[Libav-user] Bad packed header lengths: libvorbis
Yu Ang Tan
yuang86 at gmail.com
Tue Sep 13 04:20:04 EEST 2016
I use the `av_sdp_create()` to generate my SDP file but I think somethings
is wrong. Here is the output:
Output #0, rtp, to 'rtp://127.0.0.1:8554':
Stream #0:0: Audio: vorbis (libvorbis), 44100 Hz, stereo, fltp, 64 kb/s
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:64
a=rtpmap:96 vorbis/44100/2
a=fmtp:96
configuration=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
If I try to use the SDP file anyway, I get the following message:
>ffplay -i test2.sdp -protocol_whitelist file,udp,rtp
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
[NULL @ 05aea880] Bad packed header lengths (30,0,616,3793) f=0/0
[vorbis @ 05aea880] Extradata missing.
[sdp @ 05ad7ec0] Failed to open codec in av_find_stream_info
nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0
Am I overlooking something important? Am I going about this the wrong way?
Any help would be much appreciated.
Here is my source:
#include "stdafx.h"
#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>
}
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
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->max_interleave_delta = 1;
outFormatCtx->oformat->audio_codec = AV_CODEC_ID_VORBIS;
/* 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;
outAudioStream->time_base.den = 44100; // 44.100 kHz
outAudioStream->time_base.num = 1;
outCodecCtx = outAudioStream->codec;
outCodecCtx->bit_rate = 64000;
outCodecCtx->time_base.den = outAudioStream->time_base.den;
outCodecCtx->time_base.num = outAudioStream->time_base.num;
/* check that the encoder supports input */
outCodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (!check_sample_fmt(outCodec, outCodecCtx->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(outCodecCtx->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
outCodecCtx->sample_rate = select_sample_rate(outCodec);
outCodecCtx->channel_layout = select_channel_layout(outCodec);
outCodecCtx->channels =
av_get_channel_layout_nb_channels(outCodecCtx->channel_layout);
/* open it */
if (avcodec_open2(outCodecCtx, outCodec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
av_dump_format(outFormatCtx, 0, filename, 1);
char buff[2048] = { 0 };
av_sdp_create(&outFormatCtx, 1, buff, 1024);
printf("%s", buff);
(...SDP printed here...)
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;
/* the codec gives us the frame size, in samples,
* 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 < 44000; 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] = (int)(sin(t) * 10000);
for (k = 1; k < outCodecCtx->channels; k++)
samples[2 * j + k] = samples[2 * j];
t += tincr;
}
// 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) {
//fwrite(pkt.data, 1, pkt.size, f);
//pkt.stream_index = pktidx++;
write_frame(outFormatCtx, &outCodecCtx->time_base, outAudioStream, &pkt);
av_packet_unref(&pkt);
}
}
/* 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) {
//fwrite(pkt.data, 1, pkt.size, f);
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;
/* register all the codecs */
//avcodec_register_all();
av_register_all();
avformat_network_init(); // for network streaming
audio_encode_example("rtp://127.0.0.1:8554");
return 0;
}
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