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Matmos - Action at a Distance
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Getting Access Violations Exceptions in FFmpeg when trying to free IO buffer
26 juin 2015, par PatrikI’ve been trying to create an audio stream extractor/demuxer using FFmpeg.AutoGen and C#. While the code actually works (in the sense that it extracts the audio stream correctly), I can’t seem to clean up the resources afterwards. Whenever I try to free my input buffer (allocated using av_malloc) I get an Access Violation Exception, and if I don’t free it, I seem to be leaking memory corresponding to the size of the allocated input buffer.
This is my code (from a console application I used for testing) :
class Program
{
static void Main(string[] args)
{
using (var videoStream = Assembly.GetExecutingAssembly().GetManifestResourceStream("FFmpeg.Demux.test.mp4"))
using (MemoryStream output = new MemoryStream())
{
FFmpegAudioExtractor audioExtractor = new FFmpegAudioExtractor();
audioExtractor.Extract(videoStream, output);
Debug.Assert(1331200 == output.Length);
//File.WriteAllBytes(@"c:\temp\out.pcm", output.ToArray());
}
Console.WriteLine("Done");
Console.ReadLine();
}
}
public class FFmpegAudioExtractor
{
public FFmpegAudioExtractor()
{
FFmpegInvoke.av_register_all();
FFmpegInvoke.avcodec_register_all();
}
public unsafe void Extract(Stream input, Stream output)
{
AVFormatContext* inFormatContextPtr = null;
int inFomatContextOpenResult = -1;
byte* inIoBuffer = null;
AVIOContext* inIoContextPtr = null;
SwrContext* swrContextPtr = null;
AVFrame* inFramePtr = null;
AVFrame* outFramePtr = null;
AVCodecContext* inCodecContextPtr = null;
try
{
/* 1 */
inFormatContextPtr = FFmpegInvoke.avformat_alloc_context();
if (inFormatContextPtr == null)
throw new ApplicationException("Failed to allocate the input format context (AVFormatContext).");
// HACK this should alloc a fixed buffer and use callbacks to fill it
if (input.Length > int.MaxValue)
throw new ArgumentException("Data too large.", "input");
// TEST alloc a 10MB buffer to make the memory leak real obvious
int inIoBufferSize = 1024 * 1024 * 10; //(int)input.Length;
/* 2 */
inIoBuffer = (byte*)FFmpegInvoke.av_malloc((uint)inIoBufferSize);
if (inIoBuffer == null)
throw new ApplicationException("Failed to allocate the input IO buffer.");
// HACK continued, fill buffer
IntPtr inIoBufferPtr = new IntPtr(inIoBuffer);
byte[] buffer = new byte[4096];
int read, offset = 0;
while ((read = input.Read(buffer, 0, buffer.Length)) > 0)
{
Marshal.Copy(buffer, 0, inIoBufferPtr + offset, read);
offset += read;
}
/* 3 */
inIoContextPtr = FFmpegInvoke.avio_alloc_context((sbyte*)inIoBuffer,
inIoBufferSize,
0 /* writable */,
null,
IntPtr.Zero,
IntPtr.Zero,
IntPtr.Zero);
if (inIoContextPtr == null)
throw new ApplicationException("Failed to allocate the input IO context (AVIOContext).");
// configure the format context to use our custom IO contect
inFormatContextPtr->pb = inIoContextPtr;
inFormatContextPtr->flags = FFmpegInvoke.AVFMT_FLAG_CUSTOM_IO;
/* 35 */
inFomatContextOpenResult = FFmpegInvoke.avformat_open_input(&inFormatContextPtr, "", null, null);
if (inFomatContextOpenResult != 0)
throw new ApplicationException("Could not open input: " + inFomatContextOpenResult.ToString(CultureInfo.InvariantCulture));
// retrieve stream information
int avformatFindStreamInfoResult = FFmpegInvoke.avformat_find_stream_info(inFormatContextPtr, null);
if (avformatFindStreamInfoResult < 0)
throw new ApplicationException("Failed to locate stream info: " + avformatFindStreamInfoResult.ToString(CultureInfo.InvariantCulture));
// find audio stream
int inAudioStreamIndex = FFmpegInvoke.av_find_best_stream(inFormatContextPtr,
AVMediaType.AVMEDIA_TYPE_AUDIO,
-1 /* wanted_stream_nb */,
-1 /* related_stream */,
null /* [out] decoder */,
0 /* flags */);
if (inAudioStreamIndex < 0)
throw new ApplicationException("Failed to find audio stream: " + inAudioStreamIndex.ToString(CultureInfo.InvariantCulture));
// get audio stream pointer
AVStream* inAudioStreamPtr = inFormatContextPtr->streams[inAudioStreamIndex];
Contract.Assume(inAudioStreamPtr != null);
// find the decoder
AVCodec* inCodecPtr = FFmpegInvoke.avcodec_find_decoder(inAudioStreamPtr->codec->codec_id);
if (inCodecPtr == null)
throw new ApplicationException("Failed to find decoder with codec_id: " + inAudioStreamPtr->codec->codec_id.ToString());
/* 36 */
inCodecContextPtr = FFmpegInvoke.avcodec_alloc_context3(inCodecPtr);
if (FFmpegInvoke.avcodec_copy_context(inCodecContextPtr, inAudioStreamPtr->codec) != 0)
throw new ApplicationException("Failed to copy context.");
// open codec
/* 37 */
if (FFmpegInvoke.avcodec_open2(inCodecContextPtr, inCodecPtr, null) < 0)
{
string codecName = Marshal.PtrToStringAuto(new IntPtr(inCodecPtr->name));
throw new Exception("Failed to open codec: " + codecName);
}
// alloc frame
/* 38 */
inFramePtr = FFmpegInvoke.av_frame_alloc();
if (inFramePtr == null)
throw new ApplicationException("Could not allocate frame");
// initialize packet, set data to NULL, let the demuxer fill it
AVPacket packet = new AVPacket();
AVPacket* packetPtr = &packet;
FFmpegInvoke.av_init_packet(packetPtr);
packetPtr->data = null;
packetPtr->size = 0;
// alloc SWR
/* 4 */
swrContextPtr = FFmpegInvoke.swr_alloc();
AVSampleFormat outSampleFormat = AVSampleFormat.AV_SAMPLE_FMT_S16;
long outChannelLayout = FFmpegInvoke.AV_CH_FRONT_LEFT | FFmpegInvoke.AV_CH_FRONT_RIGHT;
// configure SWR
FFmpegInvoke.av_opt_set_sample_fmt(swrContextPtr, "in_sample_fmt", inCodecContextPtr->sample_fmt, 0);
FFmpegInvoke.av_opt_set_sample_fmt(swrContextPtr, "out_sample_fmt", outSampleFormat, 0);
// setting the in_channel_layout seems to break things
//FFmpegInvoke.av_opt_set_channel_layout(swrContextPtr, "in_channel_layout", (long)inCodecContextPtr->channel_layout, 0);
FFmpegInvoke.av_opt_set_channel_layout(swrContextPtr, "out_channel_layout", outChannelLayout, 0);
FFmpegInvoke.av_opt_set_int(swrContextPtr, "in_sample_rate", inCodecContextPtr->sample_rate, 0);
FFmpegInvoke.av_opt_set_int(swrContextPtr, "out_sample_rate", 44100, 0);
// allock output frane
/* 45 */
outFramePtr = FFmpegInvoke.av_frame_alloc();
outFramePtr->channel_layout = (ulong)outChannelLayout;
outFramePtr->sample_rate = 44100;
outFramePtr->format = (int)outSampleFormat;
// config done, init
FFmpegInvoke.swr_init(swrContextPtr);
bool frameFinished;
// read frames from the file
while (FFmpegInvoke.av_read_frame(inFormatContextPtr, packetPtr) >= 0)
{
AVPacket* origPacketPtr = packetPtr;
try
{
if (packetPtr->stream_index != inAudioStreamIndex)
continue;
do
{
byte[] decodedFrame;
int decodedBytes = this.DecodePacket(inCodecContextPtr,
packetPtr,
inFramePtr,
swrContextPtr,
outFramePtr,
out frameFinished,
out decodedFrame);
if (decodedBytes < 0)
break;
output.Write(decodedFrame, 0, decodedFrame.Length);
packetPtr->data += decodedBytes;
packetPtr->size -= decodedBytes;
} while (packetPtr->size > 0);
}
finally
{
FFmpegInvoke.av_free_packet(origPacketPtr);
}
}
// flush cached frames
packetPtr->data = null;
packetPtr->size = 0;
do
{
byte[] decodedFrame;
this.DecodePacket(inCodecContextPtr, packetPtr, inFramePtr, swrContextPtr, outFramePtr, out frameFinished, out decodedFrame);
if (decodedFrame != null)
output.Write(decodedFrame, 0, decodedFrame.Length);
} while (frameFinished);
}
finally
{
/* 45 */
if (outFramePtr != null)
FFmpegInvoke.av_frame_free(&outFramePtr);
/* 4 */
if (swrContextPtr != null)
FFmpegInvoke.swr_free(&swrContextPtr);
/* 38 */
if (inFramePtr != null)
FFmpegInvoke.av_frame_free(&inFramePtr);
/* 37 */
if (inCodecContextPtr != null)
FFmpegInvoke.avcodec_close(inCodecContextPtr);
/* 36 */
if (inCodecContextPtr != null)
FFmpegInvoke.avcodec_free_context(&inCodecContextPtr);
/* 35 */
if (inFomatContextOpenResult == 0)
FFmpegInvoke.avformat_close_input(&inFormatContextPtr);
/* 3 */
if (inIoContextPtr != null)
FFmpegInvoke.av_freep(&inIoContextPtr);
//* 2 */
if (inIoBuffer != null)
FFmpegInvoke.av_freep(&inIoBuffer);
/* 1 */
// This is called by avformat_close_input
if (inFormatContextPtr != null)
FFmpegInvoke.avformat_free_context(inFormatContextPtr);
}
}
private unsafe int DecodePacket(AVCodecContext* audioDecoderContextPtr, AVPacket* packetPtr, AVFrame* inFramePtr, SwrContext* swrContextPtr, AVFrame* outFramePtr, out bool frameDecoded, out byte[] decodedFrame)
{
decodedFrame = null;
/* decode audio frame */
int gotFrame;
int readBytes = FFmpegInvoke.avcodec_decode_audio4(audioDecoderContextPtr, inFramePtr, &gotFrame, packetPtr);
if (readBytes < 0)
throw new ApplicationException("Error decoding audio frame: " + readBytes.ToString(CultureInfo.InvariantCulture));
frameDecoded = gotFrame != 0;
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
int decoded = Math.Min(readBytes, packetPtr->size);
if (frameDecoded)
{
if (FFmpegInvoke.swr_convert_frame(swrContextPtr, outFramePtr, inFramePtr) != 0)
throw new ApplicationException("Failed to convert frame.");
long delay;
do
{
int unpaddedLinesize = outFramePtr->nb_samples * outFramePtr->channels * FFmpegInvoke.av_get_bytes_per_sample((AVSampleFormat)outFramePtr->format);
IntPtr dataPtr = new IntPtr(outFramePtr->extended_data[0]);
decodedFrame = new byte[unpaddedLinesize];
Marshal.Copy(dataPtr, decodedFrame, 0, unpaddedLinesize);
// check if we have more samples to convert for this frame
delay = FFmpegInvoke.swr_get_delay(swrContextPtr, 44100);
if (delay > 0)
{
if (FFmpegInvoke.swr_convert_frame(swrContextPtr, outFramePtr, null) != 0)
throw new ApplicationException("Failed to convert frame.");
}
} while (delay > 0);
}
return decoded;
}
}The failing line is :
FFmpegInvoke.av_freep(&inIoBuffer); -
FFmpeg matlab error : At least one output file must be specified ? [closed]
3 mars, par as mohI'm trying to get I frames from a video using Matlab using this command
system(sprintf('ffmpeg -i testVid.mp4 -vf "select=eq(pict_type\,I)" -vsync vfr output_%03d.png'));,but i get this message

ffmpeg version 7.1-full_build-www.gyan.dev Copyright (c) 2000-2024 the FFmpeg developers 
 built with gcc 14.2.0 (Rev1, Built by MSYS2 project) 
 configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libaribcaption --enable-libdav1d --enable-libdavs2 --enable-libopenjpeg --enable-libquirc --enable-libuavs3d --enable-libxevd --enable-libzvbi --enable-libqrencode --enable-librav1e --enable-libsvtav1 --enable-libvvenc --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxeve --enable-libxvid --enable-libaom --enable-libjxl --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libcodec2 --enable-libilbc --enable-libgsm --enable-liblc3 --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint 
 libavutil 59. 39.100 / 59. 39.100 
 libavcodec 61. 19.100 / 61. 19.100 
 libavformat 61. 7.100 / 61. 7.100 
 libavdevice 61. 3.100 / 61. 3.100 
 libavfilter 10. 4.100 / 10. 4.100 
 libswscale 8. 3.100 / 8. 3.100 
 libswresample 5. 3.100 / 5. 3.100 
 libpostproc 58. 3.100 / 58. 3.100 
Trailing option(s) found in the command: may be ignored. 
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testVid.mp4': 
 Metadata: 
 major_brand : isom 
 minor_version : 512 
 compatible_brands: isomiso2avc1mp41 
 encoder : Lavf57.83.100 
 Duration: 00:00:02.02, start: 0.000000, bitrate: 12798 kb/s 
 Stream #0:0[0x1](eng): Video: h264 (Baseline) (avc1 / 0x31637661), yuvj420p(pc, progressive), 1280x720 [SAR 1:1 DAR 16:9], 12662 kb/s, 29.74 fps, 30 tbr, 90k tbn (default) 
 Metadata: 
 handler_name : VideoHandler 
 vendor_id : [0][0][0][0] 
 Stream #0:1[0x2](eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 121 kb/s (default) 
 Metadata: 
 handler_name : SoundHandler 
 vendor_id : [0][0][0][0] 
At least one output file must be specified 


i searched and tried many cases but i don't know where is the problem, any help please ?


-
ffmpeg Non monotonous DTS, Previous DTS is always the same, audio microphone streaming [closed]
5 février, par adrien gonzalezI'm using ffmpeg to stream audio from a microphone using rtp. I'm on Raspberry and use an external sound card (HifiBerry DAC + ADC Pro).
My goal is to stream audio with the lowest latency possible to others Raspberry reading this audio with ffplay. I try not to compress the audio flux and leave it untouched as wav 48000 Hz.
I encounter often some Non Monotonous DTS errors. When this happens I have a latency of hundred of milliseconds adding itself.
I tried to add the +igndts flag but it is not changing anything. Also tried +genpts flag.


What is weird is that the previous DTS is always the same (201165 is the example below) and does not seems to change.
I looked on forums for answers but I'm unable to find one.


Here is my bash command :


ffmpeg -guess_layout_max 0 -re -f alsa -i hw -acodec pcm_s16le -ac 1 -payload_type 10 -f rtp rtp://192.168.1.152:5003

And the result from the terminal :


ffmpeg version 5.1.6-0+deb12u1+rpt1 Copyright (c) 2000-2024 the FFmpeg developers


built with gcc 12 (Debian 12.2.0-14)
 configuration: --prefix=/usr --extra-version=0+deb12u1+rpt1 --toolchain=hardened --incdir=/usr/include/aarch64-linux-gnu --enable-gpl --disable-stripping --disable-mmal --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libglslang --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librist --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sand --enable-sdl2 --disable-sndio --enable-libjxl --enable-neon --enable-v4l2-request --enable-libudev --enable-epoxy --libdir=/usr/lib/aarch64-linux-gnu --arch=arm64 --enable-pocketsphinx --enable-librsvg --enable-libdc1394 --enable-libdrm --enable-vout-drm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-libplacebo --enable-librav1e --enable-shared
 libavutil 57. 28.100 / 57. 28.100
 libavcodec 59. 37.100 / 59. 37.100
 libavformat 59. 27.100 / 59. 27.100
 libavdevice 59. 7.100 / 59. 7.100
 libavfilter 8. 44.100 / 8. 44.100
 libswscale 6. 7.100 / 6. 7.100
 libswresample 4. 7.100 / 4. 7.100
 libpostproc 56. 6.100 / 56. 6.100
Input #0, alsa, from 'hw':
 Duration: N/A, start: 1738663653.066577, bitrate: 1536 kb/s
 Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, rtp, to 'rtp://192.168.1.152:5003':
 Metadata:
 encoder : Lavf59.27.100
 Stream #0:0: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s
 Metadata:
 encoder : Lavc59.37.100 pcm_s16le
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 192.168.1.152
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 5003 RTP/AVP 10
b=AS:768

[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201160; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201155; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201149; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201142; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201134; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201124; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201114; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201102; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201089; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201075; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201060; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201044; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201027; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 201009; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 200990; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 200970; changing to 201165. This may result in incorrect timestamps in the output file.
[rtp @ 0x558b48ea90] Non-monotonous DTS in output stream 0:0; previous: 201165, current: 200949; changing to 201165. This may result in incorrect timestamps in the output file.


I tried to add the +igndts flag but it is not changing anything. Also tried +genpts flag. I expected the DTS to restore itself but I still have the same issue


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