diff options
Diffstat (limited to 'libavformat/rtp.c')
| -rw-r--r-- | libavformat/rtp.c | 44 |
1 files changed, 22 insertions, 22 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c index 2bd61ca9e9..b9758c917a 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -40,7 +40,7 @@ buffer to 'rtp_write_packet' contains all the packets for ONE frame. Each packet should have a four byte header containing the length in big endian format (same trick as - 'url_open_dyn_packet_buf') + 'url_open_dyn_packet_buf') */ /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ @@ -197,7 +197,7 @@ struct RTPDemuxContext { MpegTSContext *ts; /* only used for MP2T payloads */ int read_buf_index; int read_buf_size; - + /* rtcp sender statistics receive */ int64_t last_rtcp_ntp_time; int64_t first_rtcp_ntp_time; @@ -268,7 +268,7 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l /** * open a new RTP parse context for stream 'st'. 'st' can be NULL for * MPEG2TS streams to indicate that they should be demuxed inside the - * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) + * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) */ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data) { @@ -354,27 +354,27 @@ static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) } /** - * Parse an RTP or RTCP packet directly sent as a buffer. + * Parse an RTP or RTCP packet directly sent as a buffer. * @param s RTP parse context. * @param pkt returned packet * @param buf input buffer or NULL to read the next packets * @param len buffer len - * @return 0 if a packet is returned, 1 if a packet is returned and more can follow + * @return 0 if a packet is returned, 1 if a packet is returned and more can follow * (use buf as NULL to read the next). -1 if no packet (error or no more packet). */ -int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, +int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len) { unsigned int ssrc, h; int payload_type, seq, delta_timestamp, ret; AVStream *st; uint32_t timestamp; - + if (!buf) { /* return the next packets, if any */ if (s->read_buf_index >= s->read_buf_size) return -1; - ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, + ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, s->read_buf_size - s->read_buf_index); if (ret < 0) return -1; @@ -398,13 +398,13 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, seq = (buf[2] << 8) | buf[3]; timestamp = decode_be32(buf + 4); ssrc = decode_be32(buf + 8); - + /* NOTE: we can handle only one payload type */ if (s->payload_type != payload_type) return -1; #if defined(DEBUG) || 1 if (seq != ((s->seq + 1) & 0xffff)) { - av_log(s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", + av_log(s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", payload_type, seq, ((s->seq + 1) & 0xffff)); } #endif @@ -458,7 +458,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, memcpy(pkt->data, buf, len); break; } - + switch(st->codec->codec_id) { case CODEC_ID_MP2: case CODEC_ID_MPEG1VIDEO: @@ -599,10 +599,10 @@ static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int put_be16(&s1->pb, s->seq); put_be32(&s1->pb, s->timestamp); put_be32(&s1->pb, s->ssrc); - + put_buffer(&s1->pb, buf1, len); put_flush_packet(&s1->pb); - + s->seq++; s->octet_count += len; s->packet_count++; @@ -639,7 +639,7 @@ static void rtp_send_samples(AVFormatContext *s1, s->timestamp += n / sample_size; } } -} +} /* NOTE: we suppose that exactly one frame is given as argument here */ /* XXX: test it */ @@ -659,7 +659,7 @@ static void rtp_send_mpegaudio(AVFormatContext *s1, rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); s->buf_ptr = s->buf + 4; /* 90 KHz time stamp */ - s->timestamp = s->base_timestamp + + s->timestamp = s->base_timestamp + (s->cur_timestamp * 90000LL) / st->codec->sample_rate; } } @@ -727,7 +727,7 @@ static void rtp_send_mpegvideo(AVFormatContext *s1, *q++ = h >> 8; *q++ = h; } - + len = max_packet_size - (q - s->buf); if (len > size) len = size; @@ -736,7 +736,7 @@ static void rtp_send_mpegvideo(AVFormatContext *s1, q += len; /* 90 KHz time stamp */ - s->timestamp = s->base_timestamp + + s->timestamp = s->base_timestamp + av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps rtp_send_data(s1, s->buf, q - s->buf, (len == size)); @@ -761,7 +761,7 @@ static void rtp_send_raw(AVFormatContext *s1, len = size; /* 90 KHz time stamp */ - s->timestamp = s->base_timestamp + + s->timestamp = s->base_timestamp + av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps rtp_send_data(s1, buf1, len, (len == size)); @@ -786,7 +786,7 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1, buf1 += len; size -= len; s->buf_ptr += len; - + out_len = s->buf_ptr - s->buf; if (out_len >= s->max_payload_size) { rtp_send_data(s1, s->buf, out_len, 0); @@ -804,19 +804,19 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) int64_t ntp_time; int size= pkt->size; uint8_t *buf1= pkt->data; - + #ifdef DEBUG printf("%d: write len=%d\n", pkt->stream_index, size); #endif /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ - rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / + rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / RTCP_TX_RATIO_DEN; if (s->first_packet || rtcp_bytes >= 28) { /* compute NTP time */ /* XXX: 90 kHz timestamp hardcoded */ ntp_time = (pkt->pts << 28) / 5625; - rtcp_send_sr(s1, ntp_time); + rtcp_send_sr(s1, ntp_time); s->last_octet_count = s->octet_count; s->first_packet = 0; } |