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Autres articles (64)
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Sur d’autres sites (11565)
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How do you properly free a BitStreamFilter (bsf) without getting a double free error ?
19 janvier 2021, par Alexis WilkeI'm trying to write a C++ class handling MP4 movies via ffmpeg.


First I created a couple of functions to use with
std::unique<>()so that way things get released even on exceptions.

However, I get a double free when I try to free the BitStreamFilter object, yet the documentation clearly says that each
av_bsf_alloc()must be paired with anav_bsf_free()call.



@param ctxa pointer into which the pointer to the newly-allocated context
will be written. It must be freed withav_bsf_free()after the
filtering is done.



Note : emphasis mine.


However, at the time I call the
avformat_close_input()I get a double free error, even if I haven't used the two contexts for anything !? I'm thinking that there may be a packet that both allocate and both try to free. But since these two contexts are not directly connected, I really don't understand how they end up freeing something twice.

Below is code which reproduce the error (at least on an amd64 platform). Once compiled, you can just execute it. Make sure to specify a filename as in :


./ffmpeg_demuxer_test mymovie.mp4


I used the following command to compile & link the code :


$ /usr/bin/c++ -std=c++17 -DDEBUG -D_DEBUG -D_GLIBCXX_ASSERTIONS -g -O0 \
 -fsanitize=address -fsanitize=enum -fsanitize=unreachable \
 -o ffmpeg_demuxer_test ffmpeg_demuxer_test.cpp \
 -lavformat -lavcodec


Notice the
-fsanitize=...options to capture errors such as a double free error.

Here is the code :


extern "C" {
#include <libavformat></libavformat>avformat.h>
#include <libavformat></libavformat>avio.h>
#include <libavcodec></libavcodec>avcodec.h>
}
#include <iostream>
#include <memory>


void ffmpeg_demuxer_avformat_context_free(AVFormatContext * context)
{
 if(context != nullptr) avformat_close_input(&context);
}

void ffmpeg_demuxer_av_bsf_free(AVBSFContext * context)
{
 if(context != nullptr) av_bsf_free(&context);
}


int main(int argc, char * argv [])
{
 if(argc != 2)
 {
 std::cout << "Usage: " << argv[0] << " movie.mp4" << std::endl;
 exit(1);
 }

 // init the AV libraries
 //
 av_register_all();
 avformat_network_init();

 // allocate the AVFormatContext
 //
 AVFormatContext * format_context(nullptr);
 int const r1(avformat_open_input(
 &format_context
 , argv[1]
 , nullptr // input format
 , nullptr)); // options
 if(r1 != 0
 || format_context == nullptr)
 {
 throw std::bad_alloc();
 }

 auto f_format_context = std::unique_ptr<
 AVFormatContext
 , decltype(&ffmpeg_demuxer_avformat_context_free)>(
 format_context
 , &ffmpeg_demuxer_avformat_context_free);


 // now allocate a stream
 //
 if(avformat_find_stream_info(f_format_context.get(), nullptr) < 0)
 {
 throw std::runtime_error("ffmpeg: Could not find stream info");
 }

 auto f_video_stream_index = av_find_best_stream(
 f_format_context.get()
 , AVMEDIA_TYPE_VIDEO
 , -1 // wanted stream (any)
 , -1 // related stream (none)
 , nullptr // AVCodec *
 , 0); // flags
 if(f_video_stream_index < 0)
 {
 throw std::runtime_error("ffmpeg: Could not find stream in input file");
 }
 if(static_cast<unsigned int="int">(f_video_stream_index) >= f_format_context->nb_streams)
 {
 throw std::range_error("ffmpeg: Stream index out of range");
 }

 auto f_stream = f_format_context->streams[f_video_stream_index];

 auto f_video_codec = f_stream->codecpar->codec_id;

 int f_bit_depth(0);
 switch(f_stream->codecpar->format)
 {
 case AV_PIX_FMT_YUV420P10LE:
 f_bit_depth = 10;
 break;

 case AV_PIX_FMT_YUV420P12LE:
 f_bit_depth = 12;
 break;

 default:
 f_bit_depth = 8;
 break;

 }

 bool f_mp4_h264 = f_video_codec == AV_CODEC_ID_H264 && (
 strcmp(f_format_context->iformat->long_name, "QuickTime / MOV") == 0
 || strcmp(f_format_context->iformat->long_name, "FLV (Flash Video)") == 0
 || strcmp(f_format_context->iformat->long_name, "Matroska / WebM") == 0
 );

 if(f_mp4_h264)
 {
 AVBitStreamFilter const * bsf_stream_filter(av_bsf_get_by_name("h264_mp4toannexb"));
 if(bsf_stream_filter == nullptr)
 {
 throw std::runtime_error("av_bsf_get_by_name(\"h264_mp4toannexb\") failed");
 }
 AVBSFContext * bsf_context(nullptr);
 int const r2(av_bsf_alloc(bsf_stream_filter, &bsf_context));
 if(r2 < 0
 || bsf_context == nullptr)
 {
 throw std::bad_alloc();
 }
 auto f_bsf_context = std::unique_ptr<
 AVBSFContext
 , decltype(&ffmpeg_demuxer_av_bsf_free)>(
 bsf_context
 , &ffmpeg_demuxer_av_bsf_free);
 f_bsf_context->par_in = f_stream->codecpar;
 if(av_bsf_init(f_bsf_context.get()) < 0)
 {
 throw std::runtime_error("av_bsf_init() failed");
 }
 }

 return 0;
}
</unsigned></memory></iostream>

So. Am I misreading the documentation or misusing one of the free/close function ? I don't think that the
AVBSFContextitself gets released when I close theAVFormatContext, but I may be mistaken ?

I won't put "C" as a tag since it will get removed even though this applies to C. It's not because I use C++ that I get a double free (i.e. it happens in the ffmpeg C library).


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ffmpeg radio live transcoding mp3/aac to g722
13 mai 2017, par p3tterI have tried to transcode a radio stream with ffmpeg to g722.
I get the stream to work and im able to listen to the stream.
The problem is that the output stream have faster speed than the input stream.
so the result is not good. I have tried to slow down the speed with atempo without any luck.like :
size= 241kB time=00:00:28.67 bitrate= 68.8kbits/s speed= 1.4xthis varies from 1.x to 15.x
Console output :
c:\ffmpeg\bin>ffmpeg -i http://lyd.nrk.no/nrk_radio_mp3_mp3_l -ac 1 -acodec g722 -f rtp -ab 64k -ar 16k rtp://192.168.0.99:555
ffmpeg version N-85750-ga75ef15 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 6.3.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
libavutil 55. 61.100 / 55. 61.100
libavcodec 57. 93.100 / 57. 93.100
libavformat 57. 72.101 / 57. 72.101
libavdevice 57. 7.100 / 57. 7.100
libavfilter 6. 88.100 / 6. 88.100
libswscale 4. 7.101 / 4. 7.101
libswresample 2. 8.100 / 2. 8.100
libpostproc 54. 6.100 / 54. 6.100
Input #0, mp3, from 'http://lyd.nrk.no/nrk_radio_mp3_mp3_l':
Metadata:
icy-name : NRK mP3
icy-pub : 1
Duration: N/A, start: 0.000000, bitrate: 96 kb/s
Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16p, 96 kb/s
[udp @ 000000000244bec0] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
[udp @ 00000000024781a0] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (native) -> adpcm_g722 (g722))
Press [q] to stop, [?] for help
Output #0, rtp, to 'rtp://192.168.0.99:555':
Metadata:
icy-name : NRK mP3
icy-pub : 1
encoder : Lavf57.72.101
Stream #0:0: Audio: adpcm_g722 (g722), 16000 Hz, mono, s16, 64 kb/s
Metadata:
encoder : Lavc57.93.100 g722
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 192.168.0.99
t=0 0
a=tool:libavformat 57.72.101
m=audio 555 RTP/AVP 9
b=AS:64
size= 413kB time=00:00:49.17 bitrate= 68.8kbits/s speed=1.43xDoes anyone know what im doing wrong ?
Thanks -
Why the bitrate shown during processing differs so much from the final bitrate after processing using ffmpeg ?
28 juillet 2016, par rpgmakerUsing ffmpeg, why the bitrate shown during processing differs so much from the final bitrate after processing ?
This command on a 1080p source :
ffmpeg -i "$name" -i "$subname" -map 0 -map 1 -vf scale=-1:720 -c:v libx264 -preset veryslow -crf 21 -c:a aac -b:a 256k -c:s:0 srt -disposition:s:0 default -metadata:s:s:0 language=eng -metadata:s:v:0 title="$title" "$new"Shows this output :
Which essentially averages 1300-1400 kBit/s. But when I calculate the bitrate of the resulting video using
mkvinfo -tthe bitrate shown sits at 970-1000 kBit/s and that’s not even close to averaging the 1300 kBit/s shown throughout processing. What is causing the disparity ?
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