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• doc/fftools-common-opts : document ffmpeg -h bsf=bitstream_filter_name

  8 juin 2019, par Jun Zhao

  doc/fftools-common-opts : document ffmpeg -h bsf=bitstream_filter_name
  document ffmpeg -h bsf=bitstream_filter_name
  Signed-off-by : Jun Zhao <barryjzhao@tencent.com>
  

  • [DH] doc/fftools-common-opts.texi

• How to print the video meta output by the browser version of ffmpeg.wasm to the console of Google Chrome ?

  17 janvier 2021, par helloAl

  I would like to ask about how to use the browser version of ffmpeg.wasm.

  


  Through my investigation, I know that the following command can be used to output the video metadata to a file in the terminal of windows or mac.

  


  ffmpeg -i testvideo.mp4 -f ffmetadata testoutput.txt


  


  and then I can get this matadata like this :


  


  I want to parse the metadata of the video through the browser, and then print the metadata to the Google console (or output to a file). At present, I know that the browser version of ffmpeg.wasm can achieve this function, but I have looked at its examples, which does not involve this part of the content. (https://github.com/ffmpegwasm/ffmpeg.wasm/blob/master/examples/browser/image2video.html)

  


  But I want to print it to the console of Google Chrome through the browser version(usage:brower) of ffmpeg.wasm (https://github.com/ffmpegwasm/ffmpeg.wasm).
So I want to ask you how to achieve this, thank you.

  


• FLAC created using libavformat/avcodec doesn't contain meta information

  20 février 2024, par Brad Mitchell

  I have a bit of code where I needed to convert PCM16LE to a FLAC which is simple enough. I've managed to do this, however, the generated FLAC file does not contain the duration of the file.

  


  i.e. ffmpeg -i on a file showing the details :

  


  Input #0, flac, from 'test.linear_l.flac':
  Metadata:
    encoder         : Lavf57.25.100
  Duration: N/A, start: 0.000000, bitrate: N/A
  Stream #0:0: Audio: flac, 16000 Hz, stereo, s16


  


  The code I'm using to convert from PCM 16 consists of the following calls :

  


  AVFormatContext* ofmt_ctx = NULL;
avformat_alloc_output_context2(&ofmt_ctx, nullptr, nullptr, outputFile);
if (!ofmt_ctx)
{
ERROR("Unable to create output context for file: " << outputFile);
return ERROR;
}

// FLAC code
AVCodec* encoder = avcodec_find_encoder(AV_CODEC_ID_FLAC);
if (!encoder)
{
avformat_free_context(ofmt_ctx);
ERROR("FLAC encoder not found for file: " << outputFile);
return ERROR;
}

AVStream* flacStream = avformat_new_stream(ofmt_ctx, encoder);
if (!flacStream)
{
avformat_free_context(ofmt_ctx);
ERROR("Could not create new flac stream for file: " << outputFile);
return ERROR;
}

AVCodecContext* enc_ctx = flacStream->codec;
enc_ctx->sample_fmt = AV_SAMPLE_FMT_S16;
enc_ctx->sample_rate = sampleRate;
enc_ctx->channels = theMaxChannels;
enc_ctx->bit_rate = 0;

flacStream->time_base.den = enc_ctx->sample_rate;
flacStream->time_base.num = 1;

if (avcodec_open2(enc_ctx, encoder, nullptr) < 0)
{
avformat_free_context(ofmt_ctx);
ERROR("Could not open codec context for file: " << outputFile);
return ERROR;
}

if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;


if (avio_open(&ofmt_ctx->pb, outputFile, AVIO_FLAG_WRITE) < 0)
{
avformat_free_context(ofmt_ctx);
ERROR("Could not open FLAC output file: " << outputFile);
return ERROR;
}

if (avformat_write_header(ofmt_ctx, nullptr) < 0)
{
avformat_free_context(ofmt_ctx);
ERROR("Could not write FLAC file header for file: " << outputFile);
return ERROR;
}

AVPacket pkt;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;

AVFrame* frame = av_frame_alloc();
frame->nb_samples = enc_ctx->frame_size;
frame->format = enc_ctx->sample_fmt;
frame->channel_layout = enc_ctx->channel_layout;

int buffer_size = av_samples_get_buffer_size(NULL, enc_ctx->channels, enc_ctx->frame_size, enc_ctx->sample_fmt, 0);
if (buffer_size < 0)
{
avformat_free_context(ofmt_ctx);
ERROR("Could not get sample buffer size");
return ERROR;
}
uint16_t* samples = (uint16_t*)av_malloc(buffer_size);
if (!samples)
{
avformat_free_context(ofmt_ctx);
ERROR("Could not allocate " << buffer_size << " bytes for samples buffer");
return ERROR;
}

int ret = 0;

/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, enc_ctx->channels, enc_ctx->sample_fmt, (const uint8_t*)samples, buffer_size, 0);```


  


  For each block of RAW PCM16 LE data :

  


  int got_output = 0;
frame->pts = pts;
pts+= (bytesRead/2);
ret = avcodec_encode_audio2(enc_ctx, &pkt, frame, &got_output);
if (got_output) 
{
av_interleaved_write_frame(ofmt_ctx, &pkt);
av_packet_unref(&pkt);
}```


  


  Then finish off :

  


  av_write_trailer(ofmt_ctx);
for (i = 0; i < ofmt_ctx->nb_streams; i++)
{
avcodec_close(ofmt_ctx->streams[i]->codec);
}
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);   
av_freep(&samples);
av_frame_free(&frame);


  


  I'm sure I'm just missing something simple. I've tried looking at examples and doco but there isn't anything explicitly about encoding to a flac file etc in this case unless I'm missing it.