H.264 muxed to MP4 using libavformat not playing back
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.
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.
I suspect it has to be something with the way the stream is being created or something.
Init:
AVOutputFormat* fmt = av_guess_format( 0, "out.mp4", 0 );
oc = avformat_alloc_context();
oc->oformat = fmt;
strcpy(oc->filename, filename);
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);
}
I write packets out:
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);
}
Finally to close the file off:
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);
If I take the H.264 data alone and convert it:
ffmpeg -i recording.h264 -vcodec copy recording.mp4
All but the "footer" of the files are the same.
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.
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
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!
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.
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.
Globals:
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;
Main:
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;
}
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
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
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);
}
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
You would still need to know what the frame rate (time base) is for the video & audio.
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 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.
--
I have found additionally today that Annex B isn't really compatible with any other player so AVCC format should really be used.
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
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];
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.
Answer
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.