[FFmpeg-trac] #3556(undetermined:new): bug when resampling stereo to stereo
FFmpeg
trac at avcodec.org
Mon Apr 14 10:41:07 CEST 2014
#3556: bug when resampling stereo to stereo
-------------------------------------+-------------------------------------
Reporter: olegog | Type: defect
Status: new | Priority: normal
Component: | Version:
undetermined | unspecified
Keywords: | Blocked By:
Blocking: | Reproduced by developer: 0
Analyzed by developer: 0 |
-------------------------------------+-------------------------------------
I found a bug in ffmpeg.
{{{
#include "stdafx.h"
#include <iostream>
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
//#include "swscale.h"
#include "libswresample/swresample.h"
};
FILE *fin,
*fout;
int ffmpeg_audio_decode( const char * inFile, const char * outFile)
{
// Initialize FFmpeg
av_register_all();
AVFrame* frame = avcodec_alloc_frame();
if (!frame)
{
std::cout << "Error allocating the frame" << std::endl;
return 1;
}
// you can change the file name "01 Push Me to the Floor.wav" to
whatever the file is you're reading, like "myFile.ogg" or
// "someFile.webm" and this should still work
AVFormatContext* formatContext = NULL;
//if (avformat_open_input(&formatContext, "01 Push Me to the
Floor.wav", NULL, NULL) != 0)
if (avformat_open_input(&formatContext, inFile, NULL, NULL) != 0)
{
av_free(frame);
std::cout << "Error opening the file" << std::endl;
return 1;
}
if (avformat_find_stream_info(formatContext, NULL) < 0)
{
av_free(frame);
av_close_input_file(formatContext);
std::cout << "Error finding the stream info" << std::endl;
return 1;
}
AVStream* audioStream = NULL;
// Find the audio stream (some container files can have multiple
streams in them)
for (unsigned int i = 0; i < formatContext->nb_streams; ++i)
{
if (formatContext->streams[i]->codec->codec_type ==
AVMEDIA_TYPE_AUDIO)
{
audioStream = formatContext->streams[i];
break;
}
}
if (audioStream == NULL)
{
av_free(frame);
av_close_input_file(formatContext);
std::cout << "Could not find any audio stream in the file"
<< std::endl;
return 1;
}
AVCodecContext* codecContext = audioStream->codec;
codecContext->codec =
avcodec_find_decoder(codecContext->codec_id);
if (codecContext->codec == NULL)
{
av_free(frame);
av_close_input_file(formatContext);
std::cout << "Couldn't find a proper decoder" <<
std::endl;
return 1;
}
else if (avcodec_open2(codecContext, codecContext->codec, NULL) !=
0)
{
av_free(frame);
av_close_input_file(formatContext);
std::cout << "Couldn't open the context with the decoder"
<< std::endl;
return 1;
}
std::cout << "This stream has " << codecContext->channels << "
channels and a sample rate of " << codecContext->sample_rate << "Hz" <<
std::endl;
std::cout << "The data is in the format " <<
av_get_sample_fmt_name(codecContext->sample_fmt) << std::endl;
//codecContext->sample_fmt = AV_SAMPLE_FMT_S16;
int64_t outChannelLayout =
AV_CH_LAYOUT_STEREO;//AV_CH_LAYOUT_MONO; //AV_CH_LAYOUT_STEREO;
AVSampleFormat outSampleFormat = AV_SAMPLE_FMT_S16; // Packed
audio, non-planar (this is the most common format, and probably what you
want; also, WAV needs it)
int outSampleRate = 44100;//8000;//44100;
/*
Wav wav;
wav.sampleRate = outSampleRate;
wav.sampleSize = av_get_bytes_per_sample(outSampleFormat);
wav.channels =
av_get_channel_layout_nb_channels(outChannelLayout);
*/
// Note that AVCodecContext::channel_layout may or may not be set
by libavcodec. Because of this,
// we won't use it, and will instead try to guess the layout from
the number of channels.
SwrContext* swrContext = swr_alloc_set_opts(NULL,
outChannelLayout,
outSampleFormat,
outSampleRate,
av_get_default_channel_layout(codecContext->channels),
codecContext->sample_fmt,
codecContext->sample_rate,
0,
NULL);
if (swrContext == NULL)
{
av_free(frame);
avcodec_close(codecContext);
avformat_close_input(&formatContext);
std::cout << "Couldn't create the SwrContext" <<
std::endl;
return 1;
}
if (swr_init(swrContext) != 0)
{
av_free(frame);
avcodec_close(codecContext);
avformat_close_input(&formatContext);
swr_free(&swrContext);
std::cout << "Couldn't initialize the SwrContext" <<
std::endl;
return 1;
}
fout = fopen(outFile, "wb+");
AVPacket packet;
av_init_packet(&packet);
// Read the packets in a loop
while (av_read_frame(formatContext, &packet) == 0)
{
if (packet.stream_index == audioStream->index)
{
AVPacket decodingPacket = packet;
while (decodingPacket.size > 0)
{
// Try to decode the packet into a frame
int frameFinished = 0;
int result =
avcodec_decode_audio4(codecContext, frame, &frameFinished,
&decodingPacket);
if (result < 0 || frameFinished == 0)
{
break;
}
//std::vector<unsigned char>
buffer(wav.channels * wav.sampleRate * wav.sampleSize);
unsigned char buffer[100000] = {NULL};
unsigned char* pointers[SWR_CH_MAX] =
{NULL};
pointers[0] = &buffer[0];
int numSamplesOut =
swr_convert(swrContext,
pointers,
outSampleRate, //wav.sampleRate,
(const unsigned char**)frame->extended_data,
//(const uint8_t**)frame->extended_data[0],
frame->nb_samples);
//processFrame(frame, swrContext, wav);
//fwrite( frame->data[0], sizeof(short),
(size_t)(frame->nb_samples), fout);
//fwrite( frame->extended_data,
sizeof(short), (size_t)(frame->nb_samples), fout);
//uint16_t uiCnt_1 = (uint16_t
)frame->extended_data[0];
//uint16_t uiCnt_2 = (uint16_t
)frame->extended_data[1];
/*
ReSampleContext *rs_ctx = NULL;
// resample to 44100, stereo, s16
rs_ctx = av_audio_resample_init(
1, codecContext->channels,
8000, codecContext->sample_rate,
AV_SAMPLE_FMT_S16,
codecContext->sample_fmt,
16, 10, 0, 1);
//outbuff =
(uint8_t*)av_malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
short bufferSh[100000] = {NULL};
// resampling
//int after_sampled_len =
audio_resample(rs_ctx, (short *)buffer, (short *)frame->extended_data[0],
frame->nb_samples);
int after_sampled_len =
audio_resample(rs_ctx, bufferSh, (short *)frame->extended_data,
frame->nb_samples);
*/
fwrite( (short *)buffer, sizeof(short),
(size_t)numSamplesOut, fout);
decodingPacket.size -= result;
decodingPacket.data += result;
}
/*
// Try to decode the packet into a frame
int frameFinished = 0;
avcodec_decode_audio4(codecContext, frame,
&frameFinished, &packet);
// Some frames rely on multiple packets, so we
have to make sure the frame is finished before
// we can use it
if (frameFinished)
{
//fwrite( (short *)&(frame->data[0]),
sizeof(short), (size_t)(frame->nb_samples*2), fout);
fwrite( (short *)&(frame->data[0]),
sizeof(short), (size_t)(frame->nb_samples*2), fout);
// frame now has usable audio data in it.
How it's stored in the frame depends on the format of
// the audio. If it's packed audio, all
the data will be in frame->data[0]. If it's in planar format,
// the data will be in frame->data and
possibly frame->extended_data. Look at frame->data, frame->nb_samples,
// frame->linesize, and other related
fields on the FFmpeg docs. I don't know how you're actually using
// the audio data, so I won't add any junk
here that might confuse you. Typically, if I want to find
// documentation on an FFmpeg structure or
function, I just type "<name> doxygen" into google (like
// "AVFrame doxygen" for AVFrame's docs)
}
*/
}
// You *must* call av_free_packet() after each call to
av_read_frame() or else you'll leak memory
av_free_packet(&packet);
}
// Some codecs will cause frames to be buffered up in the decoding
process. If the CODEC_CAP_DELAY flag
// is set, there can be buffered up frames that need to be
flushed, so we'll do that
if (codecContext->codec->capabilities & CODEC_CAP_DELAY)
{
av_init_packet(&packet);
// Decode all the remaining frames in the buffer, until
the end is reached
int frameFinished = 0;
while (avcodec_decode_audio4(codecContext, frame,
&frameFinished, &packet) >= 0 && frameFinished)
{
}
}
// Clean up!
av_free(frame);
avcodec_close(codecContext);
av_close_input_file(formatContext);
fclose(fout);
}
}}}
When files 02.mp3 are converted into a format 8000 pcm mono okay.
See file voice_01_sinus_8000_mono.raw.
Any discrete mono converted well.
----
Any discrete stereo converted bad.
When converting to pcm stereo 8000 it turns wrong.
See file voice_01_ sinus_ 8000_stereo.raw.
When converting to pcm 44100 stereo also turns out not correct.
See file voice_01_ sinus_ 44100_stereo.raw. Distort the shape of a
sine wave.
--
Ticket URL: <https://trac.ffmpeg.org/ticket/3556>
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