[FFmpeg-trac] #3557(undetermined:new): bug when resampling stereo to stereo
FFmpeg
trac at avcodec.org
Mon Apr 14 11:17:19 CEST 2014
#3557: bug when resampling stereo to stereo
-------------------------------------+-------------------------------------
Reporter: olegog | Type: defect
Status: new | Priority: critical
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_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 = 8000;//44100;
// 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;
}
unsigned char buffer[100000] = {NULL};
unsigned char* pointers[SWR_CH_MAX] =
{NULL};
pointers[0] = &buffer[0];
int numSamplesOut = swr_convert(
swrContext,
pointers,
outSampleRate,
(const unsigned char**)frame->extended_data,
frame->nb_samples);
fwrite(
(short *)buffer,
sizeof(short),
(size_t)numSamplesOut,
fout);
decodingPacket.size -= result;
decodingPacket.data += result;
}
}
// 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.
[[Image(voice_01_sinus_8000_mono.JPG)]]
----
Any discrete stereo converted bad.
When converting to pcm stereo 8000 it turns wrong.
See file voice_01_ sinus_ 8000_stereo.raw.
[[Image(voice_01_sinus_8000_stereo.JPG)]]
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.
[[Image(voice_01_sinus_44100_stereo.JPG)]]
--
Ticket URL: <https://trac.ffmpeg.org/ticket/3557>
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