RTSP: Support video only.

pull/1809/head
winlin 5 years ago
parent f13a1b8ac0
commit 32c5b6fbce

@ -120,9 +120,11 @@ srs_error_t SrsRtpConn::on_udp_packet(const sockaddr* from, const int fromlen, c
// always free it. // always free it.
SrsAutoFree(SrsRtpPacket, cache); SrsAutoFree(SrsRtpPacket, cache);
if ((err = rtsp->on_rtp_packet(cache, stream_id)) != srs_success) { err = rtsp->on_rtp_packet(cache, stream_id);
return srs_error_wrap(err, "process rtp packet"); if (err != srs_success) {
srs_warn("ignore RTP packet err %s", srs_error_desc(err).c_str());
srs_freep(err);
} }
return err; return err;
@ -495,54 +497,62 @@ srs_error_t SrsRtspConn::write_sequence_header()
} }
// generate audio sh by audio specific config. // generate audio sh by audio specific config.
if (true) { if (aac_specific_config.empty()) {
std::string sh = aac_specific_config; srs_warn("no audio asc");
return err;
SrsFormat* format = new SrsFormat();
SrsAutoFree(SrsFormat, format);
if ((err = format->on_aac_sequence_header((char*)sh.c_str(), (int)sh.length())) != srs_success) {
return srs_error_wrap(err, "on aac sequence header");
}
SrsAudioCodecConfig* dec = format->acodec;
acodec->sound_format = SrsAudioCodecIdAAC;
acodec->sound_type = (dec->aac_channels == 2)? SrsAudioChannelsStereo : SrsAudioChannelsMono;
acodec->sound_size = SrsAudioSampleBits16bit;
acodec->aac_packet_type = 0;
static int srs_aac_srates[] = {
96000, 88200, 64000, 48000,
44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000,
7350, 0, 0, 0
};
switch (srs_aac_srates[dec->aac_sample_rate]) {
case 11025:
acodec->sound_rate = SrsAudioSampleRate11025;
break;
case 22050:
acodec->sound_rate = SrsAudioSampleRate22050;
break;
case 44100:
acodec->sound_rate = SrsAudioSampleRate44100;
break;
default:
break;
};
if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), acodec, (uint32_t)dts)) != srs_success) {
return srs_error_wrap(err, "write audio raw frame");
}
} }
std::string sh = aac_specific_config;
SrsFormat* format = new SrsFormat();
SrsAutoFree(SrsFormat, format);
if ((err = format->on_aac_sequence_header((char*)sh.c_str(), (int)sh.length())) != srs_success) {
return srs_error_wrap(err, "on aac sequence header");
}
SrsAudioCodecConfig* dec = format->acodec;
acodec->sound_format = SrsAudioCodecIdAAC;
acodec->sound_type = (dec->aac_channels == 2)? SrsAudioChannelsStereo : SrsAudioChannelsMono;
acodec->sound_size = SrsAudioSampleBits16bit;
acodec->aac_packet_type = 0;
static int srs_aac_srates[] = {
96000, 88200, 64000, 48000,
44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000,
7350, 0, 0, 0
};
switch (srs_aac_srates[dec->aac_sample_rate]) {
case 11025:
acodec->sound_rate = SrsAudioSampleRate11025;
break;
case 22050:
acodec->sound_rate = SrsAudioSampleRate22050;
break;
case 44100:
acodec->sound_rate = SrsAudioSampleRate44100;
break;
default:
break;
};
if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), acodec, (uint32_t)dts)) != srs_success) {
return srs_error_wrap(err, "write audio raw frame");
}
return err; return err;
} }
srs_error_t SrsRtspConn::write_h264_sps_pps(uint32_t dts, uint32_t pts) srs_error_t SrsRtspConn::write_h264_sps_pps(uint32_t dts, uint32_t pts)
{ {
srs_error_t err = srs_success; srs_error_t err = srs_success;
if (h264_sps.empty() || h264_pps.empty()) {
srs_warn("no sps=%dB or pps=%dB", (int)h264_sps.size(), (int)h264_pps.size());
return err;
}
// h264 raw to h264 packet. // h264 raw to h264 packet.
std::string sh; std::string sh;

Loading…
Cancel
Save