Hi there, I think that the transcoder example <https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/transcoding.c> provided by ffmpeg works as well as some community sample code <https://github.com/leandromoreira/ffmpeg-libav-tutorial#chapter-3---transcoding> .
On Sat, Jan 25, 2020 at 7:31 PM Jonathan Noble <[email protected]> wrote: > 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? > 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@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 > _______________________________________________ > Libav-user mailing list > [email protected] > https://ffmpeg.org/mailman/listinfo/libav-user > > To unsubscribe, visit link above, or email > [email protected] with subject "unsubscribe".
_______________________________________________ Libav-user mailing list [email protected] https://ffmpeg.org/mailman/listinfo/libav-user To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
