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Organiser par catégorie
17 mai 2013, parDans MédiaSPIP, une rubrique a 2 noms : catégorie et rubrique.
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Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
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libswresample : Why does swr_init() change |in_ch_layout| order so it no longer matches my decoded AVFrames, causing resampling to fail ?
20 novembre 2023, par CheekyChipsI am trying to write some code that resamples an audio file to 16kHz and 1 channel and then encodes it to PCM, but I am having an issue with channel layouts.


In a nutshell :


My
AVCodecContext
and the frames I get from the stream viaavcodec_receive_frame()
have a channel layout order ofAV_CHANNEL_ORDER_UNSPEC
. But when I callswr_init()
it changes thein_ch_layout
order toAV_CHANNEL_ORDER_NATIVE
. Then when I callswr_convert_frame()
with myAVFrame
s, because the channel layout orders don't match, the resampling fails because it thinks the input changed.

More details :


I create an
AVCodecContext
from my audio stream's codec, and it has a channel layout ofAV_CHANNEL_ORDER_UNSPEC
with 2 channels, and any frames I decode from the stream viaavcodec_receive_frame()
also have a channel layout order ofAV_CHANNEL_ORDER_UNSPEC
.

I set
SwrContext
's|in_ch_layout|
to the sample channel layout from the codec context :

AVChannelLayout in_ch_layout = in_codec_context->ch_layout,
 ...
 int ret = swr_alloc_set_opts2(&swr_ctx, ...
 &in_ch_layout,
 ...);



But
SwrContext->init()
changes its internalin_ch_layout
fromAV_CHANNEL_ORDER_UNSPEC
toAV_CHANNEL_ORDER_NATIVE
meaning it fails the next time I callswr_convert_frame()
because the input frame has a different channel layout to theSwrContext
. Whenswr_init()
is called (in my case indirectly byswr_convert_frame()
, but also if I alternatively call it directly) theSwrContext->used_ch_layout
andSwrContext->in_ch_layout
are updated to have channel layout order ofAV_CHANNEL_ORDER_NATIVE
:

// swresample.c
 av_cold int swr_init(struct SwrContext *s){
 ...
 if (!av_channel_layout_check(&s->used_ch_layout)) <-- This hits if I don't set anything for used_ch_layout
 av_channel_layout_default(&s->used_ch_layout, s->in.ch_count); <-- default is AV_CHANNEL_ORDER_NATIVE
 ...
 if (s->used_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) <-- This hits if I do set used_ch_layout
 av_channel_layout_default(&s->used_ch_layout, s->used_ch_layout.nb_channels); <-- default is AV_CHANNEL_ORDER_NATIVE



Then when I next call
swr_convert_frame()
, because the frame has the same layout as the audio stream's codec (AV_CHANNEL_ORDER_UNSPEC
), and this is different toSwrContext->in_ch_layout
(AV_CHANNEL_ORDER_NATIVE
), it early exits withret |= AVERROR_INPUT_CHANGED
.

// swresample_frame.c
 int swr_convert_frame(SwrContext *s,
 AVFrame *out, const AVFrame *in)
 {
 ...
 if ((ret = config_changed(s, out, in)))
 return ret;
 ...



static int config_changed(SwrContext *s,
 const AVFrame *out, const AVFrame *in)
 {
 ...
 if ((err = av_channel_layout_copy(&ch_layout, &in->ch_layout)) < 0)
 ...
 if (av_channel_layout_compare(&s->in_ch_layout, &ch_layout) || ...) { <-- This hits the next time I call swr_convert_frame()
 ret |= AVERROR_INPUT_CHANGED;
 }



// channel_layout.c
 int av_channel_layout_compare(const AVChannelLayout *chl, const AVChannelLayout *chl1)
 {
 ...
 // if only one is unspecified -> not equal
 if ((chl->order == AV_CHANNEL_ORDER_UNSPEC) !=
 (chl1->order == AV_CHANNEL_ORDER_UNSPEC))
 return 1;



If I hardcode the channel layout order of each input
AVFrame
toAV_CHANNEL_ORDER_NATIVE
before resampling, then the resampling and subsequent encoding works, but this feels like a really bad idea and of course wouldn't work as soon as I resample an audio file with a different channel layout.

avcodec_receive_frame(in_codec_context, input_frame);

 AVChannelLayout input_frame_ch_layout;
 av_channel_layout_default(&input_frame_ch_layout, 2 /* = nb_channels*/);
 input_frame->ch_layout = input_frame_ch_layout;
 // Bad idea - but "fixes" my issue!



My questions


What do I need to do to the resampler OR/AND the decoded audio frame to make sure they have the same channel layout order and the resampling works ?


How can I make the channel order of the
AVFrame
s that I get fromavcodec_receive_frame()
match the input channel order ofSwrContext
so the resampling works ? My understanding is that the decoded frames should be 'correct' already and I shouldn't need to change any of their values, only values of the output (resampled) frames that I create.

Is there something I need to set on the
AVFrame
before I resample it ?

Why does the
SwrContext
choose to change the channel order toAV_CHANNEL_ORDER_NATIVE
?

Note :
A workaround could be to use
swr_convert()
with the raw data buffer instead ofswr_convert_frame()
, since it looks like it bypasses this check (since there are no frames involved). I haven't tried this but this shouldn't be necessary and I would like to useswr_convert_frame()
as I am working with input and output frames.

Unfortunately I can't find example code using
swr_convert_frame()
(not even the ffmpeg code seems to ever call it).

My full c++ source code
(error handling omitted for readability) :


std::string fileToUse = "/home/projects/audioFileProject/Audio files/14 Black Cadillacs.wma";
const std::string outputFilename = "out.wav";
const std::string PCMS16BE_encoder_name = "pcm_f32le";

int main()
{
 // Open audio file
 AVFormatContext* in_format_context = avformat_alloc_context();
 avformat_open_input(&in_format_context, fileToUse.c_str(), NULL, NULL);
 avformat_find_stream_info(in_format_context, NULL);
 
 // Get audio stream from file and corresponding decoder
 AVStream* in_stream = in_format_context->streams[0];
 AVCodecParameters* codec_params = in_stream->codecpar;
 const AVCodec* in_codec = avcodec_find_decoder(codec_params->codec_id);
 AVCodecContext *in_codec_context = avcodec_alloc_context3(in_codec);
 avcodec_parameters_to_context(in_codec_context, codec_params);
 avcodec_open2(in_codec_context, in_codec, NULL);

 // Prepare output stream and output encoder (PCM)
 AVFormatContext* out_format_context = nullptr;
 avformat_alloc_output_context2(&out_format_context, NULL, NULL, outputFilename.c_str());
 AVStream* out_stream = avformat_new_stream(out_format_context, NULL);
 const AVCodec* output_codec = avcodec_find_encoder_by_name(PCMS16BE_encoder_name.c_str());
 AVCodecContext* output_codec_context = avcodec_alloc_context3(output_codec);

 // -------------------------------
 
 AVChannelLayout output_ch_layout;
 av_channel_layout_default(&output_ch_layout, 1); // AV_CHANNEL_LAYOUT_MONO
 output_codec_context->ch_layout = output_ch_layout;
 
 auto out_sample_rate = 16000;
 output_codec_context->sample_rate = out_sample_rate;
 output_codec_context->sample_fmt = output_codec->sample_fmts[0];
 //output_codec_context->bit_rate = output_codec_context->bit_rate; // TODO Do we need to set the bit rate?
 output_codec_context->time_base = (AVRational){1, out_sample_rate};
 out_stream->time_base = output_codec_context->time_base;

 auto in_sample_rate = in_codec_context->sample_rate;
 AVChannelLayout in_ch_layout = in_codec_context->ch_layout,
 out_ch_layout = output_ch_layout; // AV_CHANNEL_LAYOUT_MONO;
 enum AVSampleFormat in_sample_fmt = in_codec_context->sample_fmt,
 out_sample_fmt = in_codec_context->sample_fmt;

 SwrContext *swr_ctx = nullptr;
 int ret = swr_alloc_set_opts2(&swr_ctx,
 &out_ch_layout,
 out_sample_fmt,
 out_sample_rate,
 &in_ch_layout,
 in_sample_fmt,
 in_sample_rate,
 0, // log_offset
 NULL); // log_ctx

 // Probably not necessary - documentation says "This option is
only used for special remapping."
 av_opt_set_chlayout(swr_ctx, "used_chlayout", &in_ch_layout, 0);

 // Open output file for writing
 avcodec_open2(output_codec_context, output_codec, NULL);
 avcodec_parameters_from_context(out_stream->codecpar, output_codec_context);
 
 if (out_format_context->oformat->flags & AVFMT_GLOBALHEADER)
 out_format_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

 avio_open(&out_format_context->pb, outputFilename.c_str(), AVIO_FLAG_WRITE);
 AVDictionary* muxer_opts = nullptr;
 avformat_write_header(out_format_context, &muxer_opts);

 AVFrame* input_frame = av_frame_alloc();
 AVPacket* in_packet = av_packet_alloc();

 // Loop through decoded input frames. Resample and get resulting samples in a new output frame.
 // I think PCM supports variable number of samples in frames so probably can immediately write out
 while (av_read_frame(in_format_context, in_packet) >= 0) {
 avcodec_send_packet(in_codec_context, in_packet);
 avcodec_receive_frame(in_codec_context, input_frame);

 // I don't want to do this, but it 'fixes' the error where channel layout of input frames
 // doesn't match what the resampler expects - hardcoded the number 2 to fit my sample audio file.
 AVChannelLayout input_frame_ch_layout;
 av_channel_layout_default(&input_frame_ch_layout, 2 /* = nb_channels*/);
 input_frame->ch_layout = input_frame_ch_layout;

 AVFrame* output_frame = av_frame_alloc();
 output_frame->sample_rate = out_sample_rate;
 output_frame->format = out_sample_fmt;
 output_frame->ch_layout = out_ch_layout;
 output_frame->nb_samples = output_codec_context->frame_size;
 
 // TODO Probably need to do maths to calculate new pts properly
 output_frame->pts = input_frame->pts;

 if (swr_convert_frame(swr_ctx, output_frame, input_frame))
 { logging("Swr Convert failed"); return -1; } 
 /// ^ Fails here, the second time (since the first time init() is called internally)

 AVPacket *output_packet = av_packet_alloc();
 int response = avcodec_send_frame(output_codec_context, output_frame);

 while (response >= 0) {
 response = avcodec_receive_packet(output_codec_context, output_packet);

 if (response == AVERROR(EAGAIN) || response == AVERROR_EOF) {
 break;
 }

 output_packet->stream_index = 0;
 av_packet_rescale_ts(output_packet, in_stream->time_base, out_stream->time_base);
 av_interleaved_write_frame(out_format_context, output_packet);
 }
 av_packet_unref(output_packet);
 av_packet_free(&output_packet);
 av_frame_unref(input_frame); // Free references held by the frame before reading new data into it.
 av_frame_unref(output_frame);
 }
 // TODO write last output packet flushing the buffer

 avformat_close_input(&in_format_context);
 return 0;
}



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Homepage Design : Best Practices & Examples
5 octobre 2022, par Erin -
avdevice/decklink_enc : Add support for compressed AC-3 output over SDI
7 avril 2023, par Devin Heitmuelleravdevice/decklink_enc : Add support for compressed AC-3 output over SDI
Extend the decklink output to include support for compressed AC-3,
encapsulated using the SMPTE ST 377:2015 standard.This functionality can be exercised by using the "copy" codec when
the input audio stream is AC-3. For example :./ffmpeg -i /foo.ts -codec:a copy -f decklink 'UltraStudio Mini Monitor'
Note that the default behavior continues to be to do PCM output,
which means without specifying the copy codec a stream containing
AC-3 will be decoded and downmixed to stereo audio before output.Thanks to Marton Balint for providing feedback.
Signed-off-by : Devin Heitmueller <dheitmueller@ltnglobal.com>
Signed-off-by : Marton Balint <cus@passwd.hu>