstClockTime duration, timestamp;
rtpmpapay = GST_RTP_MPA_PAY (basepayload);
size = gst_buffer_get_size (buffer);
duration = GST_BUFFER_DURATION (buffer);
timestamp = GST_BUFFER_PTS (buffer);
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (rtpmpapay, "DISCONT");
gst_rtp_mpa_pay_reset (rtpmpapay);
}
avail = gst_adapter_available (rtpmpapay->adapter);
/* get packet length of previous data and this new data,
* payload length includes a 4 byte header */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_rtp_base_payload_is_filled (basepayload,
packet_len, rtpmpapay->duration + duration)) {
ret = gst_rtp_mpa_pay_flush (rtpmpapay);
avail = 0;
} else {
ret = GST_FLOW_OK;
}
if (avail == 0) {
GST_DEBUG_OBJECT (rtpmpapay,
"first packet, save timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
rtpmpapay->first_ts = timestamp;
rtpmpapay->duration = 0;
}
gst_adapter_push (rtpmpapay->adapter, buffer);
rtpmpapay->duration = duration;
return ret;
}
static int
gst_ffmpeg_pipe_read (URLContext * h, unsigned char *buf, int size)
{
GstFFMpegPipe *ffpipe;
const guint8 *data;
guint available;
ffpipe = (GstFFMpegPipe *) h->priv_data;
GST_LOG ("requested size %d", size);
GST_FFMPEG_PIPE_MUTEX_LOCK (ffpipe);
GST_LOG ("requested size %d", size);
while ((available = gst_adapter_available (ffpipe->adapter)) < size
&& !ffpipe->eos) {
GST_DEBUG ("Available:%d, requested:%d", available, size);
ffpipe->needed = size;
GST_FFMPEG_PIPE_SIGNAL (ffpipe);
GST_FFMPEG_PIPE_WAIT (ffpipe);
}
size = MIN (available, size);
if (size) {
GST_LOG ("Getting %d bytes", size);
data = gst_adapter_peek (ffpipe->adapter, size);
memcpy (buf, data, size);
gst_adapter_flush (ffpipe->adapter, size);
GST_LOG ("%d bytes left in adapter",
gst_adapter_available (ffpipe->adapter));
ffpipe->needed = 0;
}
GST_FFMPEG_PIPE_MUTEX_UNLOCK (ffpipe);
return size;
}
static void
gst_icydemux_parse_and_send_tags (GstICYDemux * icydemux)
{
GstTagList *tags;
const guint8 *data;
int length, i;
gchar *buffer;
gchar **strings;
length = gst_adapter_available (icydemux->meta_adapter);
data = gst_adapter_peek (icydemux->meta_adapter, length);
/* Now, copy this to a buffer where we can NULL-terminate it to make things
* a bit easier, then do that parsing. */
buffer = g_strndup ((const gchar *) data, length);
tags = gst_tag_list_new ();
strings = g_strsplit (buffer, "';", 0);
for (i = 0; strings[i]; i++) {
if (!g_ascii_strncasecmp (strings[i], "StreamTitle=", 12)) {
char *title = gst_icydemux_unicodify (strings[i] + 13);
if (title && *title) {
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_TITLE,
title, NULL);
g_free (title);
}
} else if (!g_ascii_strncasecmp (strings[i], "StreamUrl=", 10)) {
char *url = gst_icydemux_unicodify (strings[i] + 11);
if (url && *url) {
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_HOMEPAGE,
url, NULL);
g_free (url);
}
}
}
g_strfreev (strings);
g_free (buffer);
gst_adapter_clear (icydemux->meta_adapter);
if (!gst_tag_list_is_empty (tags))
gst_icydemux_tag_found (icydemux, tags);
else
gst_tag_list_free (tags);
}