[Libav-user] Incorrect compressed audio
Jonathan Noble
jonnobleuk at gmail.com
Mon Jan 27 22:23:57 EET 2020
On Sun, 26 Jan 2020 at 02:56, Andrew Randrianasulu <randrianasulu at gmail.com>
wrote:
> В сообщении от Sunday 26 January 2020 01:30:44 Jonathan Noble написал(а):
> > Hi,
> > I don't know why I am not getting a reply to my previous e-mails, so I am
> > trying again. Please let me know if they were there.
> >
> > I have a pipeline of reader->decoder->encoder->writer where the codec
> > parameters are the same on encoder and decoder.
> > The resultant file has audio that is distorted and considerably shorter
> > than the file that was being read from. There are no messages returned
> from
> > ffmpeg to indicate anything going wrong.
> >
> > I've spent the last week looking into this and i've not got any further
> > than at the start. Instead of posting my original code I thought i'd
> make a
> > small representation of my code use ffmpeg calls only.
> >
> > What is it that I am missing? Are my e-mails getting through?
>
> I can see your emails (so they arrive at Libav-user), but sadly can't help
> -
> myself is only curious user
>
Thanks. I wondered what was going on. I guess this question joins the
countless many in this ML that effectively is piped to /dev/null
Off to bang my head against the wall with ffmpeg libs, again.
>
> > Thanks in advance.
> > Jon Noble
> >
> > ### The code ###
> >
> > #include <assert.h>
> > #include <libavformat/avformat.h>
> > #include <libavcodec/avcodec.h>
> > #include <stdio.h>
> >
> > const char* source =
> >
> "/home/jon/Projects/Code/mediahandling/RegressionTests/ReferenceMedia/Audio/ogg/monotone.ogg";
> > const char* destination = "/tmp/vorbis.ogg";
> >
> > AVFormatContext* d_ctx = NULL;
> > AVCodec* d_codec = NULL;
> > AVCodecParameters* d_codec_params = NULL;
> > int d_stream_index = -1;
> > AVFrame* d_frame = NULL;
> >
> > AVFormatContext* e_ctx = NULL;
> > AVCodec* e_codec = NULL;
> > AVCodecContext* e_codec_ctx = NULL;
> > AVStream* e_stream = NULL;
> > AVFrame* e_frame = NULL;
> >
> >
> > int sample_count = 0;
> > int sample_rate = 0;
> >
> > int open_source()
> > {
> > d_ctx = avformat_alloc_context();
> > int code = avformat_open_input(&d_ctx, source, NULL, NULL);
> > assert(code >= 0);
> > code = avformat_find_stream_info(d_ctx, NULL);
> > assert(code >= 0);
> >
> > for (int i = 0; i < d_ctx->nb_streams; ++i)
> > {
> > AVCodecParameters* local = d_ctx->streams[i]->codecpar;
> > assert(local != NULL);
> > if (local->codec_type == AVMEDIA_TYPE_AUDIO)
> > {
> > sample_rate = local->sample_rate;
> > d_stream_index = i;
> > d_codec_params = local;
> > d_codec = avcodec_find_decoder(local->codec_id);
> > assert(d_codec != NULL);
> > av_dump_format(d_ctx,i, source, 0);
> > return 0;
> > }
> > }
> > return code;
> > }
> >
> > int setup_encoder()
> > {
> > /* allocate the output media context */
> > int ret = avformat_alloc_output_context2(&e_ctx, NULL, NULL,
> > destination);
> > assert(ret >= 0);
> >
> > AVOutputFormat *fmt = e_ctx->oformat;
> > if (fmt->audio_codec != AV_CODEC_ID_NONE)
> > {
> > e_codec = avcodec_find_encoder(fmt->audio_codec);
> > e_stream = avformat_new_stream(e_ctx, NULL);
> > e_stream->id = e_ctx->nb_streams - 1;
> > e_codec_ctx = avcodec_alloc_context3(e_codec);
> > e_codec_ctx->sample_fmt = d_codec_params->format;
> > e_codec_ctx->bit_rate = d_codec_params->bit_rate;
> > e_codec_ctx->sample_rate = d_codec_params->sample_rate;
> > e_codec_ctx->channels = d_codec_params->channels;
> > e_codec_ctx->channel_layout = d_codec_params->channel_layout;
> > e_stream->time_base = (AVRational) {1, e_codec_ctx->sample_rate};
> > if (e_ctx->oformat->flags & AVFMT_GLOBALHEADER)
> > {
> > e_codec_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
> > }
> >
> > ret = avcodec_open2(e_codec_ctx, e_codec, NULL);
> > assert(ret >= 0);
> > ret = avcodec_parameters_from_context(e_stream->codecpar,
> > e_codec_ctx);
> > assert(ret >= 0);
> >
> > av_dump_format(e_ctx, 0, destination, 1);
> > ret = avio_open(&e_ctx->pb, destination, AVIO_FLAG_WRITE);
> > assert(ret >= 0);
> >
> > int64_t sample_count = 0;
> > e_frame = av_frame_alloc();
> > e_frame->format = d_codec_params->format;
> > e_frame->channel_layout = d_codec_params->channel_layout;
> > e_frame->sample_rate = d_codec_params->sample_rate;
> > e_frame->nb_samples = e_codec_ctx->codec->capabilities &
> > AV_CODEC_CAP_VARIABLE_FRAME_SIZE ? 10000 : e_codec_ctx->frame_size;
> > ret = av_frame_get_buffer(e_frame, 0);
> > assert(ret >= 0);
> > }
> > return ret;
> > }
> >
> > int encode()
> > {
> > assert(e_frame);
> > assert(e_ctx);
> > assert(e_codec_ctx);
> > AVPacket pkt = { NULL, 0, 0 }; // data and size must be 0;
> > av_init_packet(&pkt);
> >
> > int code = av_frame_make_writable(e_frame);
> > assert(code >= 0);
> > e_frame->pts = av_rescale_q(sample_count, (AVRational){1,
> sample_rate},
> > e_codec_ctx->time_base);
> > e_frame->data[0] = d_frame->data[0];
> >
> > int got_packet;
> > code = avcodec_encode_audio2(e_codec_ctx, &pkt, e_frame,
> &got_packet);
> > assert(code >= 0);
> > if (got_packet == 1) {
> > av_packet_rescale_ts(&pkt, e_codec_ctx->time_base,
> > e_stream->time_base);
> > pkt.stream_index = e_stream->index;
> > code = av_interleaved_write_frame(e_ctx, &pkt);
> > assert(code >= 0);
> > }
> > sample_count += d_frame->nb_samples;
> > return code;
> > }
> >
> > int main()
> > {
> >
> > int code = open_source();
> > assert(code >= 0);
> > assert(d_ctx != NULL);
> > assert(d_codec != NULL);
> > assert(d_codec_params != NULL);
> > AVCodecContext* d_codec_context = avcodec_alloc_context3(d_codec);
> > assert(d_codec_context != NULL);
> > code = avcodec_parameters_to_context(d_codec_context,
> d_codec_params);
> > assert(code >= 0);
> > code = avcodec_open2(d_codec_context, d_codec, NULL);
> > assert(code >= 0);
> > d_frame = av_frame_alloc();
> > AVPacket* d_packet = av_packet_alloc();
> >
> > code = setup_encoder();
> > assert(code >= 0);
> >
> > code = avformat_write_header(e_ctx, NULL);
> > assert(code >= 0);
> >
> > // read raw packets from stream
> > while (av_read_frame(d_ctx, d_packet) >= 0)
> > {
> > if (d_packet->stream_index == d_stream_index)
> > {
> > // send packet to decoder
> > code = avcodec_send_packet(d_codec_context, d_packet);
> > assert(code >= 0);
> > while (code >= 0)
> > {
> > code = avcodec_receive_frame(d_codec_context, d_frame);
> > if (code == AVERROR(EAGAIN) || code == AVERROR_EOF)
> > {
> > break;
> > }
> > assert(code >= 0);
> > encode();
> > }
> > }
> > av_packet_unref(d_packet);
> > }
> >
> > av_frame_free(&e_frame);
> > return code;
> > }
> >
> > ### STDOUT ###
> >
> > [jon at jon-desktop ffmpegtest]$ ./a.out
> > Input #0, ogg, from
> >
> '/home/jon/Projects/Code/mediahandling/RegressionTests/ReferenceMedia/Audio/ogg/monotone.ogg':
> > Duration: 00:00:03.00, start: 0.000000, bitrate: 23 kb/s
> > Stream #0:0: Audio: vorbis, 44100 Hz, mono, fltp, 96 kb/s
> > Output #0, ogg, to '/tmp/vorbis.ogg':
> > Stream #0:0: Audio: vorbis, 44100 Hz, mono, fltp, 96 kb/s
> >
>
>
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