使用 libavformat 不播放的 H.264 混合到 MP4 [英] H.264 muxed to MP4 using libavformat not playing back
问题描述
我正在尝试将 H.264 数据复用到 MP4 文件中.将此 H.264 Annex B 数据保存为 MP4 文件似乎没有错误，但该文件无法播放.
I am trying to mux H.264 data into a MP4 file. There appear to be no errors in saving this H.264 Annex B data out to an MP4 file, but the file fails to playback.
我对文件进行了二进制比较，问题似乎出在写入 MP4 文件页脚(预告片)的内容中.
I've done a binary comparison on the files and the issue seems to be somewhere in what is being written to the footer (trailer) of the MP4 file.
我怀疑它必须与创建流的方式有关.
I suspect it has to be something with the way the stream is being created or something.
初始化:
AVOutputFormat* fmt = av_guess_format( 0, "out.mp4", 0 );
oc = avformat_alloc_context();
oc->oformat = fmt;
strcpy(oc->filename, filename);
我拥有的这个原型应用程序的一部分是为每个 IFrame 创建一个 png 文件.因此，当遇到第一个 IFrame 时，我创建视频流并写入 av 标头等:
Part of this prototype app I have is creating a png file for each IFrame. So when the first IFrame is encountered, I create the video stream and write the av header etc:
void addVideoStream(AVCodecContext* decoder)
{
videoStream = av_new_stream(oc, 0);
if (!videoStream)
{
cout << "ERROR creating video stream" << endl;
return;
}
vi = videoStream->index;
videoContext = videoStream->codec;
videoContext->codec_type = AVMEDIA_TYPE_VIDEO;
videoContext->codec_id = decoder->codec_id;
videoContext->bit_rate = 512000;
videoContext->width = decoder->width;
videoContext->height = decoder->height;
videoContext->time_base.den = 25;
videoContext->time_base.num = 1;
videoContext->gop_size = decoder->gop_size;
videoContext->pix_fmt = decoder->pix_fmt;
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
videoContext->flags |= CODEC_FLAG_GLOBAL_HEADER;
av_dump_format(oc, 0, filename, 1);
if (!(oc->oformat->flags & AVFMT_NOFILE))
{
if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
cout << "Error opening file" << endl;
}
avformat_write_header(oc, NULL);
}
我写出数据包:
unsigned char* data = block->getData();
unsigned char videoFrameType = data[4];
int dataLen = block->getDataLen();
// store pps
if (videoFrameType == 0x68)
{
if (ppsFrame != NULL)
{
delete ppsFrame; ppsFrameLength = 0; ppsFrame = NULL;
}
ppsFrameLength = block->getDataLen();
ppsFrame = new unsigned char[ppsFrameLength];
memcpy(ppsFrame, block->getData(), ppsFrameLength);
}
else if (videoFrameType == 0x67)
{
// sps
if (spsFrame != NULL)
{
delete spsFrame; spsFrameLength = 0; spsFrame = NULL;
}
spsFrameLength = block->getDataLen();
spsFrame = new unsigned char[spsFrameLength];
memcpy(spsFrame, block->getData(), spsFrameLength);
}
if (videoFrameType == 0x65 || videoFrameType == 0x41)
{
videoFrameNumber++;
}
if (videoFrameType == 0x65)
{
decodeIFrame(videoFrameNumber, spsFrame, spsFrameLength, ppsFrame, ppsFrameLength, data, dataLen);
}
if (videoStream != NULL)
{
AVPacket pkt = { 0 };
av_init_packet(&pkt);
pkt.stream_index = vi;
pkt.flags = 0;
pkt.pts = pkt.dts = 0;
if (videoFrameType == 0x65)
{
// combine the SPS PPS & I frames together
pkt.flags |= AV_PKT_FLAG_KEY;
unsigned char* videoFrame = new unsigned char[spsFrameLength+ppsFrameLength+dataLen];
memcpy(videoFrame, spsFrame, spsFrameLength);
memcpy(&videoFrame[spsFrameLength], ppsFrame, ppsFrameLength);
memcpy(&videoFrame[spsFrameLength+ppsFrameLength], data, dataLen);
// overwrite the start code (00 00 00 01 with a 32-bit length)
setLength(videoFrame, spsFrameLength-4);
setLength(&videoFrame[spsFrameLength], ppsFrameLength-4);
setLength(&videoFrame[spsFrameLength+ppsFrameLength], dataLen-4);
pkt.size = dataLen + spsFrameLength + ppsFrameLength;
pkt.data = videoFrame;
av_interleaved_write_frame(oc, &pkt);
delete videoFrame; videoFrame = NULL;
}
else if (videoFrameType != 0x67 && videoFrameType != 0x68)
{
// Send other frames except pps & sps which are caught and stored
pkt.size = dataLen;
pkt.data = data;
setLength(data, dataLen-4);
av_interleaved_write_frame(oc, &pkt);
}
最后关闭文件:
av_write_trailer(oc);
int i = 0;
for (i = 0; i < oc->nb_streams; i++)
{
av_freep(&oc->streams[i]->codec);
av_freep(&oc->streams[i]);
}
if (!(oc->oformat->flags & AVFMT_NOFILE))
{
avio_close(oc->pb);
}
av_free(oc);
如果我单独使用 H.264 数据并对其进行转换:
If I take the H.264 data alone and convert it:
ffmpeg -i recording.h264 -vcodec copy recording.mp4
除了文件的页脚"外，所有文件都相同.
All but the "footer" of the files are the same.
我的程序的输出:readrec 录音.tcp out.mp4**** 开始 **** 01-03-2013 14:26:01 180000输出 #0，mp4，到 'out.mp4':流 #0:0:视频:h264、yuv420p、352x288、q=2-31、512 kb/s、90k tbn、25 tbc**** 完 **** 01-03-2013 14:27:01 102000写了 1499 个视频帧.
Output from my program: readrec recording.tcp out.mp4 **** START **** 01-03-2013 14:26:01 180000 Output #0, mp4, to 'out.mp4': Stream #0:0: Video: h264, yuv420p, 352x288, q=2-31, 512 kb/s, 90k tbn, 25 tbc **** END **** 01-03-2013 14:27:01 102000 Wrote 1499 video frames.
如果我尝试使用 ffmpeg 转换使用 CODE 创建的 MP4 文件:
If I try to convert using ffmpeg the MP4 file created using CODE:
ffmpeg -i out.mp4 -vcodec copy out2.mp4
ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
built on Mar 7 2013 12:49:22 with suncc 0x5110
configuration: --extra-cflags=-KPIC -g --disable-mmx
--disable-protocol=udp --disable-encoder=nellymoser --cc=cc --cxx=CC
libavutil 51. 54.100 / 51. 54.100
libavcodec 54. 23.100 / 54. 23.100
libavformat 54. 6.100 / 54. 6.100
libavdevice 54. 0.100 / 54. 0.100
libavfilter 2. 77.100 / 2. 77.100
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 15.100 / 0. 15.100
h264 @ 12eaac0] no frame!
Last message repeated 1 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 23 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 74 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 64 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 34 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 49 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 24 times
[h264 @ 12eaac0] Partitioned H.264 support is incomplete
[h264 @ 12eaac0] no frame!
Last message repeated 23 times
[h264 @ 12eaac0] sps_id out of range
[h264 @ 12eaac0] no frame!
Last message repeated 148 times
[h264 @ 12eaac0] sps_id (32) out of range
Last message repeated 1 times
[h264 @ 12eaac0] no frame!
Last message repeated 33 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 128 times
[h264 @ 12eaac0] sps_id (32) out of range
Last message repeated 1 times
[h264 @ 12eaac0] no frame!
Last message repeated 3 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 3 times
[h264 @ 12eaac0] slice type too large (0) at 0 0
[h264 @ 12eaac0] decode_slice_header error
[h264 @ 12eaac0] no frame!
Last message repeated 309 times
[h264 @ 12eaac0] sps_id (32) out of range
Last message repeated 1 times
[h264 @ 12eaac0] no frame!
Last message repeated 192 times
[h264 @ 12eaac0] Partitioned H.264 support is incomplete
[h264 @ 12eaac0] no frame!
Last message repeated 73 times
[h264 @ 12eaac0] sps_id (32) out of range
Last message repeated 1 times
[h264 @ 12eaac0] no frame!
Last message repeated 99 times
[h264 @ 12eaac0] sps_id (32) out of range
Last message repeated 1 times
[h264 @ 12eaac0] no frame!
Last message repeated 197 times
[mov,mp4,m4a,3gp,3g2,mj2 @ 12e3100] decoding for stream 0 failed
[mov,mp4,m4a,3gp,3g2,mj2 @ 12e3100] Could not find codec parameters
(Video: h264 (avc1 / 0x31637661), 393539 kb/s)
out.mp4: could not find codec parameters
我真的不知道问题出在哪里，除非它与流的设置方式有关.我查看了一些其他人正在做类似事情的代码，并尝试使用此建议来设置流，但无济于事！
I really do not know where the issue is, except it has to be something to do with the way the streams are being set up. I've looked at bits of code from where other people are doing a similar thing, and tried to use this advice in setting up the streams, but to no avail!
给我一个 H.264/AAC 混合(同步)文件的最终代码如下.首先是一些背景信息.数据来自网络摄像机.数据通过第 3 方 API 以视频/音频数据包的形式呈现.视频数据包作为 RTP 有效载荷数据(无标头)呈现，并由重建并转换为附件 B 格式的 H.264 视频的 NALU 组成.AAC 音频显示为原始 AAC，并转换为 adts 格式以启用播放.这些数据包已被放入比特流格式，允许传输时间戳(自 1970 年 1 月 1 日以来的 64 位毫秒)以及其他一些内容.
The final code which gave me a H.264/AAC muxed (synced) file is as follows. First a bit of background information. The data is coming from an IP camera. The data is presented via a 3rd party API as video/audio packets. The video packets are presented as the RTP payload data (no header) and consist of NALU's that are reconstructed and converted to H.264 video in Annex B format. AAC audio is presented as raw AAC and is converted to adts format to enable playback. These packets have been put into a bitstream format that allows the transmission of the timestamp (64 bit milliseconds since Jan 1 1970) along with a few other things.
这或多或少是一个原型，在任何方面都不干净.可能漏水不好.但是，我愿意，希望这可以帮助其他人尝试实现与我相似的目标.
This is more or less a prototype and is not clean in any respects. It probably leaks bad. I do however, hope this helps anyone else out trying to achieve something similar to what I am.
全局变量:
AVFormatContext* oc = NULL;
AVCodecContext* videoContext = NULL;
AVStream* videoStream = NULL;
AVCodecContext* audioContext = NULL;
AVStream* audioStream = NULL;
AVCodec* videoCodec = NULL;
AVCodec* audioCodec = NULL;
int vi = 0; // Video stream
int ai = 1; // Audio stream
uint64_t firstVideoTimeStamp = 0;
uint64_t firstAudioTimeStamp = 0;
int audioStartOffset = 0;
char* filename = NULL;
Boolean first = TRUE;
int videoFrameNumber = 0;
int audioFrameNumber = 0;
主要内容:
int main(int argc, char* argv[])
{
if (argc != 3)
{
cout << argv[0] << " <stream playback file> <output mp4 file>" << endl;
return 0;
}
char* input_stream_file = argv[1];
filename = argv[2];
av_register_all();
fstream inFile;
inFile.open(input_stream_file, ios::in);
// Used to store the latest pps & sps frames
unsigned char* ppsFrame = NULL;
int ppsFrameLength = 0;
unsigned char* spsFrame = NULL;
int spsFrameLength = 0;
// Setup MP4 output file
AVOutputFormat* fmt = av_guess_format( 0, filename, 0 );
oc = avformat_alloc_context();
oc->oformat = fmt;
strcpy(oc->filename, filename);
// Setup the bitstream filter for AAC in adts format. Could probably also achieve
// this by stripping the first 7 bytes!
AVBitStreamFilterContext* bsfc = av_bitstream_filter_init("aac_adtstoasc");
if (!bsfc)
{
cout << "Error creating adtstoasc filter" << endl;
return -1;
}
while (inFile.good())
{
TcpAVDataBlock* block = new TcpAVDataBlock();
block->readStruct(inFile);
DateTime dt = block->getTimestampAsDateTime();
switch (block->getPacketType())
{
case TCP_PACKET_H264:
{
if (firstVideoTimeStamp == 0)
firstVideoTimeStamp = block->getTimeStamp();
unsigned char* data = block->getData();
unsigned char videoFrameType = data[4];
int dataLen = block->getDataLen();
// pps
if (videoFrameType == 0x68)
{
if (ppsFrame != NULL)
{
delete ppsFrame; ppsFrameLength = 0;
ppsFrame = NULL;
}
ppsFrameLength = block->getDataLen();
ppsFrame = new unsigned char[ppsFrameLength];
memcpy(ppsFrame, block->getData(), ppsFrameLength);
}
else if (videoFrameType == 0x67)
{
// sps
if (spsFrame != NULL)
{
delete spsFrame; spsFrameLength = 0;
spsFrame = NULL;
}
spsFrameLength = block->getDataLen();
spsFrame = new unsigned char[spsFrameLength];
memcpy(spsFrame, block->getData(), spsFrameLength);
}
if (videoFrameType == 0x65 || videoFrameType == 0x41)
{
videoFrameNumber++;
}
// Extract a thumbnail for each I-Frame
if (videoFrameType == 0x65)
{
decodeIFrame(h264, spsFrame, spsFrameLength, ppsFrame, ppsFrameLength, data, dataLen);
}
if (videoStream != NULL)
{
AVPacket pkt = { 0 };
av_init_packet(&pkt);
pkt.stream_index = vi;
pkt.flags = 0;
pkt.pts = videoFrameNumber;
pkt.dts = videoFrameNumber;
if (videoFrameType == 0x65)
{
pkt.flags = 1;
unsigned char* videoFrame = new unsigned char[spsFrameLength+ppsFrameLength+dataLen];
memcpy(videoFrame, spsFrame, spsFrameLength);
memcpy(&videoFrame[spsFrameLength], ppsFrame, ppsFrameLength);
memcpy(&videoFrame[spsFrameLength+ppsFrameLength], data, dataLen);
pkt.data = videoFrame;
av_interleaved_write_frame(oc, &pkt);
delete videoFrame; videoFrame = NULL;
}
else if (videoFrameType != 0x67 && videoFrameType != 0x68)
{
pkt.size = dataLen;
pkt.data = data;
av_interleaved_write_frame(oc, &pkt);
}
}
break;
}
case TCP_PACKET_AAC:
if (firstAudioTimeStamp == 0)
{
firstAudioTimeStamp = block->getTimeStamp();
uint64_t millseconds_difference = firstAudioTimeStamp - firstVideoTimeStamp;
audioStartOffset = millseconds_difference * 16000 / 1000;
cout << "audio offset: " << audioStartOffset << endl;
}
if (audioStream != NULL)
{
AVPacket pkt = { 0 };
av_init_packet(&pkt);
pkt.stream_index = ai;
pkt.flags = 1;
pkt.pts = audioFrameNumber*1024;
pkt.dts = audioFrameNumber*1024;
pkt.data = block->getData();
pkt.size = block->getDataLen();
pkt.duration = 1024;
AVPacket newpacket = pkt;
int rc = av_bitstream_filter_filter(bsfc, audioContext,
NULL,
&newpacket.data, &newpacket.size,
pkt.data, pkt.size,
pkt.flags & AV_PKT_FLAG_KEY);
if (rc >= 0)
{
//cout << "Write audio frame" << endl;
newpacket.pts = audioFrameNumber*1024;
newpacket.dts = audioFrameNumber*1024;
audioFrameNumber++;
newpacket.duration = 1024;
av_interleaved_write_frame(oc, &newpacket);
av_free_packet(&newpacket);
}
else
{
cout << "Error filtering aac packet" << endl;
}
}
break;
case TCP_PACKET_START:
break;
case TCP_PACKET_END:
break;
}
delete block;
}
inFile.close();
av_write_trailer(oc);
int i = 0;
for (i = 0; i < oc->nb_streams; i++)
{
av_freep(&oc->streams[i]->codec);
av_freep(&oc->streams[i]);
}
if (!(oc->oformat->flags & AVFMT_NOFILE))
{
avio_close(oc->pb);
}
av_free(oc);
delete spsFrame; spsFrame = NULL;
delete ppsFrame; ppsFrame = NULL;
cout << "Wrote " << videoFrameNumber << " video frames." << endl;
return 0;
}
在名为 addVideoAndAudioStream() 的函数中添加流流/编解码器并创建标头.这个函数是从 decodeIFrame() 调用的，所以有一些假设(不一定很好)1.先有视频包2. AAC 存在
The stream stream/codecs are added and the header is created in a function called addVideoAndAudioStream(). This function is called from decodeIFrame() so there are a few assumptions (which aren't necessarily good) 1. A video packet comes first 2. AAC is present
decodeIFrame 是一种独立的原型，我在其中为每个 I Frame 创建缩略图.生成缩略图的代码来自:https://gnunet.org/svn/提取器/src/plugins/thumbnailffmpeg_extractor.c
The decodeIFrame was kind of a separate prototype by where I was creating a thumbnail for each I Frame. The code to generate thumbnails was from: https://gnunet.org/svn/Extractor/src/plugins/thumbnailffmpeg_extractor.c
decodeIFrame 函数将 AVCodecContext 传递给 addVideoAudioStream:
The decodeIFrame function passes an AVCodecContext into addVideoAudioStream:
void addVideoAndAudioStream(AVCodecContext* decoder = NULL)
{
videoStream = av_new_stream(oc, 0);
if (!videoStream)
{
cout << "ERROR creating video stream" << endl;
return;
}
vi = videoStream->index;
videoContext = videoStream->codec;
videoContext->codec_type = AVMEDIA_TYPE_VIDEO;
videoContext->codec_id = decoder->codec_id;
videoContext->bit_rate = 512000;
videoContext->width = decoder->width;
videoContext->height = decoder->height;
videoContext->time_base.den = 25;
videoContext->time_base.num = 1;
videoContext->gop_size = decoder->gop_size;
videoContext->pix_fmt = decoder->pix_fmt;
audioStream = av_new_stream(oc, 1);
if (!audioStream)
{
cout << "ERROR creating audio stream" << endl;
return;
}
ai = audioStream->index;
audioContext = audioStream->codec;
audioContext->codec_type = AVMEDIA_TYPE_AUDIO;
audioContext->codec_id = CODEC_ID_AAC;
audioContext->bit_rate = 64000;
audioContext->sample_rate = 16000;
audioContext->channels = 1;
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
{
videoContext->flags |= CODEC_FLAG_GLOBAL_HEADER;
audioContext->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(oc, 0, filename, 1);
if (!(oc->oformat->flags & AVFMT_NOFILE))
{
if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
cout << "Error opening file" << endl;
}
}
avformat_write_header(oc, NULL);
}
据我所知，一些假设似乎并不重要，例如:1. 比特率.实际视频比特率约为 262k 而我指定为 512kbit2. AAC 频道.我指定了单声道，虽然实际输出是内存中的立体声
As far as I can tell, a number of assumptions didn't seem to matter, for example: 1. Bit Rate. The actual video bit rate was ~262k whereas I specified 512kbit 2. AAC channels. I specified mono, although the actual output was Stereo from memory
您仍然需要知道视频的帧速率(时基)是多少?音频.
You would still need to know what the frame rate (time base) is for the video & audio.
与许多其他示例相反，在设置 pts & 时dts 在视频数据包上，无法播放.我需要知道时基 (25fps)，然后设置 pts &dts 根据该时基，即第一帧 = 0(PPS、SPS、I)，第二帧 = 1(中间帧，不管它叫什么；)).
Contrary to a lot of other examples, when setting pts & dts on the video packets, it was not playable. I needed to know the time base (25fps) and then set the pts & dts according to that time base, i.e. first frame = 0 (PPS, SPS, I), second frame = 1 (intermediate frame, whatever its called ;)).
AAC 我还必须假设它是 16000 hz.每个 AAC 数据包 1024 个样本(我认为您也可以使用 AAC @ 960 个样本)来确定音频偏移".我将此添加到 pts &dts.因此 pts/dts 是要播放的样本编号.您还需要确保在写入之前在数据包中设置了 1024 的持续时间.
AAC I also had to make the assumption that it was 16000 hz. 1024 samples per AAC packet (You can also have AAC @ 960 samples I think) to determine the audio "offset". I added this to the pts & dts. So the pts/dts are the sample number that it is to played back at. You also need to make sure that the duration of 1024 is set in the packet before writing also.
--
我今天另外发现附件 B 与任何其他播放器并不真正兼容，因此应该真正使用 AVCC 格式.
I have found additionally today that Annex B isn't really compatible with any other player so AVCC format should really be used.
这些 URL 有助于:使用 ffmpeg (libavcodec) 通过 RTP 解码 H264 视频的问题http://aviadr1.blogspot.com.au/2010/05/h264-extradata-partially-explained-for.html
These URLS helped: Problem to Decode H264 video over RTP with ffmpeg (libavcodec) http://aviadr1.blogspot.com.au/2010/05/h264-extradata-partially-explained-for.html
在构建视频流时，我填写了 extradata &额外数据大小:
When constructing the video stream, I filled out the extradata & extradata_size:
// Extradata contains PPS & SPS for AVCC format
int extradata_len = 8 + spsFrameLen-4 + 1 + 2 + ppsFrameLen-4;
videoContext->extradata = (uint8_t*)av_mallocz(extradata_len);
videoContext->extradata_size = extradata_len;
videoContext->extradata[0] = 0x01;
videoContext->extradata[1] = spsFrame[4+1];
videoContext->extradata[2] = spsFrame[4+2];
videoContext->extradata[3] = spsFrame[4+3];
videoContext->extradata[4] = 0xFC | 3;
videoContext->extradata[5] = 0xE0 | 1;
int tmp = spsFrameLen - 4;
videoContext->extradata[6] = (tmp >> 8) & 0x00ff;
videoContext->extradata[7] = tmp & 0x00ff;
int i = 0;
for (i=0;i<tmp;i++)
videoContext->extradata[8+i] = spsFrame[4+i];
videoContext->extradata[8+tmp] = 0x01;
int tmp2 = ppsFrameLen-4;
videoContext->extradata[8+tmp+1] = (tmp2 >> 8) & 0x00ff;
videoContext->extradata[8+tmp+2] = tmp2 & 0x00ff;
for (i=0;i<tmp2;i++)
videoContext->extradata[8+tmp+3+i] = ppsFrame[4+i];
写出帧时，不要在前面加上 SPS &PPS 帧，只需写出 I 帧 &P帧.另外，将前4个字节(0x00 0x00 0x00 0x01)中包含的Annex B起始码替换为I/P帧的大小.
When writing out the frames, don't prepend the SPS & PPS frames, just write out the I Frame & P frames. In addition, replace the Annex B start code contained in the first 4 bytes (0x00 0x00 0x00 0x01) with the size of the I/P frame.
推荐答案
请让我总结一下:您(原始)代码的问题是 av_interleaved_write_frame() 的输入不应该开始与数据包长度.如果您不删除 00 00 00 01 起始代码，该文件可能仍可播放，但恕我直言，这是播放器的弹性行为，我不会指望这一点.
Please let me sum it up: the problem with your (original) code was that the input to av_interleaved_write_frame() should not start with the packet length. The file may still be playable if you don't strip the 00 00 00 01 start codes, but that IMHO is a resilience behavior of the player, and I would not count on this.
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