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fix more dynamic protocol stuff, needed by the forthcoming h264
streaming patch. (Minor additions to give more information to the dynamic protocol handlers, and a slight rearrangement of code.) Patch by Ryan Martell %rdm4 A martellventures P com% Original thread: Date: Oct 29, 2006 2:30 AM Subject: Re: [Ffmpeg-devel] RTP patches & RFC Originally committed as revision 6831 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -328,6 +328,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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* open a new RTP parse context for stream 'st'. 'st' can be NULL for
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* MPEG2TS streams to indicate that they should be demuxed inside the
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* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
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* TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
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*/
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
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{
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@ -418,6 +419,39 @@ static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
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return 0;
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}
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/**
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* This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
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*/
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static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
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{
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switch(s->st->codec->codec_id) {
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case CODEC_ID_MP2:
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case CODEC_ID_MPEG1VIDEO:
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if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
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int64_t addend;
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int delta_timestamp;
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/* XXX: is it really necessary to unify the timestamp base ? */
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/* compute pts from timestamp with received ntp_time */
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delta_timestamp = timestamp - s->last_rtcp_timestamp;
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/* convert to 90 kHz without overflow */
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addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
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addend = (addend * 5625) >> 14;
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pkt->pts = addend + delta_timestamp;
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}
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break;
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case CODEC_ID_MPEG4AAC:
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case CODEC_ID_H264:
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case CODEC_ID_MPEG4:
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pkt->pts = timestamp;
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break;
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default:
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/* no timestamp info yet */
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break;
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}
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pkt->stream_index = s->st->index;
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}
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/**
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* Parse an RTP or RTCP packet directly sent as a buffer.
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* @param s RTP parse context.
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@ -431,15 +465,20 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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const uint8_t *buf, int len)
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{
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unsigned int ssrc, h;
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int payload_type, seq, delta_timestamp, ret;
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int payload_type, seq, ret;
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AVStream *st;
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uint32_t timestamp;
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int rv= 0;
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if (!buf) {
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/* return the next packets, if any */
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if(s->st && s->parse_packet) {
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return s->parse_packet(s, pkt, 0, NULL, 0);
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timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
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rv= s->parse_packet(s, pkt, ×tamp, NULL, 0);
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finalize_packet(s, pkt, timestamp);
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return rv;
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} else {
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// TODO: Move to a dynamic packet handler (like above)
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if (s->read_buf_index >= s->read_buf_size)
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return -1;
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ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
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@ -548,12 +587,11 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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}
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s->read_buf_size = len;
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s->buf_ptr = buf;
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pkt->stream_index = s->st->index;
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return 0; ///< Temporary return.
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rv= 0;
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break;
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default:
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if(s->parse_packet) {
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return s->parse_packet(s, pkt, timestamp, buf, len);
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rv= s->parse_packet(s, pkt, ×tamp, buf, len);
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} else {
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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@ -561,32 +599,10 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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break;
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}
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switch(st->codec->codec_id) {
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case CODEC_ID_MP2:
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case CODEC_ID_MPEG1VIDEO:
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if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
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int64_t addend;
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/* XXX: is it really necessary to unify the timestamp base ? */
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/* compute pts from timestamp with received ntp_time */
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delta_timestamp = timestamp - s->last_rtcp_timestamp;
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/* convert to 90 kHz without overflow */
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addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
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addend = (addend * 5625) >> 14;
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pkt->pts = addend + delta_timestamp;
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}
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break;
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case CODEC_ID_MPEG4AAC:
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case CODEC_ID_H264:
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case CODEC_ID_MPEG4:
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pkt->pts = timestamp;
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break;
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default:
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/* no timestamp info yet */
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break;
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}
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pkt->stream_index = s->st->index;
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// now perform timestamp things....
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finalize_packet(s, pkt, timestamp);
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}
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return 0;
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return rv;
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}
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void rtp_parse_close(RTPDemuxContext *s)
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@ -25,7 +25,7 @@
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typedef int (*DynamicPayloadPacketHandlerProc) (struct RTPDemuxContext * s,
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AVPacket * pkt,
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uint32_t timestamp,
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uint32_t *timestamp,
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const uint8_t * buf,
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int len);
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@ -200,6 +200,8 @@ static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payl
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i = atoi(buf);
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if (i > 0)
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codec->channels = i;
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// TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the
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// frequency. No problem, but the sample rate is being set here by the sdp line. Upcoming patch forthcoming. (rdm)
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}
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av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
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av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
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@ -287,6 +289,25 @@ static attrname_map_t attr_names[]=
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{NULL, -1, -1},
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};
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/** parse the attribute line from the fmtp a line of an sdp resonse. This is broken out as a function
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* because it is used in rtp_h264.c, which is forthcoming.
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*/
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int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
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{
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skip_spaces(p);
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if(**p)
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{
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get_word_sep(attr, attr_size, "=", p);
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if (**p == '=')
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(*p)++;
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get_word_sep(value, value_size, ";", p);
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if (**p == ';')
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(*p)++;
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return 1;
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}
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return 0;
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}
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/* parse a SDP line and save stream attributes */
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static void sdp_parse_fmtp(AVStream *st, const char *p)
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{
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@ -298,6 +319,7 @@ static void sdp_parse_fmtp(AVStream *st, const char *p)
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AVCodecContext *codec = st->codec;
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rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
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// TODO (Replace with rtsp_next_attr_and_value)
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/* loop on each attribute */
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for(;;) {
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skip_spaces(&p);
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@ -471,6 +493,19 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
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}
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}
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}
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} else if(strstart(p, "framesize:", &p)) {
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// let dynamic protocol handlers have a stab at the line.
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get_word(buf1, sizeof(buf1), &p);
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payload_type = atoi(buf1);
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for(i = 0; i < s->nb_streams;i++) {
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st = s->streams[i];
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rtsp_st = st->priv_data;
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if (rtsp_st->sdp_payload_type == payload_type) {
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if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
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rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf);
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}
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}
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}
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}
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break;
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}
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