pjmedia系列之媒体流pjmedia_stream

前面讲过，一个会话可以有多个流，比如音频流、视频流，这里讲一个音频流有哪些东西。

媒体流

struct pjmedia_stream
{
    pjmedia_endpt	    *endpt;	    /**< Media endpoint.	    */
    pjmedia_codec_mgr	    *codec_mgr;	    /**< Codec manager instance.    */
    pjmedia_stream_info	     si;	    /**< Creation parameter.        */
    pjmedia_port	     port;	    /**< Port interface.	    */
    pjmedia_channel	    *enc;	    /**< Encoding channel.	    */
    pjmedia_channel	    *dec;	    /**< Decoding channel.	    */

    pjmedia_transport	    *transport;	    /**< Stream transport.	    */

    pjmedia_codec	    *codec;	    /**< Codec instance being used. */
    pjmedia_codec_param	     codec_param;   /**< Codec param.		    */


    pj_mutex_t		    *jb_mutex;
    pjmedia_jbuf	    *jb;	    /**< Jitter buffer.		    */

    pjmedia_rtcp_session     rtcp;	    /**< RTCP for incoming RTP.	    */
}

 此结构体很庞大，这里省略了大部分成员，其中endpt就是流的上一级端点，transport会指向上一节创建的传输对象。

媒体流创建

在示例程序simpleua.c中，当sip协商成功调用call_on_media_update，在里面会创建并启动流。

    /* Create new audio media stream, passing the stream info, and also the
     * media socket that we created earlier.
     */
    status = pjmedia_stream_create(g_med_endpt, inv->dlg->pool, &stream_info,
				   g_med_transport[0], NULL, &g_med_stream);

    /* Start the UDP media transport */
    pjmedia_transport_media_start(g_med_transport[0], 0, 0, 0, 0);

先看创建流

创建流的流程很多，这里抽取一些关键性代码

1、申请媒体流空间

2、初始化流的若干参数

3、codec管理者及codec相关的操作

4、设置第一组回调put_frame和get_frame，这组回调是音频设备要用的，下一节再讲。

5、创建jitterbuffer，这个后面会单独讲

6、创建编码通道和解码通道

7、调用上一节中提到的媒体传输attach，传入第2组回调on_rx_rtp和on_rx_rtcp

/*
 * Create media stream.
 */
PJ_DEF(pj_status_t) pjmedia_stream_create( pjmedia_endpt *endpt,
					   pj_pool_t *pool,
					   const pjmedia_stream_info *info,
					   pjmedia_transport *tp,
					   void *user_data,
					   pjmedia_stream **p_stream)

{
    pjmedia_stream *stream;
    pjmedia_transport_attach_param att_param;

    /* Allocate the media stream: */
    stream = PJ_POOL_ZALLOC_T(pool, pjmedia_stream);

    /* Init stream: */
    stream->endpt = endpt;
    stream->codec_mgr = pjmedia_endpt_get_codec_mgr(endpt);
    stream->user_data = user_data;

    /* Create and initialize codec: */
    status = pjmedia_codec_mgr_alloc_codec( stream->codec_mgr,
					    &info->fmt, &stream->codec);

	stream->port.put_frame = &put_frame;
	stream->port.get_frame = &get_frame;

    /* Create jitter buffer */
    status = pjmedia_jbuf_create(pool, &stream->port.info.name,
				 stream->frame_size,
				 stream->codec_param.info.frm_ptime,
				 jb_max, &stream->jb);


    /* Create decoder channel: */

    status = create_channel( pool, stream, PJMEDIA_DIR_DECODING,
			     info->rx_pt, info, &stream->dec);

    /* Create encoder channel: */
    status = create_channel( pool, stream, PJMEDIA_DIR_ENCODING,
			     info->tx_pt, info, &stream->enc);

    pj_bzero(&att_param, sizeof(att_param));
    att_param.stream = stream;
    att_param.media_type = PJMEDIA_TYPE_AUDIO;
    att_param.user_data = stream;
    pj_sockaddr_cp(&att_param.rem_addr, &info->rem_addr);
    pj_sockaddr_cp(&stream->rem_rtp_addr, &info->rem_addr);
    if (stream->si.rtcp_mux) {
	pj_sockaddr_cp(&att_param.rem_rtcp, &info->rem_addr);    	
    } else if (pj_sockaddr_has_addr(&info->rem_rtcp.addr)) {
	pj_sockaddr_cp(&att_param.rem_rtcp, &info->rem_rtcp);
    }
    att_param.addr_len = pj_sockaddr_get_len(&info->rem_addr);
    att_param.rtp_cb2 = &on_rx_rtp;
    att_param.rtcp_cb = &on_rx_rtcp;

    /* Only attach transport when stream is ready. */
    status = pjmedia_transport_attach2(tp, &att_param);
}

到这里就可以知道数据的流向，先从pjmedia_transport通过ioqueue接收到音频数据，然后通过几层回调，最终调用到stream的on_rx_rtp。

再看启动流，没有太多东西，就是设置两个标志位。

PJ_DEF(pj_status_t) pjmedia_stream_start(pjmedia_stream *stream)
{
    if (stream->enc && (stream->dir & PJMEDIA_DIR_ENCODING)) {
	stream->enc->paused = 0;
    }

    if (stream->dec && (stream->dir & PJMEDIA_DIR_DECODING)) {
	stream->dec->paused = 0;
    } 

    return PJ_SUCCESS;
}

on_rx_rtp

当音频数据流到on_rx_rtp时

1、更新RTP session参数

2、解码

3、插入到jitterbuffer

/*
 * This callback is called by stream transport on receipt of packets
 * in the RTP socket.
 */
static void on_rx_rtp( pjmedia_tp_cb_param *param)
{
    pjmedia_stream *stream = (pjmedia_stream*) param->user_data;
    void *pkt = param->pkt;
    pj_ssize_t bytes_read = param->size;
    pjmedia_channel *channel = stream->dec;
    const pjmedia_rtp_hdr *hdr;
    const void *payload;
    unsigned payloadlen;
    pjmedia_rtp_status seq_st;
    pj_status_t status;
    pj_bool_t pkt_discarded = PJ_FALSE;

    /* Update RTP and RTCP session. */
    status = pjmedia_rtp_decode_rtp(&channel->rtp, pkt, (int)bytes_read,
				    &hdr, &payload, &payloadlen);

    /* Update RTP session (also checks if RTP session can accept
     * the incoming packet.
     */
    pjmedia_rtp_session_update2(&channel->rtp, hdr, &seq_st,
			        hdr->pt != stream->rx_event_pt);

	/* Parse the payload. */
	status = pjmedia_codec_parse(stream->codec, (void*)payload,
				     payloadlen, &ts, &count, frames);

	/* Put each frame to jitter buffer. */
	for (i=0; i<count; ++i) {
	    unsigned ext_seq;
	    pj_bool_t discarded;

	    ext_seq = (unsigned)(frames[i].timestamp.u64 / ts_span);
	    pjmedia_jbuf_put_frame2(stream->jb, frames[i].buf, frames[i].size,
				    frames[i].bit_info, ext_seq, &discarded);
	    if (discarded)
		pkt_discarded = PJ_TRUE;
	}
}

总结

pjmedia_stream是pjmedia最复杂的结构体，大部分的操作都在这个对象里进行，其中两组回调是连接数据流的关键，一组跟网络绑定，一组跟设备绑定，使得整个媒体流可以串起来。本节先介绍这些，pjmedia_stream还有很多子模块，比如jitterbuffer，后面再单独介绍。