将RTSP流录制为mp4文件

将RTSP流录制为mp4文件

录制程序要继续添加新功能：模拟电视，板卡发送出来的是rtsp流（h264视频+alaw(pcma)音频）。

由于之前做过将rtp流（h264视频+aac音频）录制合成mp4文件（参见http://www.cnblogs.com/chutianyao/archive/2012/04/13/2446140.html），很自然的就决定将其合成为mp4文件。

 

但是有些不同：

（1）需要解析RTSP协议。研究了一下RFC2326，发现也不是很复杂。

　　rtsp分控制流和数据流：控制流就是客户端向服务端发送控制命令，包括查看节目信息、播放、停止节目等，一般是通过TCP协议通信的；数据流就是服务端将音视频数据发送到指定的地址、端口上，我们的音频和视频单独发送到两个不同的端口上，采用的是UDP协议。采用TCP或UDP，在RTSP协议中并没有明确规定，可以根据实际情况确定。

　　控制流采用的是HTTP文本协议，比较简单、方便调试，这个RTSP协议中也没有规定必须使用HTTP，不过一般都是采用HTTP来实现的。

 

　　大致步骤：

　　1. 客户端连接rtsp服务器，发送option方法；服务器返回可用的方法，通常有DESCRIBE,SETUP,PLAY,TEARDOWN等，由于板卡端的rtsp服务程序也是我们自己实现的，可以确保已经实现了这些方法，因此客户端就没有进行检查了；

　　2. 客户端发送DESCRIBE方法，服务器返回RTSP流的相关信息，包括video stream,audio stream的个数、码率、分辨率等参数信息；

　　3. 根据返回的参数信息，客户端决定要播放哪些video stream,audio stream，发送SETUP方法；

　　　我们的RTSP流为：一个alaw audio 和一个h264 video，需要指定音视频数据分别发送到哪个端口上，通过下面的代码来构造发送消息：　

 1 int RTSP::Set_Setup()
 2 {
 3     int nRet = -1;
 4     int m_nIndex = 0;
 5 
 6     if (m_pBuf != NULL)
 7     {
 8 //        if (m_pContentBase == NULL)
 9 //        {
10 //            sprintf(m_pBuf, "SETUP %s/%s %s\r\n", m_strUrl.c_str(), m_pMedia->p_control, RTSP_VERSSION);
11 //        }
12 //        else
13 //        {
14 //            sprintf(m_pBuf, "SETUP %s%s %s\r\n", m_pContentBase, m_pMedia->p_control, RTSP_VERSSION);
15 //        }
16 //        printf("m_pContentBase:%s\n", m_pContentBase);
17 //        printf("m_strUrl:%s\n", m_strUrl.c_str());
18 //        printf("m_pMedia->p_control:%s\n", m_pMedia->p_control);
19 //        printf("m_pBuf:%s\n", m_pBuf);
20         sprintf(m_pBuf, "SETUP %s %s\r\n", m_pMedia->p_control, RTSP_VERSSION);
21 
22         m_nIndex = strlen(m_pBuf);
23         sprintf(m_pBuf + m_nIndex, "CSeq: %d\r\n", m_nSeqNum);
24         m_nIndex = strlen(m_pBuf);
25 
26         if (m_pMedia->i_media_type == VIDEO)
27         {
28             GetVideoPort();
29             sprintf(m_pBuf + m_nIndex, "Transport: %s;%s;client_port=%d-%d\r\n", "RTP/AVP", "unicast", m_nVideoPort, m_nVideoPort + 1);
30             m_nIndex = strlen(m_pBuf);
31         }
32         else if (m_pMedia->i_media_type == AUDIO)
33         {
34             GetAudioPort();
35             sprintf(m_pBuf + m_nIndex, "Transport: %s;%s;client_port=%d-%d\r\n", "RTP/AVP", "unicast", m_nAudioPort, m_nAudioPort + 1);
36             m_nIndex = strlen(m_pBuf);
37         }
38 
39         if (m_pSession[0] != 0)
40         {
41             sprintf(m_pBuf + m_nIndex, "Session: %s\r\n", m_pSession);
42             m_nIndex = strlen(m_pBuf);
43         }
44 
45         sprintf(m_pBuf + m_nIndex, "User-Agent: %s\r\n", USER_AGENT_STR);
46         m_nIndex = strlen(m_pBuf);
47         sprintf(m_pBuf + m_nIndex, "\r\n");
48         m_nIndex = strlen(m_pBuf);
49         m_nBufSize = m_nIndex;
50 
51         nRet = 0;
52     }
53 
54     return nRet;
55 }

 

　　4. SETUP成功之后，通过PLAY命令就可以进行播放了：

 1 int RTSP::Set_Play()
 2 {
 3     int nRet = -1;
 4     int m_nIndex = 0;
 5 
 6     if (m_pBuf != NULL)
 7     {
 8         sprintf(m_pBuf, "PLAY %s %s\r\n", m_strUrl.c_str(), RTSP_VERSSION);
 9         m_nIndex = strlen(m_pBuf);
10         sprintf(m_pBuf + m_nIndex, "CSeq: %d\r\n", m_nSeqNum);
11         m_nIndex = strlen(m_pBuf);
12         sprintf(m_pBuf + m_nIndex, "Session: %s\r\n", m_pSession);
13         m_nIndex = strlen(m_pBuf);
14         sprintf(m_pBuf + m_nIndex, "Range: npt=0.000-\r\n");
15         m_nIndex = strlen(m_pBuf);
16         sprintf(m_pBuf + m_nIndex, "User-Agent: %s\r\n", USER_AGENT_STR);
17         m_nIndex = strlen(m_pBuf);
18         sprintf(m_pBuf + m_nIndex, "\r\n");
19         m_nIndex = strlen(m_pBuf);
20         m_nBufSize = m_nIndex;
21 
22         nRet = 0;
23     }
24 
25     return nRet;
26 }

　　这样我们就可以在刚才指定的端口上接收UDP的音视频数据了。

 更详细的可以参考rtsp协议的实现。

 

（2）合成MP4.

我们已经知道音视频格式分别为:alaw(pcma), h264;查看文档发现，mp4v2刚好支持这两种格式，剩下就很简单了：

 1 bool COutputATV::CreateMp4File(string filename)
 2 {
 3     m_Mp4File = MP4CreateEx(filename.c_str());
 4     if (m_Mp4File == MP4_INVALID_FILE_HANDLE)
 5     {
 6         return false;
 7     }
 8 
 9     MP4SetTimeScale(m_Mp4File, 90000);
10     m_nVideoTrack = MP4AddH264VideoTrack(m_Mp4File,
11                                          90000,                                     //timescale
12                                          3214,                                       //sample duration:/*(90000 / 25)*/
13                                                                                           /*  NOTICE:
14                                                                                            *  why 3214? read the commets below.
15                                                                                            */
16                                          320,                                         //width:
17                                          240,                                         //height:
18                                          0x64, //sps[1] AVCProfileIndication
19                                          0x00, //sps[2] profile_compat
20                                          0x1f, //sps[3] AVCLevelIndication
21                                          3); // 4 bytes length before each NAL unit
22     if (m_nVideoTrack == MP4_INVALID_TRACK_ID)
23     {
24         LOG(LOG_TYPE_ERROR, "CreateMp4File():MP4AddH264VideoTrack() failed.");
25         return false;
26     }
27     MP4SetVideoProfileLevel(m_Mp4File, 0x7F);
28 
29     m_nAudioTrack = MP4AddALawAudioTrack(m_Mp4File,
30                                          8000,  //timescale
31                                          500);  //sampleDuration.
32     /* NOTICE:
33      * in standard release of mp4v2 library(v1.9.1, and trunk-r479),the function MP4AddALawAudioTrack() does not specify the 3rd param:
34      * 'sampleDuration', it calculate a fixed duration value with the following formula:
35      *                      uint32_t fixedSampleDuration = (timeScale * 20)/1000; // 20mSec/Sample
36      * please read the source code of MP4AddALawAudioTrack().
37      * they can do it in this way because RFC3551 defines PCMA(a-law) as 20msec per sample, so the duration is a fixed value, please read RFC
38      * 3551:http://www.ietf.org/rfc/rfc3551.txt
39      * but, the souce boards' we used does not follow the RFC specifition, we found the sample duration value is 500.
40      * (why the param is 500? every rtp packet contains  a timestamp, the duration is the difference of two samples(not rtp packets), the same as
41      * h264 tracks in rtp). SO:
42      * I modified the declarion of MP4AddALawAudioTrack(), add the 3rd param:'sampleDuration', to pass the actual duration value,I also modified
43      * the implmention of MP4AddALawAudioTrack().
44      *
45      * as a result:
46      * ***************************               IMPORTANT                ***************************
47      * when distribute the Record software, you MUST use the mp4v2 library distribute with it,
48      * please DO NOT use the standard release download from network!
49      * ***********************************************************************************
50      *
51      * we use the default value of duration when creating mp4 file, we will modify it later when begin to write the first two samples with its
52      * actual value.
53      *
54      * Added by:Zhengfeng Rao.
55      * 2012-05-08
56      */
57 
58     MP4SetTrackIntegerProperty(m_Mp4File,
59                                m_nAudioTrack,
60                                "mdia.minf.stbl.stsd.alaw.channels",
61                                 1);
62 
63     if (m_nAudioTrack == MP4_INVALID_TRACK_ID)
64     {
65         LOG(LOG_TYPE_ERROR, "CreateMp4File():MP4AddAudioTrack() failed.");
66         return false;
67     }
68     MP4SetAudioProfileLevel(m_Mp4File, 0x02);
69 
70     return true;
71 }

 

写音视频数据：

  1 void COutputATV::DecodeRtp(unsigned char *pbuf, int datalength)
  2 {
  3     if((pbuf == NULL) || (datalength <= 0))
  4     {
  5         return;
  6     }
  7 
  8     rtp_header_t rtp_header;
  9     char cType = pbuf[0];
 10 
 11     //the 1st byte indicate the node is audio/video, it's added by the input thread, so we need to remove it.
 12     pbuf += 1;
 13     datalength -= 1;
 14     int i_header_size = GetRtpHeader(&rtp_header, pbuf, datalength);
 15 
 16     if(i_header_size <=0 )
 17     {
 18         LOG(LOG_TYPE_ERROR, "COutputATV::DecodeRtp() Invalid header size:%d", i_header_size);
 19         return;
 20     }
 21 
 22     if(cType == 'A')
 23     {
 24         if (rtp_header.i_pt == 0x8)//AUDIO
 25         {
 26             int i_size = datalength - i_header_size;
 27             if (m_nAudioTimeStamp == 0)
 28             {
 29                 m_nAudioTimeStamp = rtp_header.i_timestamp;
 30             }
 31 
 32             if (m_nAudioTimeStamp != rtp_header.i_timestamp)//got a frame
 33             {
 34                 MP4WriteSample(m_Mp4File, m_nAudioTrack, m_pAudioFrame, m_nAudioFrameIndex);
 35                 m_nAudioFrameIndex = 0;
 36 
 37                 m_nAudioTimeStamp = rtp_header.i_timestamp;
 38                 memcpy(m_pAudioFrame + m_nAudioFrameIndex, pbuf + i_header_size, i_size);
 39                 m_nAudioFrameIndex += i_size;
 40             }
 41             else
 42             {
 43                 memcpy(m_pAudioFrame + m_nAudioFrameIndex, pbuf + i_header_size, i_size);
 44                 m_nAudioFrameIndex += i_size;
 45             }
 46         }
 47         else
 48         {
 49             //INVALID packet.
 50         }
 51     }
 52     else if(cType == 'V')
 53     {
 54         if (rtp_header.i_pt == 0x60)// VIDEO
 55         {
 56             char p_save_buf[4096] = {0};
 57             int i_size = RtpToH264(pbuf, datalength, p_save_buf, &m_nNaluOkFlag, &m_nLastPktNum);
 58             if(i_size <= 0)
 59             {
 60                 DumpFrame(pbuf, datalength);
 61                 LOG_PERIOD(LOG_TYPE_WARN, "RtpToH264() Illegal packet, igonred. datalength = %d, i_size = %d", datalength-1, i_size);
 62                 return;
 63             }
 64 
 65             if (m_nVideoTimeStamp == 0)
 66             {
 67                 m_nVideoTimeStamp = rtp_header.i_timestamp;
 68 
 69                 m_nVideoFrameIndex = 0;
 70                 memcpy(m_pVideoFrame + m_nVideoFrameIndex, p_save_buf, i_size);
 71                 m_nVideoFrameIndex += i_size;
 72             }
 73 
 74             if (m_nVideoTimeStamp != rtp_header.i_timestamp || p_save_buf[12] == 0x78)
 75             {
 76                 if (m_nVideoFrameIndex >= 4)
 77                 {
 78                     unsigned int* p = (unsigned int*) (&m_pVideoFrame[0]);
 79                     *p = htonl(m_nVideoFrameIndex - 4);
 80 
 81                    MP4WriteSample(m_Mp4File, m_nVideoTrack, m_pVideoFrame, m_nVideoFrameIndex, MP4_INVALID_DURATION, 0, 1);
 82                     //DumpFrame(m_pVideoFrame, m_nVideoFrameIndex);
 83                 }
 84 
 85                 m_nVideoFrameIndex = 0;
 86                 m_nVideoTimeStamp = rtp_header.i_timestamp;
 87                 memcpy(m_pVideoFrame + m_nVideoFrameIndex, p_save_buf, i_size);
 88                 m_nVideoFrameIndex += i_size;
 89             }
 90             else
 91             {
 92                 //printf("2.3.3*************i_size:%d, m_nVideoFrameIndex:%d\n", i_size, m_nVideoFrameIndex);
 93                 memcpy(m_pVideoFrame + m_nVideoFrameIndex, p_save_buf, i_size);
 94                 m_nVideoFrameIndex += i_size;
 95             }
 96         }
 97         else
 98         {
 99             //INVALID packet.
100         }
101     }
102     else
103     {
104         //INVALID packet.
105     }
106 }

 

需要说明的是：

libmp4v2通过MP4AddALawAudioTrack(mp4file, timescale,sampleDuration)添加alaw音频时,第三个参数sampleDuration是我自己修改libmp4v2库添加的。

因为libmp4v2中 MP4AddALawAudioTrack接口为：MP4AddALawAudioTrack(mp4file, timescale)，sampleDuration是通过如下公式计算得到的：

uint32_t fixedSampleDuration = (timeScale * 20)/1000; // 20mSec/Sample

而这计算出来的值，并不符合我们的实际情况，所以我添加了这第三个参数，可以自己指定sample duration。