#3557 closed defect (invalid)
bug when resampling stereo to stereo
Reported by: | Oleg | Owned by: | |
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Priority: | normal | Component: | swresample |
Version: | unspecified | Keywords: | |
Cc: | Blocked By: | ||
Blocking: | Reproduced by developer: | no | |
Analyzed by developer: | no |
Description
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.
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.
Attachments (8)
Change History (16)
by , 11 years ago
by , 11 years ago
Attachment: | voice_01_sinus_8000_mono.JPG added |
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by , 11 years ago
Attachment: | voice_01_sinus_8000_stereo.raw added |
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by , 11 years ago
Attachment: | voice_01_sinus_8000_mono.raw added |
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by , 11 years ago
Attachment: | voice_01_sinus_44100_stereo.raw added |
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by , 11 years ago
Attachment: | voice_01_sinus_8000_stereo.JPG added |
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by , 11 years ago
Attachment: | voice_01_sinus_44100_stereo.JPG added |
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by , 11 years ago
Attachment: | decode_audio_test.cpp added |
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comment:1 by , 11 years ago
Component: | undetermined → swresample |
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Priority: | critical → normal |
comment:2 by , 11 years ago
only
ffmpeg-20140414-git-5e379cd-win32-dev
ffmpeg-20140414-git-5e379cd-win32-shared
comment:4 by , 11 years ago
ffmpeg.exe convert and resampling ok
What's the problem?
Why not get on С++ ?
comment:5 by , 11 years ago
f:\>ffmpeg -i 02.mp3 -ar 8000 -ac 2 out.wav
ffmpeg version N-62439-g5e379cd Copyright (c) 2000-2014 the FFmpeg developers
built on Apr 13 2014 22:01:33 with gcc 4.8.2 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp
ack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable
-zlib
libavutil 52. 76.100 / 52. 76.100
libavcodec 55. 58.103 / 55. 58.103
libavformat 55. 37.100 / 55. 37.100
libavdevice 55. 13.100 / 55. 13.100
libavfilter 4. 4.100 / 4. 4.100
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
[mp3 @ 02c8eca0] Header missing
[mp3 @ 02c8aba0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from '02.mp3':
Duration: 00:00:39.93, start: 0.000000, bitrate: 128 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Output #0, wav, to 'out.wav':
Metadata:
ISFT : Lavf55.37.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, stereo, s16,
256 kb/s
Stream mapping:
Press [q] to stop, ? for help
[mp3 @ 02c8eca0] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
size= 1249kB time=00:00:39.99 bitrate= 255.8kbits/s
video:0kB audio:1249kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.006255%
comment:6 by , 11 years ago
ffmpeg -i 02.mp3 -ar 8000 -ac 2 -f s16le out.pcm
convert and resampling ok
comment:7 by , 11 years ago
I found a bug in my program.
from
fwrite( (short *)buffer, sizeof(short), (size_t)numSamplesOut, fout);
to
fwrite( (short *)buffer, sizeof(short), (size_t)numSamplesOut*frame->channels, fout);
ffmpeg has no errors.
topic can be closed.
comment:8 by , 11 years ago
Resolution: | → invalid |
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Status: | new → closed |
Is this not reproducible with
ffmpeg
(the application)?Did you test current git head or another version?