<div dir="ltr"><div dir="ltr"><br></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Sun, 26 Jan 2020 at 02:56, Andrew Randrianasulu <<a href="mailto:randrianasulu@gmail.com">randrianasulu@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">В сообщении от Sunday 26 January 2020 01:30:44 Jonathan Noble написал(а):<br>
> Hi,<br>
> I don't know why I am not getting a reply to my previous e-mails, so I am<br>
> trying again. Please let me know if they were there.<br>
> <br>
> I have a pipeline of reader->decoder->encoder->writer where the codec<br>
> parameters are the same on encoder and decoder.<br>
> The resultant file has audio that is distorted and considerably shorter<br>
> than the file that was being read from. There are no messages returned from<br>
> ffmpeg to indicate anything going wrong.<br>
> <br>
> I've spent the last week looking into this and i've not got any further<br>
> than at the start. Instead of posting my original code I thought i'd make a<br>
> small representation of my code use ffmpeg calls only.<br>
> <br>
> What is it that I am missing? Are my e-mails getting through?<br>
<br>
I can see your emails (so they arrive at Libav-user), but sadly can't help -<br>
myself is only curious user<br></blockquote><div>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</div><div>Off to bang my head against the wall with ffmpeg libs, again.<br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<br>
> Thanks in advance.<br>
> Jon Noble<br>
> <br>
> ### The code ###<br>
> <br>
> #include <assert.h><br>
> #include <libavformat/avformat.h><br>
> #include <libavcodec/avcodec.h><br>
> #include <stdio.h><br>
> <br>
> const char* source =<br>
> "/home/jon/Projects/Code/mediahandling/RegressionTests/ReferenceMedia/Audio/ogg/monotone.ogg";<br>
> const char* destination = "/tmp/vorbis.ogg";<br>
> <br>
> AVFormatContext* d_ctx = NULL;<br>
> AVCodec* d_codec = NULL;<br>
> AVCodecParameters* d_codec_params = NULL;<br>
> int d_stream_index = -1;<br>
> AVFrame* d_frame = NULL;<br>
> <br>
> AVFormatContext* e_ctx = NULL;<br>
> AVCodec* e_codec = NULL;<br>
> AVCodecContext* e_codec_ctx = NULL;<br>
> AVStream* e_stream = NULL;<br>
> AVFrame* e_frame = NULL;<br>
> <br>
> <br>
> int sample_count = 0;<br>
> int sample_rate = 0;<br>
> <br>
> int open_source()<br>
> {<br>
> d_ctx = avformat_alloc_context();<br>
> int code = avformat_open_input(&d_ctx, source, NULL, NULL);<br>
> assert(code >= 0);<br>
> code = avformat_find_stream_info(d_ctx, NULL);<br>
> assert(code >= 0);<br>
> <br>
> for (int i = 0; i < d_ctx->nb_streams; ++i)<br>
> {<br>
> AVCodecParameters* local = d_ctx->streams[i]->codecpar;<br>
> assert(local != NULL);<br>
> if (local->codec_type == AVMEDIA_TYPE_AUDIO)<br>
> {<br>
> sample_rate = local->sample_rate;<br>
> d_stream_index = i;<br>
> d_codec_params = local;<br>
> d_codec = avcodec_find_decoder(local->codec_id);<br>
> assert(d_codec != NULL);<br>
> av_dump_format(d_ctx,i, source, 0);<br>
> return 0;<br>
> }<br>
> }<br>
> return code;<br>
> }<br>
> <br>
> int setup_encoder()<br>
> {<br>
> /* allocate the output media context */<br>
> int ret = avformat_alloc_output_context2(&e_ctx, NULL, NULL,<br>
> destination);<br>
> assert(ret >= 0);<br>
> <br>
> AVOutputFormat *fmt = e_ctx->oformat;<br>
> if (fmt->audio_codec != AV_CODEC_ID_NONE)<br>
> {<br>
> e_codec = avcodec_find_encoder(fmt->audio_codec);<br>
> e_stream = avformat_new_stream(e_ctx, NULL);<br>
> e_stream->id = e_ctx->nb_streams - 1;<br>
> e_codec_ctx = avcodec_alloc_context3(e_codec);<br>
> e_codec_ctx->sample_fmt = d_codec_params->format;<br>
> e_codec_ctx->bit_rate = d_codec_params->bit_rate;<br>
> e_codec_ctx->sample_rate = d_codec_params->sample_rate;<br>
> e_codec_ctx->channels = d_codec_params->channels;<br>
> e_codec_ctx->channel_layout = d_codec_params->channel_layout;<br>
> e_stream->time_base = (AVRational) {1, e_codec_ctx->sample_rate};<br>
> if (e_ctx->oformat->flags & AVFMT_GLOBALHEADER)<br>
> {<br>
> e_codec_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;<br>
> }<br>
> <br>
> ret = avcodec_open2(e_codec_ctx, e_codec, NULL);<br>
> assert(ret >= 0);<br>
> ret = avcodec_parameters_from_context(e_stream->codecpar,<br>
> e_codec_ctx);<br>
> assert(ret >= 0);<br>
> <br>
> av_dump_format(e_ctx, 0, destination, 1);<br>
> ret = avio_open(&e_ctx->pb, destination, AVIO_FLAG_WRITE);<br>
> assert(ret >= 0);<br>
> <br>
> int64_t sample_count = 0;<br>
> e_frame = av_frame_alloc();<br>
> e_frame->format = d_codec_params->format;<br>
> e_frame->channel_layout = d_codec_params->channel_layout;<br>
> e_frame->sample_rate = d_codec_params->sample_rate;<br>
> e_frame->nb_samples = e_codec_ctx->codec->capabilities &<br>
> AV_CODEC_CAP_VARIABLE_FRAME_SIZE ? 10000 : e_codec_ctx->frame_size;<br>
> ret = av_frame_get_buffer(e_frame, 0);<br>
> assert(ret >= 0);<br>
> }<br>
> return ret;<br>
> }<br>
> <br>
> int encode()<br>
> {<br>
> assert(e_frame);<br>
> assert(e_ctx);<br>
> assert(e_codec_ctx);<br>
> AVPacket pkt = { NULL, 0, 0 }; // data and size must be 0;<br>
> av_init_packet(&pkt);<br>
> <br>
> int code = av_frame_make_writable(e_frame);<br>
> assert(code >= 0);<br>
> e_frame->pts = av_rescale_q(sample_count, (AVRational){1, sample_rate},<br>
> e_codec_ctx->time_base);<br>
> e_frame->data[0] = d_frame->data[0];<br>
> <br>
> int got_packet;<br>
> code = avcodec_encode_audio2(e_codec_ctx, &pkt, e_frame, &got_packet);<br>
> assert(code >= 0);<br>
> if (got_packet == 1) {<br>
> av_packet_rescale_ts(&pkt, e_codec_ctx->time_base,<br>
> e_stream->time_base);<br>
> pkt.stream_index = e_stream->index;<br>
> code = av_interleaved_write_frame(e_ctx, &pkt);<br>
> assert(code >= 0);<br>
> }<br>
> sample_count += d_frame->nb_samples;<br>
> return code;<br>
> }<br>
> <br>
> int main()<br>
> {<br>
> <br>
> int code = open_source();<br>
> assert(code >= 0);<br>
> assert(d_ctx != NULL);<br>
> assert(d_codec != NULL);<br>
> assert(d_codec_params != NULL);<br>
> AVCodecContext* d_codec_context = avcodec_alloc_context3(d_codec);<br>
> assert(d_codec_context != NULL);<br>
> code = avcodec_parameters_to_context(d_codec_context, d_codec_params);<br>
> assert(code >= 0);<br>
> code = avcodec_open2(d_codec_context, d_codec, NULL);<br>
> assert(code >= 0);<br>
> d_frame = av_frame_alloc();<br>
> AVPacket* d_packet = av_packet_alloc();<br>
> <br>
> code = setup_encoder();<br>
> assert(code >= 0);<br>
> <br>
> code = avformat_write_header(e_ctx, NULL);<br>
> assert(code >= 0);<br>
> <br>
> // read raw packets from stream<br>
> while (av_read_frame(d_ctx, d_packet) >= 0)<br>
> {<br>
> if (d_packet->stream_index == d_stream_index)<br>
> {<br>
> // send packet to decoder<br>
> code = avcodec_send_packet(d_codec_context, d_packet);<br>
> assert(code >= 0);<br>
> while (code >= 0)<br>
> {<br>
> code = avcodec_receive_frame(d_codec_context, d_frame);<br>
> if (code == AVERROR(EAGAIN) || code == AVERROR_EOF)<br>
> {<br>
> break;<br>
> }<br>
> assert(code >= 0);<br>
> encode();<br>
> }<br>
> }<br>
> av_packet_unref(d_packet);<br>
> }<br>
> <br>
> av_frame_free(&e_frame);<br>
> return code;<br>
> }<br>
> <br>
> ### STDOUT ###<br>
> <br>
> [jon@jon-desktop ffmpegtest]$ ./a.out<br>
> Input #0, ogg, from<br>
> '/home/jon/Projects/Code/mediahandling/RegressionTests/ReferenceMedia/Audio/ogg/monotone.ogg':<br>
> Duration: 00:00:03.00, start: 0.000000, bitrate: 23 kb/s<br>
> Stream #0:0: Audio: vorbis, 44100 Hz, mono, fltp, 96 kb/s<br>
> Output #0, ogg, to '/tmp/vorbis.ogg':<br>
> Stream #0:0: Audio: vorbis, 44100 Hz, mono, fltp, 96 kb/s<br>
> <br>
<br>
<br>
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