headers prepended during raw to ADTS
conversion */
-#ifdef TIZEN_FEATURE_AACPARSE_MODIFICATION /* to get more accurate duration */
-#define AAC_MAX_ESTIMATE_DURATION_BUF (1024 * 1024) /* use first 1 Mbyte */
+#ifdef TIZEN_FEATURE_AACPARSE_MODIFICATION /* to get more accurate duration */
+#define AAC_MAX_ESTIMATE_DURATION_BUF (1024 * 1024) /* use first 1 Mbyte */
#define AAC_SAMPLE_PER_FRAME 1024
-#define AAC_MAX_PULL_RANGE_BUF (1 * 1024 * 1024) /* 1 MByte */
-#define AAC_LARGE_FILE_SIZE (2 * 1024 * 1024) /* 2 MByte */
+#define AAC_MAX_PULL_RANGE_BUF (1 * 1024 * 1024) /* 1 MByte */
+#define AAC_LARGE_FILE_SIZE (2 * 1024 * 1024) /* 2 MByte */
#define gst_aac_parse_parent_class parent_class
#endif
#ifdef TIZEN_FEATURE_AACPARSE_MODIFICATION
static guint gst_aac_parse_adts_get_fast_frame_len (const guint8 * data);
-static gboolean gst_aac_parse_src_eventfunc(GstBaseParse * parse,
+static gboolean gst_aac_parse_src_eventfunc (GstBaseParse * parse,
GstEvent * event);
/* make full aac(adts) index table when seek */
static gboolean gst_aac_parse_adts_src_eventfunc (GstBaseParse * parse,
GstEvent * event);
-int get_aac_parse_get_adts_frame_length (const unsigned char* data,
+int get_aac_parse_get_adts_frame_length (const unsigned char *data,
gint64 offset);
static gboolean gst_aac_parse_estimate_duration (GstBaseParse * parse);
#endif
gboolean ret = FALSE;
aacparse->first_frame = FALSE;
- ret = gst_aac_parse_estimate_duration(parse);
+ ret = gst_aac_parse_estimate_duration (parse);
if (!ret) {
GST_WARNING_OBJECT (aacparse, "can not estimate total duration");
ret = GST_FLOW_NOT_SUPPORTED;
}
}
#ifdef TIZEN_FEATURE_AACPARSE_MODIFICATION
- else if(aacparse->header_type == DSPAAC_HEADER_ADIF)
- {
- /* to get more correct duration */
- int file_size = 0;
- float estimated_duration = 0;
- gint64 total_file_size;
- gst_base_parse_get_upstream_size(parse, &total_file_size);
- estimated_duration = ((total_file_size * 8) / (float)(aacparse->bitrate * 1000))* GST_SECOND;
- gst_base_parse_set_duration (parse, GST_FORMAT_TIME, estimated_duration * 1000, 0);
+ else if (aacparse->header_type == DSPAAC_HEADER_ADIF) {
+ /* to get more correct duration */
+ float estimated_duration = 0;
+ gint64 total_file_size;
+ gst_base_parse_get_upstream_size (parse, &total_file_size);
+ estimated_duration =
+ ((total_file_size * 8) / (float) (aacparse->bitrate * 1000)) *
+ GST_SECOND;
+ gst_base_parse_set_duration (parse, GST_FORMAT_TIME,
+ estimated_duration * 1000, 0);
}
#endif
*
* Returns: frame size
*/
-int get_aac_parse_get_adts_frame_length (const unsigned char* data,
- gint64 offset)
+int
+get_aac_parse_get_adts_frame_length (const unsigned char *data, gint64 offset)
{
const gint adts_header_length_no_crc = 7;
const gint adts_header_length_with_crc = 9;
gint head_size;
/* check of syncword */
- if ((data[offset+0] != 0xff) || ((data[offset+1] & 0xf6) != 0xf0)) {
- GST_ERROR("check sync word is fail\n");
- return -1;
+ if ((data[offset + 0] != 0xff) || ((data[offset + 1] & 0xf6) != 0xf0)) {
+ GST_ERROR ("check sync word is fail\n");
+ return -1;
}
/* check of protection absent */
- protection_absent = (data[offset+1] & 0x01);
+ protection_absent = (data[offset + 1] & 0x01);
/*check of frame length */
- frame_size = (data[offset+3] & 0x3) << 11 | data[offset+4] << 3 | data[offset+5] >> 5;
+ frame_size =
+ (data[offset + 3] & 0x3) << 11 | data[offset + 4] << 3 | data[offset +
+ 5] >> 5;
/* check of header size */
/* protectionAbsent is 0 if there is CRC */
- head_size = protection_absent ? adts_header_length_no_crc : adts_header_length_with_crc;
+ head_size =
+ protection_absent ? adts_header_length_no_crc :
+ adts_header_length_with_crc;
if (head_size > frame_size) {
- GST_ERROR("return frame length as 0 (frameSize %u < headSize %u)", frame_size, head_size);
+ GST_ERROR ("return frame length as 0 (frameSize %u < headSize %u)",
+ frame_size, head_size);
return 0;
}
{
gboolean ret = FALSE;
GstFlowReturn res = GST_FLOW_OK;
- gint64 pull_size = 0, file_size = 0, offset = 0, num_frames=0, duration=0;
- guint profile = 0, sample_rate_index = 0, sample_rate = 0, channel = 0;
+ gint64 pull_size = 0, file_size = 0, offset = 0, num_frames = 0, duration = 0;
+ guint sample_rate_index = 0, sample_rate = 0, channel = 0;
guint frame_size = 0, frame_duration_us = 0, estimated_bitrate = 0;
- guint lost_sync_count=0;
+ guint lost_sync_count = 0;
GstClockTime estimated_duration = GST_CLOCK_TIME_NONE;
GstBuffer *buffer = NULL;
guint8 *buf = NULL;
GstPadMode pad_mode = GST_PAD_MODE_NONE;
GstAacParse *aacparse;
gint64 buffer_size = 0;
+ GstMapInfo map;
aacparse = GST_AAC_PARSE (parse);
GST_LOG_OBJECT (aacparse, "gst_aac_parse_estimate_duration enter");
/* check baseparse define these fuction */
- gst_base_parse_get_pad_mode(parse, &pad_mode);
+ gst_base_parse_get_pad_mode (parse, &pad_mode);
if (pad_mode != GST_PAD_MODE_PULL) {
- GST_INFO_OBJECT (aacparse, "aac parser is not pull mode. can not estimate duration");
+ GST_INFO_OBJECT (aacparse,
+ "aac parser is not pull mode. can not estimate duration");
return FALSE;
}
return FALSE;
}
- pull_size = MIN(file_size, AAC_MAX_ESTIMATE_DURATION_BUF);
+ pull_size = MIN (file_size, AAC_MAX_ESTIMATE_DURATION_BUF);
res = gst_pad_pull_range (parse->sinkpad, 0, pull_size, &buffer);
if (res != GST_FLOW_OK) {
return FALSE;
}
- GstMapInfo map;
- gst_buffer_map(buffer, &map, GST_MAP_READ);
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
buf = map.data;
buffer_size = map.size;
if (buffer_size != pull_size) {
- GST_ERROR_OBJECT(aacparse, "We got different buffer_size(%d) with pull_size(%d).",
- buffer_size, pull_size);
+ GST_ERROR_OBJECT (aacparse,
+ "We got different buffer_size(%" G_GINT64_FORMAT ") with pull_size(%"
+ G_GINT64_FORMAT ").", buffer_size, pull_size);
}
/* MODIFICATION : add defence codes for real buffer_size is different with pull_size */
- for (i = 0; i < buffer_size; i ++) {
- if ((buf[i] == 0xff) && ((buf[i+1] & 0xf6) == 0xf0)) { /* aac sync word */
- profile = (buf[i+2] >> 6) & 0x3;
- sample_rate_index = (buf[i+2] >> 2) & 0xf;
- sample_rate = gst_aac_parse_get_sample_rate_from_index(sample_rate_index);
+ for (i = 0; i < buffer_size; i++) {
+ if ((buf[i] == 0xff) && ((buf[i + 1] & 0xf6) == 0xf0)) { /* aac sync word */
+ //guint profile = (buf[i+2] >> 6) & 0x3;
+ sample_rate_index = (buf[i + 2] >> 2) & 0xf;
+ sample_rate =
+ gst_aac_parse_get_sample_rate_from_index (sample_rate_index);
if (sample_rate == 0) {
- GST_WARNING_OBJECT (aacparse, "Invalid sample rate index (0)");
- goto EXIT;
+ GST_WARNING_OBJECT (aacparse, "Invalid sample rate index (0)");
+ goto EXIT;
}
- channel = (buf[i+2] & 0x1) << 2 | (buf[i+3] >> 6);
+ channel = (buf[i + 2] & 0x1) << 2 | (buf[i + 3] >> 6);
- GST_INFO_OBJECT (aacparse, "found sync. aac fs=%d, ch=%d", sample_rate, channel);
+ GST_INFO_OBJECT (aacparse, "found sync. aac fs=%d, ch=%d", sample_rate,
+ channel);
/* count number of frames */
- /*MODIFICATION : add defence codes for real buffer_size is different with pull_size*/
+ /* MODIFICATION : add defence codes for real buffer_size is different with pull_size */
//while (offset < pull_size) {
while (offset < buffer_size) {
- frame_size = get_aac_parse_get_adts_frame_length(buf, i + offset);
- if (frame_size == 0) {
- GST_ERROR_OBJECT (aacparse, "framesize error at offset %"G_GINT64_FORMAT, offset);
+ frame_size = get_aac_parse_get_adts_frame_length (buf, i + offset);
+ if (frame_size == 0) {
+ GST_ERROR_OBJECT (aacparse,
+ "framesize error at offset %" G_GINT64_FORMAT, offset);
break;
} else if (frame_size == -1) {
offset++;
lost_sync_count++; // lost sync count limmitation 2K Bytes
- if (lost_sync_count > (1024*2)) {
- GST_WARNING_OBJECT (aacparse, "lost_sync_count is larger than 2048");
+ if (lost_sync_count > (1024 * 2)) {
+ GST_WARNING_OBJECT (aacparse,
+ "lost_sync_count is larger than 2048");
goto EXIT;
}
} else {
offset += frame_size;
num_frames++;
- lost_sync_count=0;
+ lost_sync_count = 0;
}
- } /* while */
+ } /* while */
/* if we can got full file, we can calculate the accurate duration */
- /*MODIFICATION : add defence codes for real buffer_size is different with pull_size*/
+ /* MODIFICATION : add defence codes for real buffer_size is different with pull_size */
//if (pull_size == file_size) {
if (buffer_size == file_size) {
gfloat duration_for_one_frame = 0;
GstClockTime calculated_duration = GST_CLOCK_TIME_NONE;
- GST_INFO_OBJECT (aacparse, "we got total file (%d bytes). do not estimate but make Accurate total duration.", pull_size);
+ GST_INFO_OBJECT (aacparse,
+ "we got total file (%" G_GINT64_FORMAT
+ " bytes). do not estimate but make Accurate total duration.",
+ pull_size);
- duration_for_one_frame = (gfloat)AAC_SAMPLE_PER_FRAME / (gfloat)sample_rate;
- calculated_duration = num_frames * duration_for_one_frame * 1000 * 1000 * 1000;
+ duration_for_one_frame =
+ (gfloat) AAC_SAMPLE_PER_FRAME / (gfloat) sample_rate;
+ calculated_duration =
+ num_frames * duration_for_one_frame * 1000 * 1000 * 1000;
- GST_INFO_OBJECT (aacparse, "duration_for_one_frame %f ms", duration_for_one_frame);
- GST_INFO_OBJECT (aacparse, "calculated duration = %"GST_TIME_FORMAT,
- GST_TIME_ARGS(calculated_duration));
+ GST_INFO_OBJECT (aacparse, "duration_for_one_frame %f ms",
+ duration_for_one_frame);
+ GST_INFO_OBJECT (aacparse, "calculated duration = %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (calculated_duration));
/* 0 means disable estimate */
- gst_base_parse_set_duration (parse, GST_FORMAT_TIME, calculated_duration, 0);
+ gst_base_parse_set_duration (parse, GST_FORMAT_TIME,
+ calculated_duration, 0);
} else {
- GST_INFO_OBJECT (aacparse, "we got %d bytes in total file (%"G_GINT64_FORMAT
- "). can not make accurate duration but Estimate.", pull_size, file_size);
- frame_duration_us = (1024 * 1000000ll + (sample_rate - 1)) / sample_rate;
+ GST_INFO_OBJECT (aacparse,
+ "we got %" G_GUINT64_FORMAT " bytes in total file (%"
+ G_GINT64_FORMAT "). can not make accurate duration but Estimate.",
+ pull_size, file_size);
+ frame_duration_us =
+ (1024 * 1000000ll + (sample_rate - 1)) / sample_rate;
duration = num_frames * frame_duration_us;
if (duration == 0) {
GST_WARNING_OBJECT (aacparse, "Invalid duration");
goto EXIT;
}
- estimated_bitrate = (gint)((gfloat)(offset * 8) / (gfloat)(duration / 1000));
+ estimated_bitrate =
+ (gint) ((gfloat) (offset * 8) / (gfloat) (duration / 1000));
if (estimated_bitrate == 0) {
GST_WARNING_OBJECT (aacparse, "Invalid estimated_bitrate");
goto EXIT;
}
- estimated_duration = (GstClockTime)((file_size * 8) / (estimated_bitrate * 1000)) * GST_SECOND;
-
- GST_INFO_OBJECT (aacparse, "number of frame = %"G_GINT64_FORMAT, num_frames);
- GST_INFO_OBJECT (aacparse, "duration = %"G_GINT64_FORMAT, duration / 1000000);
- GST_INFO_OBJECT (aacparse, "byte = %"G_GINT64_FORMAT, offset);
- GST_INFO_OBJECT (aacparse, "estimated bitrate = %d bps", estimated_bitrate);
- GST_INFO_OBJECT (aacparse, "estimated duration = %"GST_TIME_FORMAT, GST_TIME_ARGS(estimated_duration));
+ estimated_duration =
+ (GstClockTime) ((file_size * 8) / (estimated_bitrate * 1000)) *
+ GST_SECOND;
+
+ GST_INFO_OBJECT (aacparse, "number of frame = %" G_GINT64_FORMAT,
+ num_frames);
+ GST_INFO_OBJECT (aacparse, "duration = %" G_GINT64_FORMAT,
+ duration / 1000000);
+ GST_INFO_OBJECT (aacparse, "byte = %" G_GINT64_FORMAT, offset);
+ GST_INFO_OBJECT (aacparse, "estimated bitrate = %d bps",
+ estimated_bitrate);
+ GST_INFO_OBJECT (aacparse, "estimated duration = %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (estimated_duration));
gst_base_parse_set_average_bitrate (parse, estimated_bitrate * 1000);
/* set update_interval as duration(sec)/2 */
- gst_base_parse_set_duration (parse, GST_FORMAT_TIME, estimated_duration, (gint)(duration/2));
+ gst_base_parse_set_duration (parse, GST_FORMAT_TIME, estimated_duration,
+ (gint) (duration / 2));
}
break;
ret = TRUE;
EXIT:
- gst_buffer_unmap(buffer, &map);
- gst_buffer_unref(buffer);
+ gst_buffer_unmap (buffer, &map);
+ gst_buffer_unref (buffer);
return ret;
}
gst_aac_audio_parse_do_push_seek (GstBaseParse * parse,
GstPad * pad, GstEvent * event)
{
- GstAacParse *aacparse;
- aacparse = GST_AAC_PARSE(parse);
-
+ GstAacParse *aacparse = GST_AAC_PARSE (parse);
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GST_INFO_OBJECT (parse, "doing aac push-based seek");
- gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, &stop_type, &stop);
+ gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
+ &stop_type, &stop);
/* FIXME, always play to the end */
stop = -1;
if (rate <= 0)
goto unsupported_seek;
- if ( cur == 0 ) {
+ if (cur == 0) {
/* handle rewind only */
cur_type = GST_SEEK_TYPE_SET;
byte_cur = 0;
flags |= GST_SEEK_FLAG_FLUSH;
esimate_byte = (cur / (1000 * 1000)) * aacparse->frame_byte;
- if (aacparse->sample_rate> 0)
+ if (aacparse->sample_rate > 0)
frame_dur = (aacparse->spf * 1000) / aacparse->sample_rate;
else
goto unsupported_seek;
if (frame_dur > 0)
- byte_cur = esimate_byte / (frame_dur);
+ byte_cur = esimate_byte / (frame_dur);
else
goto unsupported_seek;
- GST_INFO_OBJECT(parse, "frame_byte(%d) spf(%d) rate (%d) ", aacparse->frame_byte, aacparse->spf, aacparse->sample_rate);
- GST_INFO_OBJECT(parse, "seek cur (%"G_GINT64_FORMAT") = (%"GST_TIME_FORMAT") ", cur, GST_TIME_ARGS (cur));
- GST_INFO_OBJECT(parse, "esimate_byte(%"G_GINT64_FORMAT") esimate_byte (%d)", esimate_byte, frame_dur );
+ GST_INFO_OBJECT (parse, "frame_byte(%d) spf(%d) rate (%d) ",
+ aacparse->frame_byte, aacparse->spf, aacparse->sample_rate);
+ GST_INFO_OBJECT (parse,
+ "seek cur (%" G_GINT64_FORMAT ") = (%" GST_TIME_FORMAT ") ", cur,
+ GST_TIME_ARGS (cur));
+ GST_INFO_OBJECT (parse,
+ "esimate_byte(%" G_GINT64_FORMAT ") esimate_byte (%d)", esimate_byte,
+ frame_dur);
}
/* obtain real upstream total bytes */
if (!gst_pad_peer_query_duration (parse->sinkpad, fmt, &upstream_total_bytes))
upstream_total_bytes = 0;
- GST_INFO_OBJECT (aacparse, "gst_pad_query_peer_duration -upstream_total_bytes (%"G_GUINT64_FORMAT")", upstream_total_bytes);
- aacparse->file_size = upstream_total_bytes;
-
- if ( (byte_cur == -1) || (byte_cur > aacparse->file_size))
- {
- GST_INFO_OBJECT(parse, "[WEB-ERROR] seek cur (%"G_GINT64_FORMAT") > file_size (%"G_GINT64_FORMAT") ", cur, aacparse->file_size );
+ GST_INFO_OBJECT (aacparse,
+ "gst_pad_query_peer_duration -upstream_total_bytes (%" G_GUINT64_FORMAT
+ ")", upstream_total_bytes);
+ aacparse->file_size = upstream_total_bytes;
+
+ if ((byte_cur == -1) || (byte_cur > aacparse->file_size)) {
+ GST_INFO_OBJECT (parse,
+ "[WEB-ERROR] seek cur (%" G_GINT64_FORMAT ") > file_size (%"
+ G_GINT64_FORMAT ") ", cur, aacparse->file_size);
goto abort_seek;
}
- GST_INFO_OBJECT (parse, "Pushing BYTE seek rate %g, " "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, byte_cur, stop);
+ GST_INFO_OBJECT (parse,
+ "Pushing BYTE seek rate %g, " "start %" G_GINT64_FORMAT ", stop %"
+ G_GINT64_FORMAT, rate, byte_cur, stop);
if (!(flags & GST_SEEK_FLAG_KEY_UNIT)) {
- GST_INFO_OBJECT (parse, "Requested seek time: %" GST_TIME_FORMAT ", calculated seek offset: %" G_GUINT64_FORMAT, GST_TIME_ARGS (cur), byte_cur);
+ GST_INFO_OBJECT (parse,
+ "Requested seek time: %" GST_TIME_FORMAT ", calculated seek offset: %"
+ G_GUINT64_FORMAT, GST_TIME_ARGS (cur), byte_cur);
}
/* BYTE seek event */
- event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, byte_cur, stop_type, stop);
+ event =
+ gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, byte_cur,
+ stop_type, stop);
res = gst_pad_push_event (parse->sinkpad, event);
return res;
abort_seek:
{
- GST_DEBUG_OBJECT (parse, "could not determine byte position to seek to, " "seek aborted.");
+ GST_DEBUG_OBJECT (parse,
+ "could not determine byte position to seek to, " "seek aborted.");
return FALSE;
}
{
int length;
if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
- length = ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5);
+ length =
+ ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5);
} else {
length = 0;
}
}
static gboolean
-gst_aac_parse_src_eventfunc(GstBaseParse * parse, GstEvent * event)
+gst_aac_parse_src_eventfunc (GstBaseParse * parse, GstEvent * event)
{
- gboolean handled = FALSE;
+ gboolean handled = FALSE;
GstAacParse *aacparse;
- aacparse = GST_AAC_PARSE(parse);
+ aacparse = GST_AAC_PARSE (parse);
- GST_DEBUG("Entering gst_aac_parse_src_eventfunc header type = %d", aacparse->header_type);
- if(aacparse->header_type == DSPAAC_HEADER_ADTS)
+ GST_DEBUG ("Entering gst_aac_parse_src_eventfunc header type = %d",
+ aacparse->header_type);
+ if (aacparse->header_type == DSPAAC_HEADER_ADTS)
return gst_aac_parse_adts_src_eventfunc (parse, event);
else
goto aac_seek_null_exit;
gst_aac_parse_adts_src_eventfunc (GstBaseParse * parse, GstEvent * event)
{
gboolean handled = FALSE;
- GstAacParse *aacparse;
- aacparse = GST_AAC_PARSE (parse);
+ GstAacParse *aacparse = GST_AAC_PARSE (parse);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
- {
- GstFlowReturn res = GST_FLOW_OK;
- gint64 base_offset = 0, sync_offset = 0, cur = 0;
- gint32 frame_count = 1; /* do not add first frame because it is already in index table */
- gint64 second_count = 0; /* initial 1 second */
- gint64 total_file_size = 0, start_offset = 0;
- GstClockTime current_ts = GST_CLOCK_TIME_NONE;
- GstPadMode pad_mode = GST_PAD_MODE_NONE;
- gboolean large_file_flag = FALSE;
-
- /* check baseparse define these fuction */
- gst_base_parse_get_pad_mode(parse, &pad_mode);
- if (pad_mode != GST_PAD_MODE_PULL) {
- gboolean ret = FALSE;
- GST_INFO_OBJECT (aacparse, "aac parser is PUSH MODE.");
- GstPad* srcpad = parse->srcpad;
- /* check NULL */
- ret = gst_aac_audio_parse_do_push_seek(parse, srcpad, event);
- gst_object_unref(srcpad);
- return ret;
- }
- gst_base_parse_get_upstream_size(parse, &total_file_size);
- gst_base_parse_get_index_last_offset(parse, &start_offset);
- gst_base_parse_get_index_last_ts(parse, ¤t_ts);
-
- if (total_file_size > AAC_LARGE_FILE_SIZE ) {
- large_file_flag = TRUE;
- gst_base_parse_set_seek_mode(parse, 0);
- GST_INFO_OBJECT (aacparse, "larger than big size (2MB).");
- goto aac_seek_null_exit;
- }
+ {
+ GstFlowReturn res = GST_FLOW_OK;
+ gint64 base_offset = 0, cur = 0;
+ gint32 frame_count = 1; /* do not add first frame because it is already in index table */
+ gint64 second_count = 0; /* initial 1 second */
+ gint64 total_file_size = 0, start_offset = 0;
+ GstClockTime current_ts = GST_CLOCK_TIME_NONE;
+ GstPadMode pad_mode = GST_PAD_MODE_NONE;
+
+ /* check baseparse define these fuction */
+ gst_base_parse_get_pad_mode (parse, &pad_mode);
+ if (pad_mode != GST_PAD_MODE_PULL) {
+ gboolean ret = FALSE;
+ GstPad *srcpad = parse->srcpad;
+ GST_INFO_OBJECT (aacparse, "aac parser is PUSH MODE.");
+ /* check NULL */
+ ret = gst_aac_audio_parse_do_push_seek (parse, srcpad, event);
+ gst_object_unref (srcpad);
+ return ret;
+ }
+ gst_base_parse_get_upstream_size (parse, &total_file_size);
+ gst_base_parse_get_index_last_offset (parse, &start_offset);
+ gst_base_parse_get_index_last_ts (parse, ¤t_ts);
+
+ if (total_file_size > AAC_LARGE_FILE_SIZE) {
+ gst_base_parse_set_seek_mode (parse, 0);
+ GST_INFO_OBJECT (aacparse, "larger than big size (2MB).");
+ goto aac_seek_null_exit;
+ }
+
+ GST_DEBUG ("gst_aac_parse_adts_src_eventfunc GST_EVENT_SEEK enter");
- GST_DEBUG("gst_aac_parse_adts_src_eventfunc GST_EVENT_SEEK enter");
+ if (total_file_size == 0 || start_offset >= total_file_size) {
+ GST_ERROR ("last index offset %" G_GINT64_FORMAT
+ " is larger than file size %" G_GINT64_FORMAT, start_offset,
+ total_file_size);
+ break;
+ }
- if (total_file_size == 0 || start_offset >= total_file_size) {
- GST_ERROR("last index offset %d is larger than file size %d", start_offset, total_file_size);
+ gst_event_parse_seek (event, NULL, NULL, NULL, NULL, &cur, NULL, NULL);
+ if (cur <= current_ts) {
+ GST_INFO ("seek to %" GST_TIME_FORMAT " within index table %"
+ GST_TIME_FORMAT ". do not make index table", GST_TIME_ARGS (cur),
+ GST_TIME_ARGS (current_ts));
+ break;
+ } else {
+ GST_INFO ("seek to %" GST_TIME_FORMAT " without index table %"
+ GST_TIME_FORMAT ". make index table", GST_TIME_ARGS (cur),
+ GST_TIME_ARGS (current_ts));
+ }
+
+ GST_INFO ("make AAC(ADTS) Index Table. file_size = %" G_GINT64_FORMAT
+ " last idx offset=%" G_GINT64_FORMAT ", last idx ts=%"
+ GST_TIME_FORMAT, total_file_size, start_offset,
+ GST_TIME_ARGS (current_ts));
+
+ base_offset = start_offset; /* set base by start offset */
+ second_count = current_ts + GST_SECOND; /* 1sec */
+
+ /************************************/
+ /* STEP 0: Setting parse information */
+ /************************************/
+ aacparse->spf = aacparse->frame_samples;
+ aacparse->frame_duration = (aacparse->spf * 1000 * 100) / aacparse->sample_rate; /* duration per frame (msec) */
+ aacparse->frame_per_sec = (aacparse->sample_rate) / aacparse->spf; /* frames per second (ea) */
+
+ /************************************/
+ /* STEP 1: MAX_PULL_RANGE_BUF cycle */
+ /************************************/
+ while (total_file_size - base_offset >= AAC_MAX_PULL_RANGE_BUF) {
+ gint64 offset = 0;
+ GstBuffer *buffer = NULL;
+ guint8 *buf = NULL;
+ GstMapInfo map;
+ GST_INFO ("gst_pad_pull_range %d bytes (from %" G_GINT64_FORMAT
+ ") use max size", AAC_MAX_PULL_RANGE_BUF, base_offset);
+ res =
+ gst_pad_pull_range (parse->sinkpad, base_offset,
+ base_offset + AAC_MAX_PULL_RANGE_BUF, &buffer);
+ if (res != GST_FLOW_OK) {
+ GST_ERROR ("gst_pad_pull_range failed!");
break;
}
- gst_event_parse_seek (event, NULL, NULL, NULL, NULL, &cur, NULL, NULL);
- if (cur <= current_ts) {
- GST_INFO("seek to %"GST_TIME_FORMAT" within index table %"GST_TIME_FORMAT". do not make index table",
- GST_TIME_ARGS(cur), GST_TIME_ARGS(current_ts));
- break;
- } else {
- GST_INFO("seek to %"GST_TIME_FORMAT" without index table %"GST_TIME_FORMAT". make index table",
- GST_TIME_ARGS(cur), GST_TIME_ARGS(current_ts));
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+ buf = map.data;
+ if (buf == NULL) {
+ gst_buffer_unmap (buffer, &map);
+ GST_WARNING ("buffer is NULL in make aac seek table's STEP1");
+ gst_buffer_unref (buffer);
+ goto aac_seek_null_exit;
}
- GST_INFO("make AAC(ADTS) Index Table. file_size = %"G_GINT64_FORMAT" last idx offset=%"G_GINT64_FORMAT
- ", last idx ts=%"GST_TIME_FORMAT, total_file_size, start_offset, GST_TIME_ARGS(current_ts));
-
- base_offset = start_offset; /* set base by start offset */
- second_count = current_ts + GST_SECOND; /* 1sec */
-
- /************************************/
- /* STEP 0: Setting parse information */
- /************************************/
- aacparse->spf = aacparse->frame_samples;
- aacparse->frame_duration = (aacparse->spf * 1000 * 100) / aacparse->sample_rate; /* duration per frame (msec) */
- aacparse->frame_per_sec = (aacparse->sample_rate) / aacparse->spf; /* frames per second (ea) */
-
- /************************************/
- /* STEP 1: MAX_PULL_RANGE_BUF cycle */
- /************************************/
- while (total_file_size - base_offset >= AAC_MAX_PULL_RANGE_BUF) {
- gint64 offset = 0;
- GstBuffer *buffer = NULL;
- guint8 *buf = NULL;
-
- GST_INFO("gst_pad_pull_range %d bytes (from %"G_GINT64_FORMAT") use max size", AAC_MAX_PULL_RANGE_BUF, base_offset);
- res = gst_pad_pull_range (parse->sinkpad, base_offset, base_offset + AAC_MAX_PULL_RANGE_BUF, &buffer);
- if (res != GST_FLOW_OK) {
- GST_ERROR ("gst_pad_pull_range failed!");
- break;
- }
+ while (offset <= AAC_MAX_PULL_RANGE_BUF) {
+ gint frame_size = 0;
- GstMapInfo map;
- gst_buffer_map(buffer, &map, GST_MAP_READ);
- buf = map.data;
- if (buf == NULL) {
- gst_buffer_unmap(buffer, &map);
- GST_WARNING("buffer is NULL in make aac seek table's STEP1");
- gst_buffer_unref (buffer);
- goto aac_seek_null_exit;
- }
+ /* make sure the values in the frame header look sane */
+ frame_size = gst_aac_parse_adts_get_fast_frame_len (buf);
- while (offset <= AAC_MAX_PULL_RANGE_BUF) {
- gint frame_size = 0;
- guint32 header;
-
- /* make sure the values in the frame header look sane */
- frame_size = gst_aac_parse_adts_get_fast_frame_len (buf);
-
- if ((frame_size > 0) && (frame_size < (AAC_MAX_PULL_RANGE_BUF - offset))) {
- if (current_ts > second_count) { /* 1 sec == xx frames. we make idx per sec */
- gst_base_parse_add_index_entry (parse, base_offset +offset, current_ts, TRUE, TRUE); /* force */
- GST_DEBUG("Adding index ts=%"GST_TIME_FORMAT" offset %"G_GINT64_FORMAT,
- GST_TIME_ARGS(current_ts), base_offset + offset);
- second_count += GST_SECOND; /* 1sec */
- }
-
- current_ts += (aacparse->frame_duration * GST_MSECOND) / 100; /* each frame is (frame_duration) ms */
- offset += frame_size;
- buf += frame_size;
- frame_count++;
- } else if (frame_size >= (AAC_MAX_PULL_RANGE_BUF - offset)) {
- GST_DEBUG("we need refill buffer");
- break;
- } else {
- GST_WARNING("we lost sync");
- buf++;
- offset++;
+ if ((frame_size > 0)
+ && (frame_size < (AAC_MAX_PULL_RANGE_BUF - offset))) {
+ if (current_ts > second_count) { /* 1 sec == xx frames. we make idx per sec */
+ gst_base_parse_add_index_entry (parse, base_offset + offset, current_ts, TRUE, TRUE); /* force */
+ GST_DEBUG ("Adding index ts=%" GST_TIME_FORMAT " offset %"
+ G_GINT64_FORMAT, GST_TIME_ARGS (current_ts),
+ base_offset + offset);
+ second_count += GST_SECOND; /* 1sec */
}
- } /* while */
-
- base_offset = base_offset + offset;
- gst_buffer_unmap(buffer, &map);
- gst_buffer_unref (buffer);
- } /* end MAX buffer cycle */
-
- /*******************************/
- /* STEP 2: Remain Buffer cycle */
- /*******************************/
- if (total_file_size - base_offset > 0) {
- gint64 offset = 0;
- GstBuffer *buffer = NULL;
- guint8 *buf = NULL;
-
- GST_INFO("gst_pad_pull_range %"G_GINT64_FORMAT" bytes (from %"G_GINT64_FORMAT") use remain_buf size",
- total_file_size - base_offset, base_offset);
- res = gst_pad_pull_range (parse->sinkpad, base_offset, total_file_size, &buffer);
- if (res != GST_FLOW_OK) {
- GST_ERROR ("gst_pad_pull_range failed!");
+ current_ts += (aacparse->frame_duration * GST_MSECOND) / 100; /* each frame is (frame_duration) ms */
+ offset += frame_size;
+ buf += frame_size;
+ frame_count++;
+ } else if (frame_size >= (AAC_MAX_PULL_RANGE_BUF - offset)) {
+ GST_DEBUG ("we need refill buffer");
break;
+ } else {
+ GST_WARNING ("we lost sync");
+ buf++;
+ offset++;
}
+ } /* while */
+
+ base_offset = base_offset + offset;
+
+ gst_buffer_unmap (buffer, &map);
+ gst_buffer_unref (buffer);
+ } /* end MAX buffer cycle */
+
+ /*******************************/
+ /* STEP 2: Remain Buffer cycle */
+ /*******************************/
+ if (total_file_size - base_offset > 0) {
+ gint64 offset = 0;
+ GstBuffer *buffer = NULL;
+ guint8 *buf = NULL;
+ GstMapInfo map;
+
+ GST_INFO ("gst_pad_pull_range %" G_GINT64_FORMAT " bytes (from %"
+ G_GINT64_FORMAT ") use remain_buf size",
+ total_file_size - base_offset, base_offset);
+ res =
+ gst_pad_pull_range (parse->sinkpad, base_offset, total_file_size,
+ &buffer);
+ if (res != GST_FLOW_OK) {
+ GST_ERROR ("gst_pad_pull_range failed!");
+ break;
+ }
- GstMapInfo map;
- gst_buffer_map(buffer, &map, GST_MAP_READ);
- buf = map.data;
- if (buf == NULL) {
- gst_buffer_unmap(buffer, &map);
- GST_WARNING("buffer is NULL in make aac seek table's STEP2");
- gst_buffer_unref (buffer);
- goto aac_seek_null_exit;
- }
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+ buf = map.data;
+ if (buf == NULL) {
+ gst_buffer_unmap (buffer, &map);
+ GST_WARNING ("buffer is NULL in make aac seek table's STEP2");
+ gst_buffer_unref (buffer);
+ goto aac_seek_null_exit;
+ }
+
+ while (base_offset + offset < total_file_size) {
+ gint frame_size = 0;
+
+ /* make sure the values in the frame header look sane */
+ frame_size = gst_aac_parse_adts_get_fast_frame_len (buf);
- while (base_offset + offset < total_file_size) {
- gint frame_size = 0;
- guint32 header;
-
- /* make sure the values in the frame header look sane */
- frame_size = gst_aac_parse_adts_get_fast_frame_len (buf);
-
- if ((frame_size > 0) && (frame_size <= (total_file_size - (base_offset + offset)))) {
- if (current_ts > second_count) { /* 1 sec == xx frames. we make idx per sec */
- gst_base_parse_add_index_entry (parse, base_offset +offset, current_ts, TRUE, TRUE); /* force */
- GST_DEBUG("Adding index ts=%"GST_TIME_FORMAT" offset %"G_GINT64_FORMAT,
- GST_TIME_ARGS(current_ts), base_offset + offset);
- second_count += GST_SECOND; /* 1sec */
- }
-
- current_ts += (aacparse->frame_duration * GST_MSECOND) / 100; /* each frame is (frame_duration) ms */
- offset += frame_size;
- buf += frame_size;
- frame_count++;
- } else if (frame_size == 0) {
- GST_DEBUG("Frame size is 0 so, Decoding end..");
- break;
- } else {
- GST_WARNING("we lost sync");
- buf++;
- offset++;
+ if ((frame_size > 0)
+ && (frame_size <= (total_file_size - (base_offset + offset)))) {
+ if (current_ts > second_count) { /* 1 sec == xx frames. we make idx per sec */
+ gst_base_parse_add_index_entry (parse, base_offset + offset, current_ts, TRUE, TRUE); /* force */
+ GST_DEBUG ("Adding index ts=%" GST_TIME_FORMAT " offset %"
+ G_GINT64_FORMAT, GST_TIME_ARGS (current_ts),
+ base_offset + offset);
+ second_count += GST_SECOND; /* 1sec */
}
- } /* while */
- gst_buffer_unmap(buffer, &map);
- gst_buffer_unref (buffer);
- } /* end remain_buf buffer cycle */
+ current_ts += (aacparse->frame_duration * GST_MSECOND) / 100; /* each frame is (frame_duration) ms */
+ offset += frame_size;
+ buf += frame_size;
+ frame_count++;
+ } else if (frame_size == 0) {
+ GST_DEBUG ("Frame size is 0 so, Decoding end..");
+ break;
+ } else {
+ GST_WARNING ("we lost sync");
+ buf++;
+ offset++;
+ }
+ } /* while */
- GST_DEBUG("gst_aac_parse_adts_src_eventfunc GST_EVENT_SEEK leave");
+ gst_buffer_unmap (buffer, &map);
+ gst_buffer_unref (buffer);
}
- break;
+ /* end remain_buf buffer cycle */
+ GST_DEBUG ("gst_aac_parse_adts_src_eventfunc GST_EVENT_SEEK leave");
+ }
+ break;
default:
break;
handled = GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event);
return handled;
}
-#endif //end of #ifdef TIZEN_FEATURE_AACPARSE_MODIFICATION
\ No newline at end of file
+#endif //end of #ifdef TIZEN_FEATURE_AACPARSE_MODIFICATION
static void gst_mpeg_audio_parse_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_mpeg_audio_parse_get_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static gboolean gst_mpeg_audio_parse_src_eventfunc (GstBaseParse * parse, GstEvent * event);
+ GValue * value, GParamSpec * pspec);
+static gboolean gst_mpeg_audio_parse_src_eventfunc (GstBaseParse * parse,
+ GstEvent * event);
#endif
static GstFlowReturn gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_get_sink_caps);
#ifdef TIZEN_FEATURE_MP3PARSE_MODIFICATION
- object_class->set_property = gst_mpeg_audio_parse_set_property;
- object_class->get_property = gst_mpeg_audio_parse_get_property;
+ object_class->set_property =
+ GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_set_property);
+ object_class->get_property =
+ GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_get_property);
g_object_class_install_property (object_class, PROP_CHECK_HTTP_SEEK,
- g_param_spec_boolean ("http-pull-mp3dec", "enable/disable",
- "enable/disable mp3dec http seek pull mode",
- DEFAULT_CHECK_HTTP_SEEK,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_param_spec_boolean ("http-pull-mp3dec", "enable/disable",
+ "enable/disable mp3dec http seek pull mode",
+ DEFAULT_CHECK_HTTP_SEEK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/* T.B.D : make full mp3 index table when seek */
parse_class->src_event = gst_mpeg_audio_parse_src_eventfunc;
#endif
const GValue * value, GParamSpec * pspec)
{
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (object);
- GST_INFO_OBJECT (mp3parse, "set_property() START- prop_id(%d)",prop_id);
+ GST_INFO_OBJECT (mp3parse, "set_property() START- prop_id(%d)", prop_id);
switch (prop_id) {
case PROP_CHECK_HTTP_SEEK:
mp3parse->http_seek_flag = g_value_get_boolean (value);
- GST_INFO_OBJECT (mp3parse, "http_seek_flag(%d)", mp3parse->http_seek_flag);
+ GST_INFO_OBJECT (mp3parse, "http_seek_flag(%d)",
+ mp3parse->http_seek_flag);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
static void
gst_mpeg_audio_parse_get_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
+ GValue * value, GParamSpec * pspec)
{
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (object);
- GST_INFO_OBJECT (mp3parse, "get_property() START- prop_id(%d)",prop_id);
+ GST_INFO_OBJECT (mp3parse, "get_property() START- prop_id(%d)", prop_id);
switch (prop_id) {
case PROP_CHECK_HTTP_SEEK:
g_value_set_boolean (value, mp3parse->http_seek_flag);
- GST_INFO_OBJECT (mp3parse, "http_seek_flag(%d)", mp3parse->http_seek_flag);
+ GST_INFO_OBJECT (mp3parse, "http_seek_flag(%d)",
+ mp3parse->http_seek_flag);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
{
GST_INFO_OBJECT (mp3parse, "GST_EVENT_SEEK enter");
if (mp3parse->http_seek_flag) {
- GST_INFO_OBJECT (mp3parse, "souphttpsrc is PULL MODE (so accurate seek mode is OFF)");
+ GST_INFO_OBJECT (mp3parse,
+ "souphttpsrc is PULL MODE (so accurate seek mode is OFF)");
/* Check the declaration of this function in the baseparse */
- gst_base_parse_set_seek_mode(parse, 0);
+ gst_base_parse_set_seek_mode (parse, 0);
goto mp3_seek_null_exit;
}
GST_INFO_OBJECT (mp3parse, "GST_EVENT_SEEK leave");
return handled;
}
#endif
-
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
static void
-gst_rtspsrc_post_error_message ( GstRTSPSrc * src, GstRTSPSrcError error_id, gchar * error_string )
+gst_rtspsrc_post_error_message (GstRTSPSrc * src, GstRTSPSrcError error_id,
+ const gchar * error_string)
{
- GstMessage *message;
- GstStructure *structure;
- gboolean ret = TRUE;
+ GstMessage *message;
+ GstStructure *structure;
+ gboolean ret = TRUE;
- GST_ERROR_OBJECT ( src, "[%d] %s", error_id, error_string );
+ GST_ERROR_OBJECT (src, "[%d] %s", error_id, error_string);
- structure = gst_structure_new ("streaming_error",
- "error_id", G_TYPE_UINT, error_id,
- "error_string", G_TYPE_STRING, error_string, NULL);
+ structure = gst_structure_new ("streaming_error",
+ "error_id", G_TYPE_UINT, error_id,
+ "error_string", G_TYPE_STRING, error_string, NULL);
- message = gst_message_new_custom ( GST_MESSAGE_ERROR, GST_OBJECT(src), structure );
+ message =
+ gst_message_new_custom (GST_MESSAGE_ERROR, GST_OBJECT (src), structure);
- ret = gst_element_post_message ( GST_ELEMENT (src), message );
- if ( !ret )
- GST_ERROR_OBJECT ( src, "fail to post error message." );
+ ret = gst_element_post_message (GST_ELEMENT (src), message);
+ if (!ret)
+ GST_ERROR_OBJECT (src, "fail to post error message.");
- return;
+ return;
}
#endif
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
rtspsrc->is_audio_codec_supported = FALSE;
rtspsrc->is_video_codec_supported = FALSE;
- if (rtspsrc->audio_codec)
- {
- g_free(rtspsrc->audio_codec);
- rtspsrc->audio_codec = NULL;
+ if (rtspsrc->audio_codec) {
+ g_free (rtspsrc->audio_codec);
+ rtspsrc->audio_codec = NULL;
}
- if (rtspsrc->video_codec)
- {
- g_free(rtspsrc->video_codec);
- rtspsrc->video_codec = NULL;
+ if (rtspsrc->video_codec) {
+ g_free (rtspsrc->video_codec);
+ rtspsrc->video_codec = NULL;
}
- if (rtspsrc->video_frame_size)
- {
- g_free(rtspsrc->video_frame_size);
- rtspsrc->video_frame_size = NULL;
+ if (rtspsrc->video_frame_size) {
+ g_free (rtspsrc->video_frame_size);
+ rtspsrc->video_frame_size = NULL;
}
#endif
gst_rtsp_ext_list_free (rtspsrc->extensions);
break;
case PROP_RESUME_POSITION:
rtspsrc->last_pos = g_value_get_uint64 (value);
- GST_DEBUG_OBJECT (rtspsrc, "src->last_pos value set to %"GST_TIME_FORMAT,GST_TIME_ARGS (rtspsrc->last_pos));
+ GST_DEBUG_OBJECT (rtspsrc, "src->last_pos value set to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (rtspsrc->last_pos));
break;
#endif
case PROP_TCP_TIMEOUT:
const gchar *enc;
PtMapItem item;
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gchar *encoder , *mediatype;
+ const gchar *encoder, *mediatype;
#endif
pt = atoi (gst_sdp_media_get_format (media, i));
if (strcmp (enc, "X-ASF-PF") == 0)
stream->container = TRUE;
}
-
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- if ((mediatype = gst_structure_get_string (s, "media")))
- {
- GST_DEBUG_OBJECT (src, " mediatype : %s",mediatype);
- if(!strcmp(mediatype, "video"))
- {
- if ((encoder = gst_structure_get_string (s, "encoding-name")))
- {
- GST_DEBUG_OBJECT (src, " encoder : %s",encoder);
- if((!strcmp(encoder,"H261")) ||
- (!strcmp(encoder,"H263")) ||
- (!strcmp(encoder,"H263-1998")) || (!strcmp(encoder,"H263-2000")) ||
- (!strcmp(encoder,"H264")) ||
- (!strcmp(encoder,"MP4V-ES")))
- {
- src->is_video_codec_supported = TRUE;
- GST_DEBUG_OBJECT (src, "Supported Video Codec %s",encoder);
- }
- else
- {
- GST_DEBUG_OBJECT (src, "Unsupported Video Codec %s",encoder);
- }
- }
-
- src->video_codec = g_strdup(encoder);
- src->video_frame_size = g_strdup(gst_structure_get_string(s, "a-framesize"));
- GST_DEBUG_OBJECT (src, "video_codec %s , video_frame_size %s ",src->video_codec,src->video_frame_size);
- }
- else if (!strcmp(mediatype, "audio"))
- {
- if ((encoder = gst_structure_get_string (s, "encoding-name")))
- {
- GST_DEBUG_OBJECT (src, " encoder : %s",encoder);
- if((!strcmp(encoder,"MP4A-LATM")) ||
- (!strcmp(encoder,"AMR")) ||(!strcmp(encoder,"AMR-WB")) ||(!strcmp(encoder,"AMR-NB")) ||
- (!strcmp(encoder,"mpeg4-generic")) ||(!strcmp(encoder,"MPEG4-GENERIC")) ||
- (!strcmp(encoder,"QCELP")) || ((strstr(encoder, "G726"))||(strstr(encoder, "PCMU"))))
- {
- src->is_audio_codec_supported = TRUE;
- GST_DEBUG_OBJECT (src, "Supported Audio Codec %s",encoder);
- }
- else
- {
- GST_DEBUG_OBJECT (src, "Unsupported Audio Codec %s",encoder);
- }
- }
-
- src->audio_codec = g_strdup(encoder);
- GST_DEBUG_OBJECT (src, "audio_codec %s ",src->audio_codec);
- }
+ if ((mediatype = gst_structure_get_string (s, "media"))) {
+ GST_DEBUG_OBJECT (src, " mediatype : %s", mediatype);
+ if (!strcmp (mediatype, "video")) {
+ if ((encoder = gst_structure_get_string (s, "encoding-name"))) {
+ GST_DEBUG_OBJECT (src, " encoder : %s", encoder);
+ if ((!strcmp (encoder, "H261")) ||
+ (!strcmp (encoder, "H263")) ||
+ (!strcmp (encoder, "H263-1998"))
+ || (!strcmp (encoder, "H263-2000")) || (!strcmp (encoder, "H264"))
+ || (!strcmp (encoder, "MP4V-ES"))) {
+ src->is_video_codec_supported = TRUE;
+ GST_DEBUG_OBJECT (src, "Supported Video Codec %s", encoder);
+ } else {
+ GST_DEBUG_OBJECT (src, "Unsupported Video Codec %s", encoder);
+ }
+ }
+
+ src->video_codec = g_strdup (encoder);
+ src->video_frame_size =
+ g_strdup (gst_structure_get_string (s, "a-framesize"));
+ GST_DEBUG_OBJECT (src, "video_codec %s , video_frame_size %s ",
+ src->video_codec, src->video_frame_size);
+ } else if (!strcmp (mediatype, "audio")) {
+ if ((encoder = gst_structure_get_string (s, "encoding-name"))) {
+ GST_DEBUG_OBJECT (src, " encoder : %s", encoder);
+ if ((!strcmp (encoder, "MP4A-LATM")) ||
+ (!strcmp (encoder, "AMR")) || (!strcmp (encoder, "AMR-WB"))
+ || (!strcmp (encoder, "AMR-NB"))
+ || (!strcmp (encoder, "mpeg4-generic"))
+ || (!strcmp (encoder, "MPEG4-GENERIC"))
+ || (!strcmp (encoder, "QCELP")) || ((strstr (encoder, "G726"))
+ || (strstr (encoder, "PCMU")))) {
+ src->is_audio_codec_supported = TRUE;
+ GST_DEBUG_OBJECT (src, "Supported Audio Codec %s", encoder);
+ } else {
+ GST_DEBUG_OBJECT (src, "Unsupported Audio Codec %s", encoder);
+ }
+ }
+
+ src->audio_codec = g_strdup (encoder);
+ GST_DEBUG_OBJECT (src, "audio_codec %s ", src->audio_codec);
+ }
}
#endif
{
GList *walk;
- GST_WARNING_OBJECT(src, "Setting [%s] element state to: %s \n", GST_ELEMENT_NAME(GST_ELEMENT_CAST (src)),gst_element_state_get_name(state));
+ GST_WARNING_OBJECT (src, "Setting [%s] element state to: %s \n",
+ GST_ELEMENT_NAME (GST_ELEMENT_CAST (src)),
+ gst_element_state_get_name (state));
if (src->manager)
gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
gchar *str = gst_rtsp_strresult (res);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_SERVER,"Could not receive message.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_SERVER,
+ "Could not receive message.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
gchar *str = gst_rtsp_strresult (res);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_SERVICE_UNAVAILABLE,"Could not handle server message.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_SERVICE_UNAVAILABLE,
+ "Could not handle server message.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not handle server message. (%s)", str));
src->conninfo.connected = FALSE;
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_CONNECTION_FAIL,"Could not connect to server.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not connect to server.");
#else
- GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
+ GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not connect to server. (%s)", str));
#endif
g_free (str);
gchar *str = gst_rtsp_strresult (res);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_SERVER_DISCONNECTED,"Could not receive message.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_SERVER_DISCONNECTED,
+ "Could not receive message.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
gst_rtsp_message_unset (&message);
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_SERVICE_UNAVAILABLE,"Could not handle server message.");
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_SERVICE_UNAVAILABLE,
+ "Could not handle server message.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
- ("Could not handle server message. (%s)", str));
+ ("Could not handle server message. (%s)", str));
#endif
g_free (str);
ret = GST_FLOW_ERROR;
src->cur_protocols = 0;
/* no transport possible, post an error and stop */
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_TRANSPORT,"Could not receive any UDP packets for seconds, maybe your firewall is blocking it. No other protocols to try.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_TRANSPORT,
+ "Could not receive any UDP packets for seconds, maybe your firewall is blocking it. No other protocols to try.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- GstStructure *s;
+ GstMessage *s;
#endif
GST_WARNING_OBJECT (src, "Got cmd %s", cmd_to_string (cmd));
switch (cmd) {
case CMD_OPEN:
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- GST_DEBUG_OBJECT (src, "rtsp_duration %"GST_TIME_FORMAT", rtsp_audio_codec %s , rtsp_video_codec %s , rtsp_video_frame_size %s",
- GST_TIME_ARGS(src->segment.duration),src->audio_codec,src->video_codec,src->video_frame_size);
-
- /* post message */
- s = gst_message_new_element (GST_OBJECT_CAST (src),
- gst_structure_new ("rtspsrc_properties",
- "rtsp_duration",G_TYPE_UINT64,src->segment.duration,
- "rtsp_audio_codec", G_TYPE_STRING, src->audio_codec,
- "rtsp_video_codec", G_TYPE_STRING, src->video_codec,
- "rtsp_video_frame_size", G_TYPE_STRING, src->video_frame_size,NULL));
-
- gst_element_post_message (GST_ELEMENT_CAST (src), s);
+ GST_DEBUG_OBJECT (src,
+ "rtsp_duration %" GST_TIME_FORMAT
+ ", rtsp_audio_codec %s , rtsp_video_codec %s , rtsp_video_frame_size %s",
+ GST_TIME_ARGS (src->segment.duration), src->audio_codec,
+ src->video_codec, src->video_frame_size);
+
+ /* post message */
+ s = gst_message_new_element (GST_OBJECT_CAST (src),
+ gst_structure_new ("rtspsrc_properties",
+ "rtsp_duration", G_TYPE_UINT64, src->segment.duration,
+ "rtsp_audio_codec", G_TYPE_STRING, src->audio_codec,
+ "rtsp_video_codec", G_TYPE_STRING, src->video_codec,
+ "rtsp_video_frame_size", G_TYPE_STRING, src->video_frame_size,
+ NULL));
+
+ gst_element_post_message (GST_ELEMENT_CAST (src), s);
#endif
GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
/* rtspsrc PAUSE state should be here for parsing sdp before PAUSE state changed. */
- g_mutex_lock(&(src)->pause_lock);
+ g_mutex_lock (&(src)->pause_lock);
g_cond_signal (&(src)->open_end);
- g_mutex_unlock(&(src)->pause_lock);
+ g_mutex_unlock (&(src)->pause_lock);
#endif
break;
case CMD_PLAY:
case CMD_OPEN:
GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
-/* Ending conditional wait for pause when open fails.*/
- g_mutex_lock(&(src)->pause_lock);
+ /* Ending conditional wait for pause when open fails.*/
+ g_mutex_lock (&(src)->pause_lock);
g_cond_signal (&(src)->open_end);
- g_mutex_unlock(&(src)->pause_lock);
- GST_WARNING_OBJECT (src, "ending conditional wait for pause as open is failed.");
+ g_mutex_unlock (&(src)->pause_lock);
+ GST_WARNING_OBJECT (src,
+ "ending conditional wait for pause as open is failed.");
#endif
break;
case CMD_PLAY:
/* Output an error indicating that we couldn't connect because there were
* no supported authentication protocols */
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_NOT_AUTHORIZED,"No supported authentication protocol was found");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_NOT_AUTHORIZED,
+ "No supported authentication protocol was found");
#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("No supported authentication protocol was found"));
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_CONNECTION_FAIL,"Could not send message.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
#else
- GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+ GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
#endif
} else {
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_SERVER_DISCONNECTED,"Could not receive message.");
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_SERVER_DISCONNECTED,
+ "Could not receive message.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
switch (response->type_data.response.code) {
case GST_RTSP_STS_NOT_FOUND:
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_REQUEST,"STS NOT FOUND");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "STS NOT FOUND");
#else
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
response->type_data.response.reason));
break;
case GST_RTSP_STS_UNAUTHORIZED:
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_NOT_AUTHORIZED,"STS NOT AUTHORIZED");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_NOT_AUTHORIZED,
+ "STS NOT AUTHORIZED");
#else
GST_ELEMENT_ERROR (src, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
response->type_data.response.reason));
break;
default:
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_UNEXPECTED_MSG,"Got error response from Server");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_UNEXPECTED_MSG,
+ "Got error response from Server");
#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Got error response: %d (%s).", response->type_data.response.code,
no_describe:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,"Server does not support DESCRIBE.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+ "Server does not support DESCRIBE.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support DESCRIBE."));
no_setup:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,"Server does not support SETUP.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+ "Server does not support SETUP.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support SETUP."));
no_protocols:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_INVALID_PROTOCOL,"Could not connect to server, no protocols left");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_INVALID_PROTOCOL,
+ "Could not connect to server, no protocols left");
#else
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
no_streams:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_CONTENT_NOT_FOUND,"SDP contains no streams");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONTENT_NOT_FOUND,
+ "SDP contains no streams");
#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("SDP contains no streams"));
gchar *str = gst_rtsp_strresult (res);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_REQUEST,"Could not create request.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request.");
#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
setup_transport_failed:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_REQUEST,"Could not setup transport.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not setup transport.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not setup transport."));
}
response_error:
{
+#ifndef TIZEN_FEATURE_RTSP_MODIFICATION
const gchar *str = gst_rtsp_status_as_text (code);
+#endif
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_UNEXPECTED_MSG,"Error from Server .");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_UNEXPECTED_MSG,
+ "Error from Server .");
#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Error (%d): %s", code, GST_STR_NULL (str)));
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_CONNECTION_FAIL,"Could not send message.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
no_transport:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_TRANSPORT,"Server did not select transport.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_TRANSPORT,
+ "Server did not select transport.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server did not select transport."));
/* none of the available error codes is really right .. */
if (unsupported_real) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,"No supported stream was found. You might need to install a GStreamer RTSP extension plugin for Real media streams.");
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+ "No supported stream was found. You might need to install a GStreamer RTSP extension plugin for Real media streams.");
#else
GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
(_("No supported stream was found. You might need to install a "
#endif
} else {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,"No supported stream was found. You might need to allow more transport protocols or may otherwise be missing the right GStreamer RTSP extension plugin.");
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+ "No supported stream was found. You might need to allow more transport protocols or may otherwise be missing the right GStreamer RTSP extension plugin.");
#else
GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
(_("No supported stream was found. You might need to allow "
src->state = GST_RTSP_STATE_INIT;
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
/* Check for the support for the Media codecs */
- if ((!src->is_audio_codec_supported)&&(!src->is_video_codec_supported))
- {
- GST_ERROR_OBJECT (src, "UnSupported Media Type !!!! \n");
- goto unsupported_file_type;
- }
- else
- {
- GST_DEBUG_OBJECT (src, "Supported Media Type. \n");
+ if ((!src->is_audio_codec_supported) && (!src->is_video_codec_supported)) {
+ GST_ERROR_OBJECT (src, "UnSupported Media Type !!!! \n");
+ goto unsupported_file_type;
+ } else {
+ GST_DEBUG_OBJECT (src, "Supported Media Type. \n");
}
#endif
/* setup streams */
return res;
}
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- unsupported_file_type:
+unsupported_file_type:
{
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,"No supported stream was found");
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+ "No supported stream was found");
res = GST_RTSP_ERROR;
gst_rtspsrc_cleanup (src);
return res;
no_url:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_INVALID_URL,"No valid RTSP URL was provided");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_INVALID_URL,
+ "No valid RTSP URL was provided");
#else
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
("No valid RTSP URL was provided"));
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_CONNECTION_FAIL,"Failed to connect.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Failed to connect.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Failed to connect. (%s)", str));
gchar *str = gst_rtsp_strresult (res);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_REQUEST,"Could not create request.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request.");
#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
wrong_content_type:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_OPTION_NOT_SUPPORTED,"Server does not support SDP. ");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_OPTION_NOT_SUPPORTED,
+ "Server does not support SDP. ");
#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server does not support SDP, got %s.", respcont));
no_describe:
{
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,"Server can not provide an SDP.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+ "Server can not provide an SDP.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server can not provide an SDP."));
gchar *str = gst_rtsp_strresult (res);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_REQUEST,"Could not create request.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request.");
#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_CONNECTION_FAIL,"Could not send message.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
{
gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- if (src->start_position !=0 && segment->position == 0) {
+ if (src->start_position != 0 && segment->position == 0) {
segment->position = src->start_position;
src->start_position = 0;
}
}
}
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- GST_DEBUG_OBJECT (src, "Range Header Added : npt=%s-",val_str);
+ GST_DEBUG_OBJECT (src, "Range Header Added : npt=%s-", val_str);
#endif
return g_strdup_printf ("npt=%s-", val_str);
}
src->need_segment = TRUE;
}
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- else
- {
+ else {
/*
Updating position with the MSL current position as gst_rtspsrc_get_position() does not return correct position.
*/
- GST_DEBUG_OBJECT (src, " During normal pause-resume , segment->position=%" GST_TIME_FORMAT",src->start_position=%"GST_TIME_FORMAT,
- GST_TIME_ARGS (segment->position),GST_TIME_ARGS (src->start_position));
+ GST_DEBUG_OBJECT (src,
+ " During normal pause-resume , segment->position=%" GST_TIME_FORMAT
+ ",src->start_position=%" GST_TIME_FORMAT,
+ GST_TIME_ARGS (segment->position),
+ GST_TIME_ARGS (src->start_position));
segment->position = src->last_pos;
}
gchar *str = gst_rtsp_strresult (res);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_REQUEST,"Could not create request. ");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request. ");
#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_CONNECTION_FAIL,"Could not send message.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
gchar *str = gst_rtsp_strresult (res);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_BAD_REQUEST,"Could not create request.");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request.");
#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- gst_rtspsrc_post_error_message ( src,
- GST_RTSPSRC_ERROR_CONNECTION_FAIL,"Could not send message. ");
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message. ");
#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
#endif
rtspsrc = GST_RTSPSRC (element);
- GST_WARNING_OBJECT(rtspsrc, "State change transition: %d \n", transition);
+ GST_WARNING_OBJECT (rtspsrc, "State change transition: %d \n", transition);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
case GST_STATE_CHANGE_READY_TO_PAUSED:
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
/* don't change to PAUSE state before complete stream opend.
- see gst_rtspsrc_loop_complete_cmd() */
- g_mutex_lock(&(rtspsrc)->pause_lock);
+ see gst_rtspsrc_loop_complete_cmd() */
+ g_mutex_lock (&(rtspsrc)->pause_lock);
end_time = g_get_monotonic_time () + 10 * G_TIME_SPAN_SECOND;
- if (!g_cond_wait_until (&(rtspsrc)->open_end, &(rtspsrc)->pause_lock, end_time)) {
- GST_WARNING_OBJECT(rtspsrc, "time out: stream opend is not completed yet..");
+ if (!g_cond_wait_until (&(rtspsrc)->open_end, &(rtspsrc)->pause_lock,
+ end_time)) {
+ GST_WARNING_OBJECT (rtspsrc,
+ "time out: stream opend is not completed yet..");
}
- g_mutex_unlock(&(rtspsrc)->pause_lock);
+ g_mutex_unlock (&(rtspsrc)->pause_lock);
#endif
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
-DTIZEN_FEATURE_GST_UPSTREAM\
-DTIZEN_FEATURE_RTSP_MODIFICATION\
-DTIZEN_FEATURE_GST_MUX_ENHANCEMENT\
- -DTIZEN_FEATURE_SOUP_MODIFICATION"
+ -DTIZEN_FEATURE_SOUP_MODIFICATION\
+ -DTIZEN_FEATURE_BASEPARSE_MODIFICATION"
%configure\
%if ! 0%{?ENABLE_AALIB}
--disable-aalib\
projects / platform / upstream / gst-plugins-good.git / commitdiff