[Libav-user] FFmpeg RTP payload 96 instead of 97
NITIN GOYAL
nitinkumgoyal at gmail.com
Sun Oct 30 09:33:23 EET 2016
RTP payload 96,97 are dynamic payloads and can be assigned as needed. RFC
2833
in general when we have both video and audio, Video goes for 96 while audio
goes for 97.
On Wed, Oct 26, 2016 at 7:08 AM, Yu Ang Tan <yuang86 at gmail.com> wrote:
> On Thu, Sep 22, 2016 at 11:46 AM Yu Ang Tan <yuang86 at gmail.com> wrote:
>
>> I want to create an rtp audio stream with ffmpeg. When I run my
>> application I get the following SDP file configuration:
>>
>> Output #0, rtp, to 'rtp://127.0.0.1:8554':
>> Stream #0:0: Audio: pcm_s16be, 8000 Hz, stereo, s16, 256 kb/s
>>
>> SDP:
>> v=0
>> o=- 0 0 IN IP4 127.0.0.1
>> s=No Name
>> c=IN IP4 127.0.0.1
>> t=0 0
>> a=tool:libavformat 57.25.101
>> m=audio 8554 RTP/AVP 96
>> b=AS:256
>> a=rtpmap:96 L16/8000/2
>>
>> However, when I try to read it with `ffplay -i test.sdp
>> -protocol_whitelist file,udp,rtp`, it fails,shows the following:
>>
>> ffplay version N-78598-g98a0053 Copyright (c) 2003-2016 the FFmpeg
>> developers
>> built with gcc 5.3.0 (GCC)
>> configuration: --disable-static --enable-shared --enable-gpl
>> --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib
>> --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv
>> --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
>> --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm
>> --enable-libilbc --enable-libmodplug --enable-libmp3lame
>> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
>> --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
>> --enable-libspeex --enable-libtheora --enable-libtwolame
>> --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis
>> --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
>> --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg
>> --enable-lzma --enable-decklink --enable-zlib
>> libavutil 55. 18.100 / 55. 18.100
>> libavcodec 57. 24.103 / 57. 24.103
>> libavformat 57. 25.101 / 57. 25.101
>> libavdevice 57. 0.101 / 57. 0.101
>> libavfilter 6. 34.100 / 6. 34.100
>> libswscale 4. 0.100 / 4. 0.100
>> libswresample 2. 0.101 / 2. 0.101
>> libpostproc 54. 0.100 / 54. 0.100
>> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0
>> (...waits indefinitely.)
>>
>> The only way to make it work again is to modify the payload type in the
>> SDP file from 96 to 97. Can someone tell me why? Where and how is this
>> number defined?
>>
>>
>> Here is my source.
>>
>> #include <math.h>
>> extern "C"
>> {
>> #include <libavutil/opt.h>
>> #include <libavcodec/avcodec.h>
>> #include <libavutil/channel_layout.h>
>> #include <libavutil/common.h>
>> #include <libavutil/imgutils.h>
>> #include <libavutil/mathematics.h>
>> #include <libavutil/samplefmt.h>
>> #include <libavformat/avformat.h>
>> }
>>
>>
>> static int write_frame(AVFormatContext *fmt_ctx, const AVRational
>> *time_base, AVStream *st, AVPacket *pkt)
>> {
>> /* rescale output packet timestamp values from codec to stream
>> timebase */
>> av_packet_rescale_ts(pkt, *time_base, st->time_base);
>>
>> /* Write the compressed frame to the media file. */
>> return av_interleaved_write_frame(fmt_ctx, pkt);
>> }
>>
>> /*
>> * Audio encoding example
>> */
>> static void audio_encode_example(const char *filename)
>> {
>> AVPacket pkt;
>> int i, j, k, ret, got_output;
>> int buffer_size;
>>
>> uint16_t *samples;
>> float t, tincr;
>>
>> AVCodec *outCodec = NULL;
>> AVCodecContext *outCodecCtx = NULL;
>> AVFormatContext *outFormatCtx = NULL;
>> AVStream * outAudioStream = NULL;
>> AVFrame *outFrame = NULL;
>>
>> ret = avformat_alloc_output_context2(&outFormatCtx, NULL, "rtp",
>> filename);
>> if (!outFormatCtx || ret < 0)
>> {
>> fprintf(stderr, "Could not allocate output context");
>> }
>>
>> outFormatCtx->flags |= AVFMT_FLAG_NOBUFFER |
>> AVFMT_FLAG_FLUSH_PACKETS;
>> outFormatCtx->oformat->audio_codec = AV_CODEC_ID_PCM_S16BE;
>>
>> /* find the encoder */
>> outCodec = avcodec_find_encoder(outFormatCtx->oformat->audio_
>> codec);
>> if (!outCodec) {
>> fprintf(stderr, "Codec not found\n");
>> exit(1);
>> }
>>
>> outAudioStream = avformat_new_stream(outFormatCtx, outCodec);
>> if (!outAudioStream)
>> {
>> fprintf(stderr, "Cannot add new audio stream\n");
>> exit(1);
>> }
>>
>> outAudioStream->id = outFormatCtx->nb_streams - 1;
>> outCodecCtx = outAudioStream->codec;
>> outCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
>>
>> /* select other audio parameters supported by the encoder */
>> outCodecCtx->sample_rate = 8000;
>> outCodecCtx->channel_layout = AV_CH_LAYOUT_STEREO;
>> outCodecCtx->channels = 2;
>>
>> /* open it */
>> if (avcodec_open2(outCodecCtx, outCodec, NULL) < 0) {
>> fprintf(stderr, "Could not open codec\n");
>> exit(1);
>> }
>>
>> // PCM has no frame, so we have to explicitly specify
>> outCodecCtx->frame_size = 1152;
>>
>> av_dump_format(outFormatCtx, 0, filename, 1);
>>
>> char buff[10000] = { 0 };
>> ret = av_sdp_create(&outFormatCtx, 1, buff, sizeof(buff));
>> printf("%s", buff);
>>
>> ret = avio_open2(&outFormatCtx->pb, filename, AVIO_FLAG_WRITE,
>> NULL, NULL);
>> ret = avformat_write_header(outFormatCtx, NULL);
>> printf("ret = %d\n", ret);
>> if (ret <0) {
>> exit(1);
>> }
>>
>> /* frame containing input audio */
>> outFrame = av_frame_alloc();
>> if (!outFrame) {
>> fprintf(stderr, "Could not allocate audio frame\n");
>> exit(1);
>> }
>>
>> outFrame->nb_samples = outCodecCtx->frame_size;
>> outFrame->format = outCodecCtx->sample_fmt;
>> outFrame->channel_layout = outCodecCtx->channel_layout;
>>
>> /* we calculate the size of the samples buffer in bytes */
>> buffer_size = av_samples_get_buffer_size(NULL,
>> outCodecCtx->channels, outCodecCtx->frame_size,
>> outCodecCtx->sample_fmt, 0);
>> if (buffer_size < 0) {
>> fprintf(stderr, "Could not get sample buffer size\n");
>> exit(1);
>> }
>> samples = (uint16_t*)av_malloc(buffer_size);
>> if (!samples) {
>> fprintf(stderr, "Could not allocate %d bytes for samples
>> buffer\n",
>> buffer_size);
>> exit(1);
>> }
>> /* setup the data pointers in the AVFrame */
>> ret = avcodec_fill_audio_frame(outFrame, outCodecCtx->channels,
>> outCodecCtx->sample_fmt,
>> (const uint8_t*)samples, buffer_size, 0);
>> if (ret < 0) {
>> fprintf(stderr, "Could not setup audio frame\n");
>> exit(1);
>> }
>>
>> /* encode a single tone sound */
>> t = 0;
>> int next_pts = 0;
>> tincr = 2 * M_PI * 440.0 / outCodecCtx->sample_rate;
>> for (i = 0; i < 400000; i++) {
>> av_init_packet(&pkt);
>> pkt.data = NULL; // packet data will be allocated by the
>> encoder
>> pkt.size = 0;
>>
>> for (j = 0; j < outCodecCtx->frame_size; j++) {
>> samples[2 * j] = (uint16_t)(sin(t) * 10000);
>>
>> for (k = 1; k < outCodecCtx->channels; k++)
>> samples[2 * j + k] = samples[2 * j];
>> t += tincr;
>> }
>> t = (t > 50000) ? 0 : t;
>>
>> // Sets time stamp
>> next_pts += outFrame->nb_samples;
>> outFrame->pts = next_pts;
>>
>> /* encode the samples */
>> ret = avcodec_encode_audio2(outCodecCtx, &pkt, outFrame,
>> &got_output);
>> if (ret < 0) {
>> fprintf(stderr, "Error encoding audio frame\n");
>> exit(1);
>> }
>> if (got_output) {
>> write_frame(outFormatCtx, &outCodecCtx->time_base,
>> outAudioStream, &pkt);
>> av_packet_unref(&pkt);
>> }
>>
>> printf("i:%d\n", i); // waste some time to avoid over-filling
>> jitter buffer
>> printf("Audio: %d\t%d\n", samples[0], samples[1]); // waste
>> some time to avoid over-filling jitter buffer
>> printf("t: %f\n", t); // waste some time to avoid
>> over-filling jitter buffer
>> }
>>
>> /* get the delayed frames */
>> for (got_output = 1; got_output; i++) {
>> ret = avcodec_encode_audio2(outCodecCtx, &pkt, NULL,
>> &got_output);
>> if (ret < 0) {
>> fprintf(stderr, "Error encoding frame\n");
>> exit(1);
>> }
>>
>> if (got_output) {
>> pkt.pts = AV_NOPTS_VALUE;
>> write_frame(outFormatCtx, &outCodecCtx->time_base,
>> outAudioStream, &pkt);
>> av_packet_unref(&pkt);
>> }
>> }
>>
>> av_freep(&samples);
>> av_frame_free(&outFrame);
>> avcodec_close(outCodecCtx);
>> av_free(outCodecCtx);
>> }
>>
>>
>> int main(int argc, char **argv)
>> {
>> const char *output;
>>
>> av_register_all();
>> avformat_network_init(); // for network streaming
>>
>> audio_encode_example("rtp://127.0.0.1:8554");
>>
>> return 0;
>> }
>>
>>
>>
> Interestingly, the same code when compiled in Linux Ubuntu gives the
> following:
>
> Output #0, rtp, to 'rtp://127.0.0.1:8554':
> Stream #0:0: Unknown: none (pcm_s16be)
> v=0
> o=- 0 0 IN IP4 127.0.0.1
> s=No Name
> c=IN IP4 127.0.0.1
> t=0 0
> a=tool:libavformat 57.48.100
> m=application 8554 RTP/AVP 3
>
> I noticed that the Output stream is now labeled as "Unknown". And in the
> SDP dump, the media name is switched from audio to application. RTP/AVP is
> now 3 instead of 96/97. What's going on?
>
> Would really appreciate it if any of you could shine some light on this.
>
>
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> Libav-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/libav-user
>
>
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