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RTC: Fix rtmp to rtc bug

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This commit is contained in:
winlin 2020-07-19 10:26:05 +08:00
parent 7ad1dfbbb8
commit 6dd77923ca
3 changed files with 119 additions and 22 deletions
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87
trunk/src/app/srs_app_rtc_source.cpp
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@ -23,6 +23,8 @@
#include <srs_app_rtc_source.hpp>
#include <unistd.h>
#include <srs_app_conn.hpp>
#include <srs_rtmp_stack.hpp>
#include <srs_app_config.hpp>
@ -37,13 +39,18 @@
#include <srs_core_autofree.hpp>
#include <srs_app_rtc_queue.hpp>
#include <srs_app_rtc_conn.hpp>
#include <srs_protocol_utility.hpp>
#ifdef SRS_FFMPEG_FIT
#include <srs_app_rtc_codec.hpp>
#endif
const int kChannel = 2;
const int kSamplerate = 48000;
const int kAudioPayloadType = 111;
const int kAudioChannel = 2;
const int kAudioSamplerate = 48000;
const int kVideoPayloadType = 102;
const int kVideoSamplerate = 90000;
// An AAC packet may be transcoded to many OPUS packets.
const int kMaxOpusPackets = 8;
@ -284,6 +291,7 @@ SrsRtcStream::SrsRtcStream()
{
_can_publish = true;
publish_stream_ = NULL;
stream_desc_ = NULL;
req = NULL;
#ifdef SRS_FFMPEG_FIT
@ -291,7 +299,6 @@ SrsRtcStream::SrsRtcStream()
#else
bridger_ = new SrsRtcDummyBridger();
#endif
stream_desc_ = NULL;
}
SrsRtcStream::~SrsRtcStream()
@ -498,7 +505,7 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcStream* source)
req = NULL;
source_ = source;
format = new SrsRtmpFormat();
codec = new SrsAudioRecode(kChannel, kSamplerate);
codec = new SrsAudioRecode(kAudioChannel, kAudioSamplerate);
discard_aac = false;
discard_bframe = false;
merge_nalus = false;
@ -506,6 +513,39 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcStream* source)
audio_timestamp = 0;
audio_sequence = 0;
video_sequence = 0;
SrsRtcStreamDescription* stream_desc = new SrsRtcStreamDescription();
SrsAutoFree(SrsRtcStreamDescription, stream_desc);
// audio track description
if (true) {
SrsRtcTrackDescription* audio_track_desc = new SrsRtcTrackDescription();
audio_track_desc->type_ = "audio";
audio_track_desc->id_ = "audio-" + srs_random_str(8);
audio_ssrc = SrsRtcSSRCGenerator::instance()->generate_ssrc();
audio_track_desc->ssrc_ = audio_ssrc;
audio_track_desc->direction_ = "recvonly";
audio_track_desc->media_ = new SrsAudioPayload(kAudioPayloadType, "opus", kAudioSamplerate, kAudioChannel);
stream_desc->audio_track_desc_ = audio_track_desc->copy();
}
// video track description
if (true) {
SrsRtcTrackDescription* video_track_desc = new SrsRtcTrackDescription();
video_track_desc->type_ = "video";
video_track_desc->id_ = "video-" + srs_random_str(8);
video_ssrc = SrsRtcSSRCGenerator::instance()->generate_ssrc();
video_track_desc->ssrc_ = video_ssrc;
video_track_desc->direction_ = "recvonly";
video_track_desc->media_ = new SrsVideoPayload(kVideoPayloadType, "H264", kVideoSamplerate);
stream_desc->video_track_descs_.push_back(video_track_desc->copy());
}
source_->set_stream_desc(stream_desc);
}
SrsRtcFromRtmpBridger::~SrsRtcFromRtmpBridger()
@ -672,6 +712,8 @@ srs_error_t SrsRtcFromRtmpBridger::package_opus(char* data, int size, SrsRtpPack
srs_error_t err = srs_success;
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->header.set_payload_type(kAudioPayloadType);
pkt->header.set_ssrc(audio_ssrc);
pkt->frame_type = SrsFrameTypeAudio;
pkt->header.set_marker(true);
pkt->header.set_sequence(audio_sequence++);
@ -813,6 +855,8 @@ srs_error_t SrsRtcFromRtmpBridger::package_stap_a(SrsRtcStream* source, SrsShare
}
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_marker(false);
pkt->header.set_sequence(video_sequence++);
@ -884,6 +928,8 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
if (nn_bytes < kRtpMaxPayloadSize) {
// Package NALUs in a single RTP packet.
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
@ -914,6 +960,8 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
}
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
@ -940,6 +988,8 @@ srs_error_t SrsRtcFromRtmpBridger::package_single_nalu(SrsSharedPtrMessage* msg,
srs_error_t err = srs_success;
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
@ -970,6 +1020,8 @@ srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSam
int packet_size = srs_min(nb_left, fu_payload_size);
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
@ -1700,3 +1752,30 @@ srs_error_t SrsRtcVideoSendTrack::on_rtcp(SrsRtpPacket2* pkt)
return err;
}
SrsRtcSSRCGenerator* SrsRtcSSRCGenerator::_instance = NULL;
SrsRtcSSRCGenerator::SrsRtcSSRCGenerator()
{
ssrc_num = 0;
}
SrsRtcSSRCGenerator::~SrsRtcSSRCGenerator()
{
}
SrsRtcSSRCGenerator* SrsRtcSSRCGenerator::instance()
{
if (!_instance) {
_instance = new SrsRtcSSRCGenerator();
}
return _instance;
}
uint32_t SrsRtcSSRCGenerator::generate_ssrc()
{
if (!ssrc_num) {
ssrc_num = ::getpid() * 10000 + ::getpid() * 100 + ::getpid();
}
return ++ssrc_num;
}