Read dumepd RTP stream in libav

Question

Hi I am in a need of a bit of a help/guidance because I got stuck in my research.

The problem:

How to convert RTP data using either gstreamer or avlib (ffmpeg) in either API (by programming) or console versions.

Data

I have RTP dump that comes from RTP/RTCP over TCP so I can get the precise start and stop for each RTP packet in file. It's a H264 video stream dump. The data is in this fashion because I need to acquire the RTCP/RTP interleaved stream via libcurl (which I'm currently doing)

Status

I've tried to use ffmpeg to consume pure RTP packets but is seems that using rtp either by console or by programming involves "starting" the whole rtsp/rtp session business in ffmpeg. I've stopped there and for the time being I didn't pursue this avenue deeper. I guess this is possible with lover level RTP API like ff_rtp_parse_packet() I'm too new with this lib to do it straight out.

Then there is the gstreamer It has somewhat more capabilities to do it without programming, but for the time being I'm not able to figure out how to pass it the RTP dump I have.

I have also tried to do a little bit of a trickery and stream the dump via socat/nc to the udp port and listen on it via ffplay with sdp file as an input, there seems to be some progress the rtp at least gets recognized, but for socat there are loads of packet missing (data sent too fast perhaps?) and in the end the data is not visualized. When I used nc the video was badly misshapen but at least there were not that much receive errors.

One way or another the data is not properly visualized.

I know I can depacketize the data "by hand" but the idea is to do it via some kind of library because in the end there would also be second stream with audio that would have to be muxed together with the video.

I would appreciate any help on how to tackle this problem. Thanks.

Answer 1

Finally after some period of time I had time to sit down at this problem again, and finally I've got the solution that satisfies me. I went on with RTP interleaved stream (RTP is interleaved with RTCP over single TCP connection).
So I had a interleaved RTCP/RTP stream that needed to be disassembled to Audio (PCM A-Law) and Video (h.264 Constrained baseline) RTP packets.
The decomposition of the RTSP stream containing RTP data is described here rfc2326.
Depacketization of the H264 is described here rfc6184, for the PCM A-Law the frames came out to be raw audio in RTP so no depacketization was necessary.

Next step was to calculate proper PTS (or presentation time stamp) for each stream, that was a bit of a hassle but finally the Live555 code came to help (see RTP lipsync synchronization).
The last task was to mux it into a container that would support PCM alaw, I've used ffmpeg's avlibraries.
There are many examples over the Internet but many of them are outdated (ffmpeg is very 'dynamic' in API changes region) so I'm posting (most important parts of) what actually worked for me in the end:

The setup part:

#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"

AVFormatContext   *formatContext;
AVOutputFormat    *outputFormat;
AVStream          *video_st;
AVStream          *audio_st;
AVCodec           *av_encode_codec = NULL;
AVCodec           *av_audio_encode_codec = NULL;
AVCodecContext    *av_video_encode_codec_ctx = NULL;
AVCodecContext    *av_audio_encode_codec_ctx = NULL;


av_register_all();
av_log_set_level(AV_LOG_TRACE);
outputFormat = av_guess_format(NULL, pu8outFileName, NULL);
outputFormat->video_codec = AV_CODEC_ID_H264;

av_encode_codec = avcodec_find_encoder(AV_CODEC_ID_H264);
av_audio_encode_codec = avcodec_find_encoder(AV_CODEC_ID_PCM_ALAW);
avformat_alloc_output_context2(&formatContext, NULL, NULL, pu8outFileName);
formatContext->oformat = outputFormat;
strcpy(formatContext->filename, pu8outFileName);
outputFormat->audio_codec  = AV_CODEC_ID_PCM_ALAW;

av_video_encode_codec_ctx = avcodec_alloc_context3(av_encode_codec);
av_audio_encode_codec_ctx = avcodec_alloc_context3(av_audio_encode_codec);

av_video_encode_codec_ctx->codec_id = outputFormat->video_codec;
av_video_encode_codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;
av_video_encode_codec_ctx->bit_rate = 4000;
av_video_encode_codec_ctx->width  = u32width;
av_video_encode_codec_ctx->height = u32height;
av_video_encode_codec_ctx->time_base = (AVRational){ 1, u8fps };
av_video_encode_codec_ctx->max_b_frames = 0;
av_video_encode_codec_ctx->pix_fmt = AV_PIX_FMT_YUV420P;

av_audio_encode_codec_ctx->sample_fmt = AV_SAMPLE_FMT_S16;
av_audio_encode_codec_ctx->codec_id = AV_CODEC_ID_PCM_ALAW; 
av_audio_encode_codec_ctx->codec_type = AVMEDIA_TYPE_AUDIO;
av_audio_encode_codec_ctx->sample_rate = 8000;
av_audio_encode_codec_ctx->channels = 1;
av_audio_encode_codec_ctx->time_base = (AVRational){ 1, u8fps };
av_audio_encode_codec_ctx->channel_layout = AV_CH_LAYOUT_MONO;

video_st = avformat_new_stream(formatContext, av_encode_codec);
audio_st = avformat_new_stream(formatContext, av_audio_encode_codec);
audio_st->index = 1;
video_st->avg_frame_rate = (AVRational){ 90000, 90000 / u8fps };
av_stream_set_r_frame_rate(video_st, (AVRational){ 90000, 90000 / u8fps });

The packets for video are written like this:

uint8_t  *pu8framePtr = video_frame;
AVPacket pkt = { 0 };
av_init_packet(&pkt);
if (0x65 == pu8framePtr[4] || 0x67 == pu8framePtr[4] || 0x68 == pu8framePtr[4]) 
{
    pkt.flags = AV_PKT_FLAG_KEY;
}

pkt.data = (uint8_t *)pu8framePtr;
pkt.size = u32LastFrameSize;

pkt.pts  = av_rescale_q(s_video_sync.fSyncTime.tv_sec * 1000000 + s_video_sync.fSyncTime.tv_usec, (AVRational){ 1, 1000000 }, video_st->time_base);
pkt.dts  = pkt.pts;
pkt.stream_index = video_st->index;
av_interleaved_write_frame(formatContext, &pkt);
av_packet_unref(&pkt);

and for the audio like this:

AVPacket pkt = { 0 };
av_init_packet(&pkt);
pkt.flags = AV_PKT_FLAG_KEY;
pkt.data = (uint8_t *)pu8framePtr;
pkt.size = u32AudioDataLen;

pkt.pts  = av_rescale_q(s_audio_sync.fSyncTime.tv_sec * 1000000 + s_audio_sync.fSyncTime.tv_usec, (AVRational){ 1, 1000000 }, audio_st->time_base);
pkt.dts  = pkt.pts;
pkt.stream_index = audio_st->index;
if (u8FirstIFrameFound) {av_interleaved_write_frame(formatContext, &pkt);}
av_packet_unref(&pkt)

and at the end some deinits:

av_write_trailer(formatContext);
av_dump_format(formatContext, 0, pu8outFileName, 1);
avcodec_free_context(&av_video_encode_codec_ctx);
avcodec_free_context(&av_audio_encode_codec_ctx);
avio_closep(&formatContext->pb);
avformat_free_context(formatContext);