external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-03-09 15:49:59 +00:00
Code Issues Releases Wiki Activity

Support composited bridges for 1:N protocols converting. v6.0.41 (#3392)

Browse source 
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
This commit is contained in:
Winlin 2023-04-01 21:34:59 +08:00 • committed by GitHub
parent 771ae0a1a6
commit dcd02fe69c
No known key found for this signature in database
GPG key ID: 4AEE18F83AFDEB23
19 changed files with 770 additions and 400 deletions
Download patch file Download diff file Expand all files Collapse all files

86
trunk/src/app/srs_app_rtc_source.hpp
View file

@ -18,8 +18,10 @@
#include <srs_app_rtc_sdp.hpp>
#include <srs_protocol_st.hpp>
#include <srs_app_source.hpp>
#include <srs_kernel_rtc_rtp.hpp>
#include <srs_app_hourglass.hpp>
#include <srs_protocol_format.hpp>
#include <srs_app_stream_bridge.hpp>
class SrsRequest;
class SrsMetaCache;
@ -27,7 +29,7 @@ class SrsSharedPtrMessage;
class SrsCommonMessage;
class SrsMessageArray;
class SrsRtcSource;
class SrsRtcFromRtmpBridge;
class SrsFrameToRtcBridge;
class SrsAudioTranscoder;
class SrsRtpPacket;
class SrsSample;
@ -38,6 +40,13 @@ class SrsRtpRingBuffer;
class SrsRtpNackForReceiver;
class SrsJsonObject;
class SrsErrorPithyPrint;
class SrsRtcFrameBuilder;
class SrsLiveSource;
// Firefox defaults as 109, Chrome is 111.
const int kAudioPayloadType = 111;
// Firefox defaults as 126, Chrome is 102.
const int kVideoPayloadType = 102;
class SrsNtp
{
@ -145,18 +154,6 @@ public:
virtual void on_consumers_finished() = 0;
};
// SrsRtcSource bridge to SrsLiveSource
class ISrsRtcSourceBridge
{
public:
ISrsRtcSourceBridge();
virtual ~ISrsRtcSourceBridge();
public:
virtual srs_error_t on_publish() = 0;
virtual srs_error_t on_rtp(SrsRtpPacket *pkt) = 0;
virtual void on_unpublish() = 0;
};
// A Source is a stream, to publish and to play with, binding to SrsRtcPublishStream and SrsRtcPlayStream.
class SrsRtcSource : public ISrsFastTimer
{
@ -172,8 +169,13 @@ private:
ISrsRtcPublishStream* publish_stream_;
// Steam description for this steam.
SrsRtcSourceDescription* stream_desc_;
private:
#ifdef SRS_FFMPEG_FIT
// Collect and build WebRTC RTP packets to AV frames.
SrsRtcFrameBuilder* frame_builder_;
#endif
// The Source bridge, bridge stream to other source.
ISrsRtcSourceBridge* bridge_;
ISrsStreamBridge* bridge_;
private:
// To delivery stream to clients.
std::vector<SrsRtcConsumer*> consumers;
@ -205,7 +207,7 @@ public:
virtual SrsContextId source_id();
virtual SrsContextId pre_source_id();
public:
void set_bridge(ISrsRtcSourceBridge *bridge);
void set_bridge(ISrsStreamBridge* bridge);
public:
// Create consumer
// @param consumer, output the create consumer.
@ -245,60 +247,64 @@ private:
};
#ifdef SRS_FFMPEG_FIT
class SrsRtcFromRtmpBridge : public ISrsLiveSourceBridge
// Convert AV frame to RTC RTP packets.
class SrsRtcRtpBuilder
{
private:
SrsRequest* req;
SrsRtcSource* source_;
SrsFrameToRtcBridge* bridge_;
// The format, codec information.
SrsRtmpFormat* format;
// The metadata cache.
SrsMetaCache* meta;
private:
bool rtmp_to_rtc;
SrsAudioCodecId latest_codec_;
SrsAudioTranscoder* codec_;
bool keep_bframe;
bool merge_nalus;
uint16_t audio_sequence;
uint16_t video_sequence;
uint32_t audio_ssrc;
uint32_t video_ssrc;
private:
uint32_t audio_ssrc_;
uint32_t video_ssrc_;
uint8_t audio_payload_type_;
uint8_t video_payload_type_;
public:
SrsRtcFromRtmpBridge(SrsRtcSource* source);
virtual ~SrsRtcFromRtmpBridge();
SrsRtcRtpBuilder(SrsFrameToRtcBridge* bridge, uint32_t assrc, uint8_t apt, uint32_t vssrc, uint8_t vpt);
virtual ~SrsRtcRtpBuilder();
public:
virtual srs_error_t initialize(SrsRequest* r);
virtual srs_error_t on_publish();
virtual void on_unpublish();
virtual srs_error_t on_frame(SrsSharedPtrMessage* frame);
private:
virtual srs_error_t on_audio(SrsSharedPtrMessage* msg);
private:
srs_error_t init_codec(SrsAudioCodecId codec);
srs_error_t transcode(SrsAudioFrame* audio);
srs_error_t package_opus(SrsAudioFrame* audio, SrsRtpPacket* pkt);
public:
private:
virtual srs_error_t on_video(SrsSharedPtrMessage* msg);
private:
srs_error_t filter(SrsSharedPtrMessage* msg, SrsFormat* format, bool& has_idr, std::vector<SrsSample*>& samples);
srs_error_t package_stap_a(SrsRtcSource* source, SrsSharedPtrMessage* msg, SrsRtpPacket* pkt);
srs_error_t package_stap_a(SrsSharedPtrMessage* msg, SrsRtpPacket* pkt);
srs_error_t package_nalus(SrsSharedPtrMessage* msg, const std::vector<SrsSample*>& samples, std::vector<SrsRtpPacket*>& pkts);
srs_error_t package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, std::vector<SrsRtpPacket*>& pkts);
srs_error_t package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, std::vector<SrsRtpPacket*>& pkts);
srs_error_t consume_packets(std::vector<SrsRtpPacket*>& pkts);
};
class SrsRtmpFromRtcBridge : public ISrsRtcSourceBridge
// Collect and build WebRTC RTP packets to AV frames.
class SrsRtcFrameBuilder
{
private:
SrsLiveSource *source_;
ISrsStreamBridge* bridge_;
private:
bool is_first_audio_;
SrsAudioTranscoder *codec_;
bool is_first_audio;
bool is_first_video;
// The format, codec information.
SrsRtmpFormat* format;
private:
const static uint16_t s_cache_size = 512;
//TODO:use SrsRtpRingBuffer
//TODO:jitter buffer class
struct RtcPacketCache {
@ -308,33 +314,33 @@ private:
uint32_t rtp_ts;
SrsRtpPacket* pkt;
};
const static uint16_t s_cache_size = 512;
RtcPacketCache cache_video_pkts_[s_cache_size];
uint16_t header_sn_;
uint16_t lost_sn_;
int64_t rtp_key_frame_ts_;
public:
SrsRtmpFromRtcBridge(SrsLiveSource *src);
virtual ~SrsRtmpFromRtcBridge();
SrsRtcFrameBuilder(ISrsStreamBridge* bridge);
virtual ~SrsRtcFrameBuilder();
public:
srs_error_t initialize(SrsRequest* r);
public:
virtual srs_error_t on_publish();
virtual srs_error_t on_rtp(SrsRtpPacket *pkt);
virtual void on_unpublish();
virtual srs_error_t on_rtp(SrsRtpPacket *pkt);
private:
srs_error_t transcode_audio(SrsRtpPacket *pkt);
void packet_aac(SrsCommonMessage* audio, char* data, int len, uint32_t pts, bool is_header);
private:
srs_error_t packet_video(SrsRtpPacket* pkt);
srs_error_t packet_video_key_frame(SrsRtpPacket* pkt);
srs_error_t packet_video_rtmp(const uint16_t start, const uint16_t end);
int32_t find_next_lost_sn(uint16_t current_sn, uint16_t& end_sn);
void clear_cached_video();
inline uint16_t cache_index(uint16_t current_sn) {
return current_sn % s_cache_size;
}
int32_t find_next_lost_sn(uint16_t current_sn, uint16_t& end_sn);
bool check_frame_complete(const uint16_t start, const uint16_t end);
srs_error_t packet_video_rtmp(const uint16_t start, const uint16_t end);
void clear_cached_video();
};
#endif
// TODO: FIXME: Rename it.