srs推flv流_SRS流媒体服务器之HLS源码分析(3)

0.引言

阅读本文前，可以先阅读前面的文章，能够帮助你更好理解本篇文章。文章列表如下:

SRS流媒体服务器之HLS源码分析(1)

SRS流媒体服务器之HLS源码分析(2)

SRS流媒体服务器之HLS配置、测试和技术选型

SRS流媒体服务器集群之Edge模式(3)

SRS流媒体服务器集群之Edge模式(2)

SRS流媒体服务器集群之Edge模式(1)

SRS流媒体服务器集群之Forward模式(2)

SRS流媒体服务器集群之Forward模式(1)

SRS流媒体服务器之HTTP-FLV框架分析(1)

SRS流媒体服务器之RTMP推流消息处理(1)

SRS流媒体服务器之RTMP协议分析(2)

SRS流媒体框架分析(1)

SRS流媒体之RTMP推流框架分析(2)

SRS流媒体之RTMP拉流框架分析(3)

SRS流媒体服务器之RTMP协议分析(1)

简述SRS流媒体服务器相关技术

流媒体推拉流实战之RTMP协议分析(BAT面试官推荐)

流媒体服务器架构与应用分析

手把手搭建流媒体服务器详细步骤

手把手搭建FFmpeg的Windows环境

超详细手把手搭建在ubuntu系统的FFmpeg环境

本篇文章主要讲解的是在推流时，音频数据是如何通过RTMP协议转化为HLS的ts封装，视频也是类似。以源码的方式进行分析。

1.源码分析

SrsSharedPtrMessage是rtmp的消息格式。

srs_error_t SrsSource::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 all consumer    if (!drop_for_reduce) {        for (int i = 0; i < (int)consumers.size(); i++) {            SrsConsumer* consumer = consumers.at(i);            if ((err = consumer->enqueue(msg, atc, jitter_algorithm)) != srs_success) {                return srs_error_wrap(err, "consume message");            }        }    }        // Copy to hub to all utilities.    if ((err = hub->on_audio(msg)) != srs_success) {        return srs_error_wrap(err, "consume audio");    }        // 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;}

把Rtmp封装好的数据，转为音视频裸数据。

srs_error_t SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio){    srs_error_t err = srs_success;        SrsSharedPtrMessage* msg = shared_audio;    //转换封装格式    if ((err = format->on_audio(msg)) != srs_success) {        return srs_error_wrap(err, "format consume audio");    }        // cache the sequence header if aac    // donot cache the sequence header to gop_cache, return here.    if (format->is_aac_sequence_header()) {        srs_assert(format->acodec);        SrsAudioCodecConfig* c = format->acodec;                static int flv_sample_sizes[] = {8, 16, 0};        static int flv_sound_types[] = {1, 2, 0};                // when got audio stream info.        SrsStatistic* stat = SrsStatistic::instance();        if ((err = stat->on_audio_info(req, SrsAudioCodecIdAAC, c->sound_rate, c->sound_type, c->aac_object)) != srs_success) {            return srs_error_wrap(err, "stat audio");        }                srs_trace("%dB audio sh, codec(%d, profile=%s, %dchannels, %dkbps, %dHZ), flv(%dbits, %dchannels, %dHZ)",                  msg->size, c->id, srs_aac_object2str(c->aac_object).c_str(), c->aac_channels,                  c->audio_data_rate / 1000, srs_aac_sr