【音视频流媒体】SRS4.0 RTMP转WebRTC

1.启动SRS

./objs/srs -c conf/rtc.conf

配置文件：

listen              1935;
max_connections     1000;
daemon              off;
srs_log_tank        console;

http_server {
    enabled         on;
    listen          8080;
    dir             ./objs/nginx/html;
}

http_api {
    enabled         on;
    listen          1985;
}
stats {
    network         0;
}
rtc_server {
    enabled on;
    # Listen at udp://8000
    listen 8000;
    #
    # The $CANDIDATE means fetch from env, if not configed, use * as default.
    #
    # The * means retrieving server IP automatically, from all network interfaces,
    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate 这里需要配置外网IP
    candidate 192.168.1.103; 
}

推流：

ffmpeg -re -i time.flv -vcodec copy -acodec copy -f flv -y rtmp://192.168.1.103/live/livestream

播放：

http:///192.168.1.103:8080/players/rtc_player.html

2.代码框架

这里主要分为几个部分：

rtmp推流到SRS

RTMP流转为RTC流

RTC客户端和SRS通过HTTP交互SDP信息

RTC客户端通过RTP拉流

RTMP推流到SRS

SrsLiveSource：代表RTMP源

SRS针对推流会创建专门的SrsLiveSource来表示源，本章主要分析一下源的创建和数据流的接受

创建Source

rtmp推流的时候就会创建SrsLiveSource和SrsRtcSource；

SrsRtmpConn::stream_service_cycle()

srs_error_t SrsRtmpConn::stream_service_cycle()
{
    srs_error_t err = srs_success;
    
    SrsRequest* req = info->req;
    
    if ((err = rtmp->identify_client(info->res->stream_id, info->type, req->stream, req->duration)) != srs_success) {
        return srs_error_wrap(err, "rtmp: identify client");
    }
    
    srs_discovery_tc_url(req->tcUrl, req->schema, req->host, req->vhost, req->app, req->stream, req->port, req->param);
    req->strip();
    srs_trace("client identified, type=%s, vhost=%s, app=%s, stream=%s, param=%s, duration=%dms",
        srs_client_type_string(info->type).c_str(), req->vhost.c_str(), req->app.c_str(), req->stream.c_str(), req->param.c_str(), srsu2msi(req->duration));
    
    // discovery vhost, resolve the vhost from config
    SrsConfDirective* parsed_vhost = _srs_config->get_vhost(req->vhost);
    if (parsed_vhost) {
        req->vhost = parsed_vhost->arg0();
    }

    if (req->schema.empty() || req->vhost.empty() || req->port == 0 || req->app.empty()) {
        return srs_error_new(ERROR_RTMP_REQ_TCURL, "discovery tcUrl failed, tcUrl=%s, schema=%s, vhost=%s, port=%d, app=%s",
            req->tcUrl.c_str(), req->schema.c_str(), req->vhost.c_str(), req->port, req->app.c_str());
    }

    // check vhost, allow default vhost.
    if ((err = check_vhost(true)) != srs_success) {
        return srs_error_wrap(err, "check vhost");
    }

    srs_trace("connected stream, tcUrl=%s, pageUrl=%s, swfUrl=%s, schema=%s, vhost=%s, port=%d, app=%s, stream=%s, param=%s, args=%s",
        req->tcUrl.c_str(), req->pageUrl.c_str(), req->swfUrl.c_str(), req->schema.c_str(), req->vhost.c_str(), req->port,
        req->app.c_str(), req->stream.c_str(), req->param.c_str(), (req->args? "(obj)":"null"));
    
    // do token traverse before serve it.
    // @see https://github.com/ossrs/srs/pull/239
    if (true) {
        info->edge = _srs_config->get_vhost_is_edge(req->vhost);
        bool edge_traverse = _srs_config->get_vhost_edge_token_traverse(req->vhost);
        if (info->edge && edge_traverse) {
            if ((err = check_edge_token_traverse_auth()) != srs_success) {
                return srs_error_wrap(err, "rtmp: check token traverse");
            }
        }
    }

    // security check
    if ((err = security->check(info->type, ip, req)) != srs_success) {
        return srs_error_wrap(err, "rtmp: security check");
    }
    
    // Never allow the empty stream name, for HLS may write to a file with empty name.
    // @see https://github.com/ossrs/srs/issues/834
    if (req->stream.empty()) {
        return srs_error_new(ERROR_RTMP_STREAM_NAME_EMPTY, "rtmp: empty stream");
    }

    // client is identified, set the timeout to service timeout.
    rtmp->set_recv_timeout(SRS_CONSTS_RTMP_TIMEOUT);
    rtmp->set_send_timeout(SRS_CONSTS_RTMP_TIMEOUT);
    
    // find a source to serve.
    SrsLiveSource* source = NULL;
    if ((err = _srs_sources->fetch_or_create(req, server, &source)) != srs_success) {
        return srs_error_wrap(err, "rtmp: fetch source");
    }
    srs_assert(source != NULL);
    
    // update the statistic when source disconveried.
    SrsStatistic* stat = SrsStatistic::instance();
    if ((err = stat->on_client(_srs_context->get_id().c_str(), req, this, info->type)) != srs_success) {
        return srs_error_wrap(err, "rtmp: stat client");
    }
    
    bool enabled_cache = _srs_config->get_gop_cache(req->vhost);
    srs_trace("source url=%s, ip=%s, cache=%d, is_edge=%d, source_id=%s/%s",
        req->get_stream_url().c_str(), ip.c_str(), enabled_cache, info->edge, source->source_id().c_str(), source->pre_source_id().c_str());
    source->set_cache(enabled_cache);
    
    switch (info->type) {
        case SrsRtmpConnPlay: {
            // response connection start play
            if ((err = rtmp->start_play(info->res->stream_id)) != srs_success) {
                return srs_error_wrap(err, "rtmp: start play");
            }
            if ((err = http_hooks_on_play()) != srs_success) {
                return srs_error_wrap(err, "rtmp: callback on play");
            }
            
            err = playing(source);
            http_hooks_on_stop();
            
            return err;
        }
        case SrsRtmpConnFMLEPublish: {
            if ((err = rtmp->start_fmle_publish(info->res->stream_id)) != srs_success) {
                return srs_error_wrap(err, "rtmp: start FMLE publish");
            }
            
            return publishing(source);
        }
        case SrsRtmpConnHaivisionPublish: {
            if ((err = rtmp->start_haivision_publish(info->res->stream_id)) != srs_success) {
                return srs_error_wrap(err, "rtmp: start HAIVISION publish");
            }
            
            return publishing(source);
        }
        case SrsRtmpConnFlashPublish: {
            if ((err = rtmp->start_flash_publish(info->res->stream_id)) != srs_success) {
                return srs_error_wrap(err, "rtmp: start FLASH publish");
            }
            
            return publishing(source);
        }
        default: {
            return srs_error_new(ERROR_SYSTEM_CLIENT_INVALID, "rtmp: unknown client type=%d", info->type);
        }
    }
    
    return err;
}

这里RTMP的业务处理中心，比较重要，推流和播放都在这里处理，同时SrsLiveSource也是在这里创建的。

数据接受

我们在source的音频回调这里打一下断点，看一下调用栈，就可以知道调用顺序

(gdb) bt
#0  SrsLiveSource::on_audio (this=0xfcd7a0, shared_audio=0x1067b80) at src/app/srs_app_source.cpp:2140
#1  0x00000000004f2241 in SrsRtmpConn::process_publish_message (this=0xfcd890, source=0xfcd7a0, msg=0x1067b80) at src/app/srs_app_rtmp_conn.cpp:1055
#2  0x00000000004f20ef in SrsRtmpConn::handle_publish_message (this=0xfcd890, source=0xfcd7a0, msg=0x1067b80) at src/app/srs_app_rtmp_conn.cpp:1034
#3  0x00000000005a0d06 in SrsPublishRecvThread::consume (this=0x10060a0, msg=0x1067b80) at src/app/srs_app_recv_thread.cpp:376
#4  0x000000000059fe7e in SrsRecvThread::do_cycle (this=0x10060c0) at src/app/srs_app_recv_thread.cpp:133
#5  0x000000000059fcea in SrsRecvThread::cycle (this=0x10060c0) at src/app/srs_app_recv_thread.cpp:102
#6  0x000000000051fe11 in SrsFastCoroutine::cycle (this=0x10679c0) at src/app/srs_app_st.cpp:253
#7  0x000000000051fe94 in SrsFastCoroutine::pfn (arg=0x10679c0) at src/app/srs_app_st.cpp:268
#8  0x00000000006346e8 in _st_thread_main () at sched.c:363
#9  0x0000000000634f5b in st_thread_create (start=0x1005daf, arg=0x10679a0, joinable=0, stk_size=16801184) at sched.c:694

SrsRecvThread::do_cycle()

srs_error_t SrsRecvThread::do_cycle()
{
    srs_error_t err = srs_success;
    
    while (true) {
        if ((err = trd->pull()) != srs_success) {
            return srs_error_wrap(err, "recv thread");
        }
        
        // When the pumper is interrupted, wait then retry.
        if (pumper->interrupted()) {
            srs_usleep(timeout);
            continue;
        }
        
        SrsCommonMessage* msg = NULL;
        
        // Process the received message.
        if ((err = rtmp->recv_message(&msg)) == srs_success) {
            err = pumper->consume(msg);
        }
        
        if (err != srs_success) {
            // Interrupt the receive thread for any error.
            trd->interrupt();
            
            // Notify the pumper to quit for error.
            pumper->interrupt(err);
            
            return srs_error_wrap(err, "recv thread");
        }
    }
    
    return err;
}

SRS有个专门的协程处理数据接受，主要关注两个rtmp->recv_message(&msg)和pumper->consume(msg)

rtmp->recv_message(&msg)：用来接受数据，最终调用SrsProtocol::recv_interlaced_message(SrsCommonMessage** pmsg)

pumper->consume(msg)：用来处理数据

SrsProtocol::recv_interlaced_message(SrsCommonMessage** pmsg)

srs_error_t SrsProtocol::read_message_payload(SrsChunkStream* chunk, SrsCommonMessage** pmsg)
{
    srs_error_t err = srs_success;
    
    // empty message
    if (chunk->header.payload_length <= 0) {
        srs_trace("get an empty RTMP message(type=%d, size=%d, time=%" PRId64 ", sid=%d)", chunk->header.message_type,
                  chunk->header.payload_length, chunk->header.timestamp, chunk->header.stream_id);
        
        *pmsg = chunk->msg;
        chunk->msg = NULL;
        
        return err;
    }
    srs_assert(chunk->header.payload_length > 0);
    
    // the chunk payload size.
    int payload_size = chunk->header.payload_length - chunk->msg->size;
    payload_size = srs_min(payload_size, in_chunk_size);
    
    // create msg payload if not initialized
    if (!chunk->msg->payload) {
        chunk->msg->create_payload(chunk->header.payload_length);
    }
    
    // read payload to buffer
    if ((err = in_buffer->grow(skt, payload_size)) != srs_success) {
        return srs_error_wrap(err, "read %d bytes payload", payload_size);
    }
    memcpy(chunk->msg->payload + chunk->msg->size, in_buffer->read_slice(payload_size), payload_size);
    chunk->msg->size += payload_size;
    
    // got entire RTMP message?
    if (chunk->header.payload_length == chunk->msg->size) {
        *pmsg = chunk->msg;
        chunk->msg = NULL;
        return err;
    }
    
    return err;
}

这里分别拷贝rtmp的header和body给SrsCommonMessage，到这里我们就算拿到rtmp的数据了。

音频的话，rtmp的header的TypeID应该为8，即rtmpt.header.typeid == 0x08。

而body是按照flv的格式组织的，相当于flv的音频tag：

• 第⼀个字节包含了⾳频数据的参数信息，
• 第⼆个字节开始为⾳频流数据。

我们抓一个音频的rtmp包分析一下：

AudioTagHeader格式：

处理数据

通过pumper->consume跟踪，发现process_publish_message()用来处理rtmp的message。

srs_error_t SrsRtmpConn::process_publish_message(SrsLiveSource* source, SrsCommonMessage* msg)
{
    srs_error_t err = srs_success;
    
    // for edge, directly proxy message to origin.
    if (info->edge) {
        if ((err = source->on_edge_proxy_publish(msg)) != srs_success) {
            return srs_error_wrap(err, "rtmp: proxy publish");
        }
        return err;
    }
    
    // process audio packet 语音包 typeid = 8
    if (msg->header.is_audio()) {
        if ((err = source->on_audio(msg)) != srs_success) {
            return srs_error_wrap(err, "rtmp: consume audio");
        }
        return err;
    }
    // process video packet 视频包 typeid = 9
    if (msg->header.is_video()) {
        if ((err = source->on_video(msg)) != srs_success) {
            return srs_error_wrap(err, "rtmp: consume video");
        }
        return err;
    }
    
    // process aggregate packet 统计消息 typeid = 22
    if (msg->header.is_aggregate()) {
        if ((err = source->on_aggregate(msg)) != srs_success) {
            return srs_error_wrap(err, "rtmp: consume aggregate");
        }
        return err;
    }
    
    // process onMetaData 数据消息 typed = 18 | 15 
    if (msg->header.is_amf0_data() || msg->header.is_amf3_data()) {
        SrsPacket* pkt = NULL;
        if ((err = rtmp->decode_message(msg, &pkt)) != srs_success) {
            return srs_error_wrap(err, "rtmp: decode message");
        }
        SrsAutoFree(SrsPacket, pkt);
        
        if (dynamic_cast<SrsOnMetaDataPacket*>(pkt)) {
            SrsOnMetaDataPacket* metadata = dynamic_cast<SrsOnMetaDataPacket*>(pkt);
            if ((err = source->on_meta_data(msg, metadata)) != srs_success) {
                return srs_error_wrap(err, "rtmp: consume metadata");
            }
            return err;
        }
        return err;
    }
    
    return err;
}

这里分别处理语音、视频等各种消息。

RTMP协议转WebRTC

SrsLiveSource：代表RTMP源

SrsRtcSource：代表RTC源

两个协议的转换，就是怎么从SrsLiveSource转到SrsRtcSource，这里需要一个桥梁SrsRtcFromRtmpBridger

创建Source

rtmp推流的时候就会创建SrsLiveSource和SrsRtcSource；

SrsRtmpConn::stream_service_cycle()

srs_error_t SrsRtmpConn::stream_service_cycle()
{
    srs_error_t err = srs_success;
    
    SrsRequest* req = info->req;
    
    if ((err = rtmp->identify_client(info->res->stream_id, info->type, req->stream, req->duration)) != srs_success) {
        return srs_error_wrap(err, "rtmp: identify client");
    }
    
    srs_discovery_tc_url(req->tcUrl, req->schema, req->host, req->vhost, req->app, req->stream, req->port, req->param);
    req->strip();
    srs_trace("client identified, type=%s, vhost=%s, app=%s, stream=%s, param=%s, duration=%dms",
        srs_client_type_string(info->type).c_str(), req->vhost.c_str(), req->app.c_str(), req->stream.c_str(), req->param.c_str(), srsu2msi(req->duration));
    
    // discovery vhost, resolve the vhost from config
    SrsConfDirective* parsed_vhost = _srs_config->get_vhost(req->vhost);
    if (parsed_vhost) {
        req->vhost = parsed_vhost->arg0();
    }

    if (req->schema.empty() || req->vhost.empty() || req->port == 0 || req->app.empty()) {
        return srs_error_new(ERROR_RTMP_REQ_TCURL, "discovery tcUrl failed, tcUrl=%s, schema=%s, vhost=%s, port=%d, app=%s",
            req->tcUrl.c_str(), req->schema.c_str(), req->vhost.c_str(), req->port, req->app.c_str());
    }

    // check vhost, allow default vhost.
    if ((err = check_vhost(true)) != srs_success) {
        return srs_error_wrap(err, "check vhost");
    }

    srs_trace("connected stream, tcUrl=%s, pageUrl=%s, swfUrl=%s, schema=%s, vhost=%s, port=%d, app=%s, stream=%s, param=%s, args=%s",
        req->tcUrl.c_str(), req->pageUrl.c_str(), req->swfUrl.c_str(), req->schema.c_str(), req->vhost.c_str(), req->port,
        req->app.c_str(), req->stream.c_str(), req->param.c_str(), (req->args? "(obj)":"null"));
    
    // do token traverse before serve it.
    // @see https://github.com/ossrs/srs/pull/239
    if (true) {
        info->edge = _srs_config->get_vhost_is_edge(req->vhost);
        bool edge_traverse = _srs_config->get_vhost_edge_token_traverse(req->vhost);
        if (info->edge && edge_traverse) {
            if ((err = check_edge_token_traverse_auth()) != srs_success) {
                return srs_error_wrap(err, "rtmp: check token traverse");
            }
        }
    }

    // security check
    if ((err = security->check(info->type, ip, req)) != srs_success) {
        return srs_error_wrap(err, "rtmp: security check");
    }
    
    // Never allow the empty stream name, for HLS may write to a file with empty name.
    // @see https://github.com/ossrs/srs/issues/834
    if (req->stream.empty()) {
        return srs_error_new(ERROR_RTMP_STREAM_NAME_EMPTY, "rtmp: empty stream");
    }

    // client is identified, set the timeout to service timeout.
    rtmp->set_recv_timeout(SRS_CONSTS_RTMP_TIMEOUT);
    rtmp->set_send_timeout(SRS_CONSTS_RTMP_TIMEOUT);
    
    // find a source to serve.
    SrsLiveSource* source = NULL;
    if ((err = _srs_sources->fetch_or_create(req, server, &source)) != srs_success) {
        return srs_error_wrap(err, "rtmp: fetch source");
    }
    srs_assert(source != NULL);
    
    // update the statistic when source disconveried.
    SrsStatistic* stat = SrsStatistic::instance();
    if ((err = stat->on_client(_srs_context->get_id().c_str(), req, this, info->type)) != srs_success) {
        return srs_error_wrap(err, "rtmp: stat client");
    }
    
    bool enabled_cache = _srs_config->get_gop_cache(req->vhost);
    srs_trace("source url=%s, ip=%s, cache=%d, is_edge=%d, source_id=%s/%s",
        req->get_stream_url().c_str(), ip.c_str(), enabled_cache, info->edge, source->source_id().c_str(), source->pre_source_id().c_str());
    source->set_cache(enabled_cache);
    
    switch (info->type) {
        case SrsRtmpConnPlay: {
            // response connection start play
            if ((err = rtmp->start_play(info->res->stream_id)) != srs_success) {
                return srs_error_wrap(err, "rtmp: start play");
            }
            if ((err = http_hooks_on_play()) != srs_success) {
                return srs_error_wrap(err, "rtmp: callback on play");
            }
            
            err = playing(source);
            http_hooks_on_stop();
            
            return err;
        }
        case SrsRtmpConnFMLEPublish: {
            if ((err = rtmp->start_fmle_publish(info->res->stream_id)) != srs_success) {
                return srs_error_wrap(err, "rtmp: start FMLE publish");
            }
            
            return publishing(source);
        }
        case SrsRtmpConnHaivisionPublish: {
            if ((err = rtmp->start_haivision_publish(info->res->stream_id)) != srs_success) {
                return srs_error_wrap(err, "rtmp: start HAIVISION publish");
            }
            
            return publishing(source);
        }
        case SrsRtmpConnFlashPublish: {
            if ((err = rtmp->start_flash_publish(info->res->stream_id)) != srs_success) {
                return srs_error_wrap(err, "rtmp: start FLASH publish");
            }
            
            return publishing(source);
        }
        default: {
            return srs_error_new(ERROR_SYSTEM_CLIENT_INVALID, "rtmp: unknown client type=%d", info->type);
        }
    }
    
    return err;
}

这里推流的业务处理，是会创建SrsLiveSource，然后我们看看publishing

SrsRtmpConn::publishing(SrsLiveSource* source)

srs_error_t SrsRtmpConn::publishing(SrsLiveSource* source)
{
    srs_error_t err = srs_success;
    
    SrsRequest* req = info->req;
    
    if (_srs_config->get_refer_enabled(req->vhost)) {
        if ((err = refer->check(req->pageUrl, _srs_config->get_refer_publish(req->vhost))) != srs_success) {
            return srs_error_wrap(err, "rtmp: referer check");
        }
    }
    
    if ((err = http_hooks_on_publish()) != srs_success) {
        return srs_error_wrap(err, "rtmp: callback on publish");
    }
    
    // TODO: FIXME: Should refine the state of publishing.
    if ((err = acquire_publish(source)) == srs_success) {
        // use isolate thread to recv,
        // @see: https://github.com/ossrs/srs/issues/237
        SrsPublishRecvThread rtrd(rtmp, req, srs_netfd_fileno(stfd), 0, this, source, _srs_context->get_id());
        err = do_publishing(source, &rtrd);
        rtrd.stop();
    }
    
    // whatever the acquire publish, always release publish.
    // when the acquire error in the midlle-way, the publish state changed,
    // but failed, so we must cleanup it.
    // @see https://github.com/ossrs/srs/issues/474
    // @remark when stream is busy, should never release it.
    if (srs_error_code(err) != ERROR_SYSTEM_STREAM_BUSY) {
        release_publish(source);
    }
    
    http_hooks_on_unpublish();
    
    return err;
}

这里先看看acquire_publish处理

srs_error_t SrsRtmpConn::acquire_publish(SrsLiveSource* source)
{
    srs_error_t err = srs_success;
    
    SrsRequest* req = info->req;

    // Check whether RTMP stream is busy.
    if (!source->can_publish(info->edge)) {
        return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtmp: stream %s is busy", req->get_stream_url().c_str());
    }
    
    // Check whether RTC stream is busy.
#ifdef SRS_RTC
    SrsRtcSource *rtc = NULL;
    bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
    bool rtc_enabled = _srs_config->get_rtc_enabled(req->vhost);
    if (rtc_server_enabled && rtc_enabled && !info->edge) {
        if ((err = _srs_rtc_sources->fetch_or_create(req, &rtc)) != srs_success) {
            return srs_error_wrap(err, "create source");
        }

        if (!rtc->can_publish()) {
            return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtc stream %s busy", req->get_stream_url().c_str());
        }
    }
#endif

    // Bridge to RTC streaming.
#if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
    if (rtc) {
        SrsRtcFromRtmpBridger *bridger = new SrsRtcFromRtmpBridger(rtc);
        if ((err = bridger->initialize(req)) != srs_success) {
            srs_freep(bridger);
            return srs_error_wrap(err, "bridger init");
        }

        source->set_bridger(bridger);
    }
#endif

    // Start publisher now.
    if (info->edge) {
        return source->on_edge_start_publish();
    } else {
        return source->on_publish();
    }
}

可以看出，如果配置了RTC，就会创建SrsRtcSource和SrsRtcFromRtmpBridger。

获取RTMP数据

SrsRtmpConn::process_publish_message(SrsLiveSource* source, SrsCommonMessage* msg)

srs_error_t SrsRtmpConn::process_publish_message(SrsLiveSource* source, SrsCommonMessage* msg)
{
    srs_error_t err = srs_success;
    
    // for edge, directly proxy message to origin.
    if (info->edge) {
        if ((err = source->on_edge_proxy_publish(msg)) != srs_success) {
            return srs_error_wrap(err, "rtmp: proxy publish");
        }
        return err;
    }
    
    // process audio packet
    if (msg->header.is_audio()) {
        if ((err = source->on_audio(msg)) != srs_success) {
            return srs_error_wrap(err, "rtmp: consume audio");
        }
        return err;
    }
    // process video packet
    if (msg->header.is_video()) {
        if ((err = source->on_video(msg)) != srs_success) {
            return srs_error_wrap(err, "rtmp: consume video");
        }
        return err;
    }
    
    // process aggregate packet
    if (msg->header.is_aggregate()) {
        if ((err = source->on_aggregate(msg)) != srs_success) {
            return srs_error_wrap(err, "rtmp: consume aggregate");
        }
        return err;
    }
    
    // process onMetaData
    if (msg->header.is_amf0_data() || msg->header.is_amf3_data()) {
        SrsPacket* pkt = NULL;
        if ((err = rtmp->decode_message(msg, &pkt)) != srs_success) {
            return srs_error_wrap(err, "rtmp: decode message");
        }
        SrsAutoFree(SrsPacket, pkt);
        
        if (dynamic_cast<SrsOnMetaDataPacket*>(pkt)) {
            SrsOnMetaDataPacket* metadata = dynamic_cast<SrsOnMetaDataPacket*>(pkt);
            if ((err = source->on_meta_data(msg, metadata)) != srs_success) {
                return srs_error_wrap(err, "rtmp: consume metadata");
            }
            return err;
        }
        return err;
    }
    
    return err;
}

这里处理收到的源数据，下面分析一下音频的处理

srs_error_t SrsLiveSource::on_audio(SrsCommonMessage* shared_audio)
{
    srs_error_t err = srs_success;
    
    // monotically increase detect.
    if (!mix_correct && is_monotonically_increase) {
        if (last_packet_time > 0 && shared_audio->header.timestamp < last_packet_time) {
            is_monotonically_increase = false;
            srs_warn("AUDIO: stream not monotonically increase, please open mix_correct.");
        }
    }
    last_packet_time = shared_audio->header.timestamp;
    
    // convert shared_audio to msg, user should not use shared_audio again.
    // the payload is transfer to msg, and set to NULL in shared_audio.
    SrsSharedPtrMessage msg;
    if ((err = msg.create(shared_audio)) != srs_success) {
        return srs_error_wrap(err, "create message");
    }
    
    // directly process the audio message. 
    if (!mix_correct) {
        return on_audio_imp(&msg);
    }
    
    // insert msg to the queue.
    mix_queue->push(msg.copy());
    
    // fetch someone from mix queue.
    SrsSharedPtrMessage* m = mix_queue->pop();
    if (!m) {
        return err;
    }
    
    // consume the monotonically increase message.
    if (m->is_audio()) {
        err = on_audio_imp(m);
    } else {
        err = on_video_imp(m);
    }
    srs_freep(m);
    
    return err;
}

SrsLiveSource::on_audio_imp(SrsSharedPtrMessage* msg)

srs_error_t SrsLiveSource::on_audio_imp(SrsSharedPtrMessage* msg)
{
    srs_error_t err = srs_success;
    
    bool is_aac_sequence_header = SrsFlvAudio::sh(msg->payload, msg->size);
    bool is_sequence_header = is_aac_sequence_header;
    
    // whether consumer should drop for the duplicated sequence header.
    bool drop_for_reduce = false;
    if (is_sequence_header && meta->previous_ash() && _srs_config->get_reduce_sequence_header(req->vhost)) {
        if (meta->previous_ash()->size == msg->size) {
            drop_for_reduce = srs_bytes_equals(meta->previous_ash()->payload, msg->payload, msg->size);
            srs_warn("drop for reduce sh audio, size=%d", msg->size);
        }
    }
    
    // Copy to hub to all utilities.
    if ((err = hub->on_audio(msg)) != srs_success) {
        return srs_error_wrap(err, "consume audio");
    }

    // For bridger to consume the message.
    if (bridger_ && (err = bridger_->on_audio(msg)) != srs_success) {
        return srs_error_wrap(err, "bridger consume audio");
    }

    // copy to all consumer
    if (!drop_for_reduce) {
        for (int i = 0; i < (int)consumers.size(); i++) {
            SrsLiveConsumer* consumer = consumers.at(i);
            if ((err = consumer->enqueue(msg, atc, jitter_algorithm)) != srs_success) {
                return srs_error_wrap(err, "consume message");
            }
        }
    }
    
    // cache the sequence header of aac, or first packet of mp3.
    // for example, the mp3 is used for hls to write the "right" audio codec.
    // TODO: FIXME: to refine the stream info system.
    if (is_aac_sequence_header || !meta->ash()) {
        if ((err = meta->update_ash(msg)) != srs_success) {
            return srs_error_wrap(err, "meta consume audio");
        }
    }
    
    // when sequence header, donot push to gop cache and adjust the timestamp.
    if (is_sequence_header) {
        return err;
    }
    
    // cache the last gop packets
    if ((err = gop_cache->cache(msg)) != srs_success) {
        return srs_error_wrap(err, "gop cache consume audio");
    }

    // if atc, update the sequence header to abs time.
    if (atc) {
        if (meta->ash()) {
            meta->ash()->timestamp = msg->timestamp;
        }
        if (meta->data()) {
            meta->data()->timestamp = msg->timestamp;
        }
    }
    
    return err;
}

这里把数据转发给SrsRtcFromRtmpBridger,上面都比较简单，下面着重分析一下SrsRtcFromRtmpBridger收到数据以后的语音转码

转码：aac转opus