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Stereo master soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Audio
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Langue : English
Type : Audio
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Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Langue : English
Type : Audio
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The Four of Us are Dying
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Langue : English
Type : Audio
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Corona Radiata
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Type : Audio
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Revision 69399 : On refait les critères dans l’inclusion des graphiques, mais on n’affiche ...
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GStreamer with MPEG-TS Video4Linux ATSC/DVB Recording
21 juin 2013, par Dustin OpreaI'm having an impossible time setting up a filtergraph to read from a recording that I made from my DVB video4linux device. Any help would be vastly appreciated.
How I made the recording :
To tune the channel :
azap -c ~/channels.conf "Florida"To record the channel :
cat /dev/dvb/adapter0/dvr0 > /tmp/testThis is the way that recordings must be made (I can not use any GST DVB plugins to do this for me).
I used tstools to identify that the recording is a TS stream :
tstools/bin$ ./stream_type ~/recordings/20130129-202049
Reading from /home/dustin/recordings/20130129-202049
It appears to be Transport Stream...but that there are no PAT/PMT frames :
tstools/bin$ ./tsinfo ~/recordings/20130129-202049
Reading from /home/dustin/recordings/20130129-202049
Scanning 10000 TS packets
Found 0 PAT packets and 0 PMT packets in 10000 TS packetsI was able to produce a single ES (elementary stream) stream, by running ts2es :
tstools/bin$ ./ts2es -pid 97 ~/recordings/20130129-202049 ~/recordings/20130129-202049.es
Reading from /home/dustin/recordings/20130129-202049
Writing to /home/dustin/recordings/20130129-202049.es
Extracting packets for PID 0061 (97)
!!! 4 bytes ignored at end of file - not enough to make a TS packet
Extracted 219258 of 248113 TS packetsI am able to play the ES stream (even though the video is frozen on the first frame) :
gst-launch-0.10 filesrc location=~/recordings/20130129-202049.es ! decodebin2 ! autovideosink
gst-launch-0.10 filesrc location=~/recordings/20130129-202049.es ! decodebin2 ! xvimagesink
gst-launch-0.10 playbin2 uri=file:///home/dustin/recordings/20130129-202049.esNo matter what I do, though, I can't get the original TS file to open. However, it opens in Mplayer/FFMPEG, perfectly (but not VLC). This is the output of FFMPEG :
ffmpeg -i 20130129-202049
ffmpeg version 0.8.5-4:0.8.5-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers
built on Jan 24 2013 18:03:14 with gcc 4.6.3
*** THIS PROGRAM IS DEPRECATED ***
This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
[mpeg2video @ 0x9be7be0] mpeg_decode_postinit() failure
Last message repeated 4 times
[mpegts @ 0x9be3aa0] max_analyze_duration reached
[mpegts @ 0x9be3aa0] PES packet size mismatch
Input #0, mpegts, from '20130129-202049':
Duration: 00:03:39.99, start: 9204.168844, bitrate: 1696 kb/s
Stream #0.0[0x61]: Video: mpeg2video (Main), yuv420p, 528x480 [PAR 40:33 DAR 4:3], 15000 kb/s, 30.57 fps, 29.97 tbr, 90k tbn, 59.94 tbc
Stream #0.1[0x64]: Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s
At least one output file must be specifiedThis tells us that the video stream has PID 0x61 (97).
I have been trying for a few days, so the following are only examples of a couple of attempts. I'll provide the playbin2 example first, since I know that thousands of people will respond, insisting that I just use that. It doesn't work.
gst-launch-0.10 playbin2 uri=file:///home/dustin/recordings/20130129-202049
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMpegTSDemux:mpegtsdemux0: Could not determine type of stream.
Additional debug info:
gstmpegtsdemux.c(2931): gst_mpegts_demux_sink_event (): /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMpegTSDemux:mpegtsdemux0:
No valid streams found at EOS
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...It fails [probably] because no PID has been specified with which to find the video (the "EOS" error, I think).
Naturally, I tried the following to start by demuxing the TS format. I believe it's the "es-pids" property that receives the PID in the absence of PMT information (which tstools said there weren't any, above), but I tried "program-number", too, just in case. gst-inspect indicates that one is hex and the other is decimal :
gst-launch-0.10 -v filesrc location=20130129-202049 ! mpegtsdemux es-pids=0x61 ! fakesink
gst-launch-0.10 -v filesrc location=20130129-202049 ! mpegtsdemux program-number=97 ! fakesinkOutput :
gst-launch-0.10 filesrc location=20130129-202049 ! mpegtsdemux es-pids=0x61 ! fakesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstMpegTSDemux:mpegtsdemux0: Could not determine type of stream.
Additional debug info:
gstmpegtsdemux.c(2931): gst_mpegts_demux_sink_event (): /GstPipeline:pipeline0/GstMpegTSDemux:mpegtsdemux0:
No valid streams found at EOS
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
dustin@dustinmicro:~/recordings$ gst-launch-0.10 -v filesrc location=20130129-202049 ! mpegtsdemux es-pids=0x61 ! fakesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstMpegTSDemux:mpegtsdemux0: Could not determine type of stream.
Additional debug info:
gstmpegtsdemux.c(2931): gst_mpegts_demux_sink_event (): /GstPipeline:pipeline0/GstMpegTSDemux:mpegtsdemux0:
No valid streams found at EOS
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
dustin@dustinmicro:~/recordings$ gst-launch-0.10 -v filesrc location=20130129-202049 ! mpegtsdemux es-pids=0x61 ! fakesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstMpegTSDemux:mpegtsdemux0: Could not determine type of stream.
Additional debug info:
gstmpegtsdemux.c(2931): gst_mpegts_demux_sink_event (): /GstPipeline:pipeline0/GstMpegTSDemux:mpegtsdemux0:
No valid streams found at EOS
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...However, when I try mpegpsdemux (for program streams (PS), as opposed to transport streams (TS)), I get further :
gst-launch-0.10 filesrc location=20130129-202049 ! mpegpsdemux ! fakesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_event_new_new_segment_full: assertion `position != -1' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_mini_object_ref: assertion `mini_object != NULL' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_pad_push_event: assertion `event != NULL' failed
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_event_new_new_segment_full: assertion `position != -1' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_mini_object_ref: assertion `mini_object != NULL' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_pad_push_event: assertion `event != NULL' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_event_new_new_segment_full: assertion `position != -1' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_mini_object_ref: assertion `mini_object != NULL' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_pad_push_event: assertion `event != NULL' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_event_new_new_segment_full: assertion `position != -1' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_mini_object_ref: assertion `mini_object != NULL' failed
(gst-launch-0.10:14805): GStreamer-CRITICAL **: gst_pad_push_event: assertion `event != NULL' failed
...
Got EOS from element "pipeline0".
Execution ended after 1654760008 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...I'll still get the same problem whenever I use the mpegtsdemux, even if it follows mpegpsdemux, above.
I don't understand this, since I haven't even picked a PID yet.
What am I doing wrong ?
Dustin
-
H.264 muxed to MP4 using libavformat not playing back
14 mai 2015, par Brad MitchellI 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.mp4All 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 parametersI 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="playback" file="file"> <output mp4="mp4" file="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;
}
</output></stream>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 presentThe 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 memoryYou 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.htmlWhen 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;iextradata[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;iextradata[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.
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