av_write_header - Error with sample format - c

I'm writing program with libav/ffmpeg to download internet radio stream and play it on soundcard with alsa.
I've managed to download stream and extract packet and frame.
I'm having problem with av_write_header() function which (according to this https://www.ffmpeg.org/doxygen/3.2/group__lavf__encoding.html#details) I must call. It crashes and gives me the following error:
[alsa # 0x55d7ba32e580] sample format 0x15001 is not supported
Number 0x15001 is 86017 in decimal, which is index in enum AVCodecID of MP3 format(AV_CODEC_ID_MP3) used by this stream. The sample format has index 3. I can't figure out why libav parses the header wrong.
Here is a part of my code that is responsible for configuring output:
avdevice_register_all();
AVOutputFormat *output = av_guess_format("alsa",NULL,NULL);
AVFormatContext *outputFormatContext = avformat_alloc_context();
outputFormatContext->oformat = output;
outputFormatContext->flags = AVFMT_NOFILE;
AVStream *stream = avformat_new_stream(outputFormatContext,NULL);
AVCodecParameters *oCodecParameters = avcodec_parameters_alloc();
ret = avcodec_parameters_copy(oCodecParameters,iCodecParameters);
if(ret < 0){
printf("avformat_parameters_copy\n");
exit(0);
}
stream->codecpar = oCodecParameters;
if(avformat_write_header(outputFormatContext,NULL)<0){
dumpParameters(stream->codecpar);
printf("avformat_write_header\n");
exit(0);
}
The full code is here: https://github.com/szymonbarszcz99/C-internet-radio

It seems that in libav we can't do simple copy. Instead I have to manually give it requested parameters. Changing avcodec_parameters_copy() to this
AVCodecParameters *oCodecParameters = avcodec_parameters_alloc();
oCodecParameters->format = 8;
oCodecParameters->codec_type = 1;
oCodecParameters->sample_rate = 44100;
oCodecParameters->channels = 2;
stream->codecpar = oCodecParameters;
fixes this problem

Related

No-op remuxing of .avi file

I want to demux and then mux .avi file without changing anything.
My program is this (redacted for brevity):
AVFormatContext *input_format_context = NULL;
avformat_open_input(
&input_format_context,
input_url,
NULL, // fmt
NULL // options
);
avformat_find_stream_info(input_format_context, NULL);
AVFormatContext *output_format_context = NULL;
avformat_alloc_output_context2(
&output_format_context,
NULL, // oformat
NULL, // format_name
output_url
);
avio_open2(
&output_format_context->pb,
output_url,
AVIO_FLAG_WRITE,
NULL, // int_cb,
NULL // options
);
for (int i = 0; i < input_format_context->nb_streams; i++) {
avformat_new_stream(output_format_context, NULL);
AVStream *input_stream = input_format_context->streams[i];
AVStream *output_stream = output_format_context->streams[i];
AVCodecParameters *params = avcodec_parameters_alloc();
avcodec_parameters_copy(params, input_stream->codecpar);
output_stream->codecpar = params;
}
avformat_write_header(output_format_context, NULL))
AVPacket *input_packet = NULL;
input_packet = av_packet_alloc();
while (!av_read_frame(
input_format_context,
input_packet
)) {
av_write_frame(output_format_context, input_packet);
av_packet_unref(input_packet);
}
av_write_trailer(output_format_context);
Problem:
Output file is created but instead of close to 10 minute video it is a 24-second slide show consisting of around 3 frames.
It seems that the problem is (perhaps not the only one) lack of PTS on the packet.
When I explicitly print it (input_packet->pts) for each packet it is -9223372036854775808. And also the following warning is printed:
[avi # 0x562868c6c000] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
How do I then "fix my code to set the timestamps properly"?
I just found a solution.
I added this:
output_stream->time_base = input_stream->time_base;
which then, I understand, allows the video player to calculate PTS on the fly.
This does not remove the warning itself, though. I understand that .avi simply does not have PTS, so it's not a bug as such. To get rid of the warning one can manually set PTS on the packets:
input_packet->pts = calculated_ts;
I would think I should be able to also just do:
output_format_context->oformat->flags |= AVFMT_NOTIMESTAMPS;
However, I cannot do that:
error: assignment of member ‘flags’ in read-only object
So, it looks like ffmpeg is requiring PTS even for .avi or there's a bug or I'm still doing something wrong.

Split ogg vorbis stream without BOS

Input: a stream of ogg/vorbis coming from an encoder chip of an embedded system.
Problem: create output chunks of one second without transcoding.
Issue: the stream is being read "in the middle", so the first page with BOS (Beginning of Stream) is not available. Since the encoder chip has always the same parameters, I'd like to recreate the BOS page using the BOS page of a stream that was read from the start (reference stream).
I am trying to use vcut. I modified it so that it creates infinite chunks of one second. It was easy, and it works with files and streams with BOS.
I also hacked it so that I wrote to a file the first pages of the reference stream and then read them before reading the production stream with no BOS. In this way, vs->headers are populated. When I detect a page serial number change, I change it so that vcut and libogg do not freak:
int process_page(vcut_state *s, ogg_page *page) {
...
else if(vs->serial != ogg_page_serialno(page))
{
// fprintf(stderr, _("Multiplexed bitstreams are not supported.\n"));
vs->stream_in.serialno = ogg_page_serialno(page);
vs->serial = ogg_page_serialno(page);
vs->granulepos = -1;
vs->initial_granpos = 0;
// ogg_stream_init(&vs->stream_in, vs->serial);
// vorbis_info_init(&vs->vi);
// vorbis_comment_init(&vs->vc);
s->vorbis_init = 1;
}
However, this gigantic hack does not work. How to solve this issue?
It actually works: see VS1053 split ogg.
What I needed to do was to consider that starting reading in the middle of the stream, granulepos was naturally high. So it was mine logical mistake.
In process_audio_packet, I added:
int process_audio_packet(vcut_state *s,
vcut_vorbis_stream *vs, ogg_packet *packet)
{
...
if(packet->granulepos >= 0)
{
if (!firstNonZeroGranule) { // my addition
firstNonZeroGranule = 1;
vs->initial_granpos = packet->granulepos - bs;
if(vs->initial_granpos < 0)
vs->initial_granpos = 0;
} else if(vs->granulepos == 0 && packet->granulepos != bs) {
...

FFMPEG remux sample without writing to file

Let's consider this very nice and easy to use remux sample by horgh.
I'd like to achieve the same task: convert an RTSP H264 encoded stream to a fragmented MP4 stream.
This code does exactly this task.
However I don't want to write the mp4 onto disk at all, but I need to get a byte buffer or array in C with the contents that would normally written to disk.
How is that achievable?
This sample uses vs_open_output to define the output format and this function needs an output url.
If I would get rid of outputting the contents to disk, how shall I modify this code?
Or there might be better alternatives as well, those are also welcomed.
Update:
As szatmary recommended, I have checked his example link.
However as I stated in the question I need the output as buffer instead of a file.
This example demonstrates nicely how can I read my custom source and give it to ffmpeg.
What I need is how can open the input as standard (with avformat_open_input) then do my custom modification with the packets and then instead writing to file, write to a buffer.
What have I tried?
Based on szatmary's example I created some buffers and initialization:
uint8_t *buffer;
buffer = (uint8_t *)av_malloc(4096);
format_ctx = avformat_alloc_context();
format_ctx->pb = avio_alloc_context(
buffer, 4096, // internal buffer and its size
1, // write flag (1=true, 0=false)
opaque, // user data, will be passed to our callback functions
0, // no read
&IOWriteFunc,
&IOSeekFunc
);
format_ctx->flags |= AVFMT_FLAG_CUSTOM_IO;
AVOutputFormat * const output_format = av_guess_format("mp4", NULL, NULL);
format_ctx->oformat = output_format;
avformat_alloc_output_context2(&format_ctx, output_format,
NULL, NULL)
Then of course I have created 'IOWriteFunc' and 'IOSeekFunc':
static int IOWriteFunc(void *opaque, uint8_t *buf, int buf_size) {
printf("Bytes read: %d\n", buf_size);
int len = buf_size;
return (int)len;
}
static int64_t IOSeekFunc (void *opaque, int64_t offset, int whence) {
switch(whence){
case SEEK_SET:
return 1;
break;
case SEEK_CUR:
return 1;
break;
case SEEK_END:
return 1;
break;
case AVSEEK_SIZE:
return 4096;
break;
default:
return -1;
}
return 1;
}
Then I need to write the header to the output buffer, and the expected behaviour here is to print "Bytes read: x":
AVDictionary * opts = NULL;
av_dict_set(&opts, "movflags", "frag_keyframe+empty_moov", 0);
av_dict_set_int(&opts, "flush_packets", 1, 0);
avformat_write_header(output->format_ctx, &opts)
In the last line during execution, it always runs into segfault, here is the backtrace:
#0 0x00007ffff7a6ee30 in () at /usr/lib/x86_64-linux-gnu/libavformat.so.57
#1 0x00007ffff7a98189 in avformat_init_output () at /usr/lib/x86_64-linux-gnu/libavformat.so.57
#2 0x00007ffff7a98ca5 in avformat_write_header () at /usr/lib/x86_64-linux-gnu/libavformat.so.57
...
The hard thing for me with the example is that it uses avformat_open_input.
However there is no such thing for the output (no avformat_open_ouput).
Update2:
I have found another example for reading: doc/examples/avio_reading.c.
There are mentions of a similar example for writing (avio_writing.c), but ffmpeg does not have this available (at least in my google search).
Is this task really this hard to solve? standard rtsp input to custom avio?
Fortunately ffmpeg.org is down. Great.
It was a silly mistake:
In the initialization part I called this:
avformat_alloc_output_context2(&format_ctx, output_format,
NULL, NULL)
However before this I already put the avio buffers into format_ctx:
format_ctx->pb = ...
Also, this line is unnecessary:
format_ctx = avformat_alloc_context();
Correct order:
AVOutputFormat * const output_format = av_guess_format("mp4", NULL, NULL);
avformat_alloc_output_context2(&format_ctx, output_format,
NULL, NULL)
format_ctx->pb = avio_alloc_context(
buffer, 4096, // internal buffer and its size
1, // write flag (1=true, 0=false)
opaque, // user data, will be passed to our callback functions
0, // no read
&IOWriteFunc,
&IOSeekFunc
);
format_ctx->flags |= AVFMT_FLAG_CUSTOM_IO;
format_ctx->oformat = output_format; //might be unncessary too
Segfault is gone now.
You need to write a AVIOContext implementation.

How to set pts, dts and duration in ffmpeg library?

I want to pack some compressed video packets(h.264) to ".mp4" container.
One word, Muxing, no decoding and no encoding.
And I have no idea how to set pts, dts and duration.
I get the packets with "pcap" library.
I removed headers before compressed video data show up. e.g. Ethernet, VLAN.
I collected data until one frame and decoded it for getting information of data. e.g. width, height. (I am not sure that it is necessary)
I initialized output context, stream and codec context.
I started to receive packets with "pcap" library again. (now for muxing)
I made one frame and put that data in AVPacket structure.
I try to set PTS, DTS and duration. (I think here is wrong part, not sure though)
*7-1. At the first frame, I saved time(msec) with packet header structure.
*7-2. whenever I made one frame, I set parameters like this : PTS(current time - start time), DTS(same PTS value), duration(current PTS - before PTS)
I think it has some error because :
I don't know how far is suitable long for dts from pts.
At least, I think duration means how long time show this frame from now to next frame, so It should have value(next PTS - current PTS), but I can not know the value next PTS at that time.
It has I-frame only.
// make input context for decoding
AVFormatContext *&ic = gInputContext;
ic = avformat_alloc_context();
AVCodec *cd = avcodec_find_decoder(AV_CODEC_ID_H264);
AVStream *st = avformat_new_stream(ic, cd);
AVCodecContext *cc = st->codec;
avcodec_open2(cc, cd, NULL);
// make packet and decode it after collect packets is be one frame
gPacket.stream_index = 0;
gPacket.size = gPacketLength[0];
gPacket.data = gPacketData[0];
gPacket.pts = AV_NOPTS_VALUE;
gPacket.dts = AV_NOPTS_VALUE;
gPacket.flags = AV_PKT_FLAG_KEY;
avcodec_decode_video2(cc, gFrame, &got_picture, &gPacket);
// I checked automatically it initialized after "avcodec_decode_video2"
// put some info that I know that not initialized
cc->time_base.den = 90000;
cc->time_base.num = 1;
cc->bit_rate = 2500000;
cc->gop_size = 1;
// make output context with input context
AVFormatContext *&oc = gOutputContext;
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
AVFormatContext *&ic = gInputContext;
AVStream *ist = ic->streams[0];
AVCodecContext *&icc = ist->codec;
AVStream *ost = avformat_new_stream(oc, icc->codec);
AVCodecContext *occ = ost->codec;
avcodec_copy_context(occ, icc);
occ->flags |= CODEC_FLAG_GLOBAL_HEADER;
avio_open(&(oc->pb), filename, AVIO_FLAG_WRITE);
// repeated part for muxing
AVRational Millisecond = { 1, 1000 };
gPacket.stream_index = 0;
gPacket.data = gPacketData[0];
gPacket.size = gPacketLength[0];
gPacket.pts = av_rescale_rnd(pkthdr->ts.tv_sec * 1000 /
+ pkthdr->ts.tv_usec / 1000 /
- gStartTime, Millisecond.den, ost->time_base.den, /
(AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
gPacket.dts = gPacket.pts;
gPacket.duration = gPacket.pts - gPrev;
gPacket.flags = AV_PKT_FLAG_KEY;
gPrev = gPacket.pts;
av_interleaved_write_frame(gOutputContext, &gPacket);
Expected and actual results is a .mp4 video file that can play.

avformat_write_header is not working properly in ffmepg

I was working on mp4 file creation project using FFMPEG, i tried to convert the stream information of video packet based on FFMPEG muxing,remuxing code, but header get damaged after convert into so file is corrupted.
/* this code used to set the stream information */
AVFormatContext *input_context,*output_context;
AVDictionary *opt;
AVStream *out_stream;
AVCodecContext *newcontext = NULL;
out_stream= avformat_new_stream(output_context,NULL);
newcontext = avcodec_alloc_context3(codec);
newcontext->codec_id=Output_fmt->video_codec;
newcontext->bit_rate =in_stream->codec->bit_rate;
newcontext->width = in_stream->codec->width;
newcontext->height = in_stream->codec->height;
newcontext->timecode_frame_start = in_stream->codec->timecode_frame_start;
newcontext->gop_size = in_stream->codec->gop_size;
newcontext->profile = in_stream->codec->profile;
newcontext->level = in_stream->codec->level;
newcontext->pix_fmt = PIX_FMT_YUV420P;
newcontext->frame_size = in_stream->codec->frame_size;
newcontext->sample_fmt = in_stream->codec->sample_fmt;
newcontext->sample_rate = in_stream->codec->sample_rate;
time_base = (double)in_stream->time_base.num / (double)in_stream->time_base.den;
duration = (double)in_stream->duration * time_base * 1000.0;
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
return;
}
ret = avcodec_copy_context(out_stream->codec,newcontext);
if (ret < 0) {
fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
goto end;
}
out_stream->codec->codec_tag = 0;
if (output_context->oformat->flags & AVFMT_GLOBALHEADER)
out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
Changed the Header Information using:
/* this code used to set the metadata */
av_dict_set(&opt, "major_brand", "mp42", 0);
av_dict_set(&opt, "minor_version","512" , 0);
av_dict_set(&opt, "compatible_brands","isomiso2avc1mp41",0);
av_dict_set(&opt, "comment","Hash=855738390",0);
output_context->metadata = opt;
ret = avformat_write_header(output_context,NULL);
after create the mp4 file check file using ffmpeg in terminal. getting Error like this:
/this error message/
[mpeg4 # 0x7ff2b9811c00] header damaged Last message repeated 39
times [mov,mp4,m4a,3gp,3g2,mj2 # 0x7ff2ba800000] decoding for stream 0
failed [mov,mp4,m4a,3gp,3g2,mj2 # 0x7ff2ba800000] Could not find codec
parameters for stream 0 (Video: mpeg4 (mp4v / 0x7634706D), none, 376
kb/s): unspecified size Consider increasing the value for the
'analyzeduration' and 'probesize' options.
Easiest thing is to just download a freeware hex editor (for your specific O.S). Next is use desktop (commandline) version of FFmpeg (download a static build)
Use the commandline FFmpeg to convert Source to MP4 (ie: as mp4_ffmpeg.mp4)
Use your code to convert Source to MP4 (ie: as mp4_code.mp4)
Open both mp4_ffmpeg.mp4 & mp4_code.mp4 and compare bytes. The working one should be mp4_ffmpeg.mp4 so what's different from bytes produced with your code?
Things to look for :
All begin with ftyp?
moov is header and should be at start (sometimes at back after mdat which holds all a/v data in one chunk. To move the header of any mp4 to front or beginning bytes then use -movflags +faststart for example in commandline use : ffmpeg -i myfile.avi -movflags +faststart newfile.mp4)
Before each of the words moov or mdat, the previous 4 bytes are the size (in bytes) after you skip the 4 letters of word... are these sizes correct?
Do you have all the MP4 atoms (metadata sections) defined? They match what FFmpeg produced for its version of the MP4 converting?

Resources