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Sur d’autres sites (7207)
-
How to record frames with ffmpeg and finish the recording
20 février 2024, par Jorge Augusto WilchenIn the following code, i trying to create a class to record frames from an IP camera (RTSP), save frames on a .avi file and finish the record, but, when i kill the operation, the video file may be corrupted. Have any other more safely way to stop the ffmpeg recording ?


.cpp file :


#include "videorecorder.h"


VideoRecorder::VideoRecorder(const std::string& rtspUrl) :
 url(rtspUrl),
 recording(false)
{

}

VideoRecorder::~VideoRecorder()
{
 end_record();
}

bool VideoRecorder::start_record(const std::string &fileName)
{
 if (recording) {
 std::cerr << "Already recording." << std::endl;
 return false;
 }

 std::string command = "ffmpeg -rtsp_transport udp -i " + url
 + " -c:v mjpeg -preset fast -qp 0 " + fileName;

 videoWriter = popen(command.c_str(), "w");
 if (!videoWriter) {
 std::cerr << "Error opening ffmpeg process." << std::endl;
 return false;
 }

 recording = true;
 ffmpegProcessId = getpid();
 std::cout << "Recording started." << std::endl;
 return true;
}

bool VideoRecorder::end_record()
{
 if (recording) {
 if (videoWriter) {
 pid_t ffmpegPID = fileno(videoWriter);

 if (kill(ffmpegPID, SIGTERM) == 0) {
 std::cout << "Recording terminated successfully." << std::endl;
 } else {
 std::cerr << "Error terminating recording." << std::endl;
 return false;
 }

 int status = pclose(videoWriter);

 if (status == 0) {
 std::cout << "Recording ended successfully." << std::endl;
 } else {
 std::cerr << "Error ending recording. pclose status: " << status << std::endl;
 return false;
 }
 } else {
 std::cerr << "Error ending recording. videoWriter is nullptr." << std::endl;
 return false;
 }

 recording = false;
 return true;
 }

 return false;
}


.h file :


#ifndef VIDEORECORDER_H
#define VIDEORECORDER_H

#include <string>
#include <iostream>
#include <fstream>
#include <cstdlib>
#include <csignal>
#include <sys></sys>wait.h>

extern "C" {
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libavutil></libavutil>avutil.h>
#include <libavutil></libavutil>opt.h>
#include <libswscale></libswscale>swscale.h>
#include 
}

#include <linux></linux>videodev2.h>

#include <opencv2></opencv2>opencv.hpp>
#include <opencv2></opencv2>videoio.hpp>
#include <opencv2></opencv2>highgui/highgui.hpp>


class VideoRecorder
{
public:
 VideoRecorder(const std::string& rtspUrl);
 ~VideoRecorder();
 bool start_record(const std::string& fileName);
 bool end_record();

private:
 std::string url;
 AVFormatContext *formatContext;
 AVStream *videoStream;
 AVCodecContext *codecContext;
 AVCodec *codec;
 SwsContext *swsContext;
 AVFrame *frame;
 AVPacket packet;
 bool recording;
 pid_t ffmpegProcessId;
 FILE* videoWriter;
};

#endif // VIDEORECORDER_H
</csignal></cstdlib></fstream></iostream></string>

I'm using the ffmpeg lib becouse i need max speed on frames recording, and OpenCV and AV Lib is much slowness than ffmpeg.


This my terminal output after recording during 10 seconds (generated a file with 23 seconds duration) :


Recording started.
ffmpeg version 4.3.6-0+deb11u1+rpt5 Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 10 (Debian 10.2.1-6)
 configuration: --prefix=/usr --extra-version=0+deb11u1+rpt5 --toolchain=hardened --incdir=/usr/include/aarch64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --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-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --disable-mmal --enable-neon --enable-v4l2-request --enable-libudev --enable-epoxy --enable-sand --libdir=/usr/lib/aarch64-linux-gnu --arch=arm64 --enable-pocketsphinx --enable-libdc1394 --enable-libdrm --enable-vout-drm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
 libavutil 56. 51.100 / 56. 51.100
 libavcodec 58. 91.100 / 58. 91.100
 libavformat 58. 45.100 / 58. 45.100
 libavdevice 58. 10.100 / 58. 10.100
 libavfilter 7. 85.100 / 7. 85.100
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 7.100 / 5. 7.100
 libswresample 3. 7.100 / 3. 7.100
 libpostproc 55. 7.100 / 55. 7.100
Input #0, rtsp, from 'rtsp://admin:[password]@[ip]:[port]/live/0/MAIN':
 Metadata:
 title : RTSP Server
 Duration: N/A, start: 0.280000, bitrate: N/A
 Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080, 25 fps, 25 tbr, 90k tbn, 50 tbc
Codec AVOption preset (Configuration preset) specified for output file #0 (/home/guardian-tech/Pictures/output_frame.avi) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
Codec AVOption qp (Constant quantization parameter rate control method) specified for output file #0 (/home/guardian-tech/Pictures/output_frame.avi) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> mjpeg (native))
Press [q] to stop, [?] for help
Output #0, avi, to '/home/guardian-tech/Pictures/output_frame.avi':
 Metadata:
 INAM : RTSP Server
 ISFT : Lavf58.45.100
 Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj420p(pc), 1920x1080, q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc
 Metadata:
 encoder : Lavc58.91.100 mjpeg
 Side data:
 cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: N/A
[rtsp @ 0x5592e7bb00] max delay reached. need to consume packet
[rtsp @ 0x5592e7bb00] RTP: missed 212 packets
[h264 @ 0x5592ebb790] concealing 2192 DC, 2192 AC, 2192 MV errors in I frame
rtsp://admin:[password]@[ip]:[port]/live/0/MAIN: corrupt decoded frame in stream 0
[rtsp @ 0x5592e7bb00] max delay reached. need to consume packet
[rtsp @ 0x5592e7bb00] RTP: missed 6 packets
[rtsp @ 0x5592e7bb00] max delay reached. need to consume packet
[rtsp @ 0x5592e7bb00] RTP: missed 14 packets
[h264 @ 0x5592f1bd30] cabac decode of qscale diff failed at 42 29
[h264 @ 0x5592f1bd30] error while decoding MB 42 29, bytestream 0
[h264 @ 0x5592f1bd30] concealing 4687 DC, 4687 AC, 4687 MV errors in I frame
rtsp://admin:[password]@[ip]:[port]/live/0/MAIN: corrupt decoded frame in stream 0
Error terminating recording.


-
VLC Player shows broken HLS stream with 4k HDR10 mkv
8 avril 2023, par goodkid38I am trying to convert a 4k mkv to an HLS stream but I am not having any luck. I have tried a few ffmpeg commands to try and fix the issue but none have worked. Here are the commands I have tried.


- 

- Basic copy command :




ffmpeg -i "video.mkv" -c copy -f hls "plexTemp/out.m3u8"

- 

- Command used to see if it was an HDR color issue :




ffmpeg -i "video.mkv" -c copy -pix_fmt yuv420p10le -f hls "plexTemp/out.m3u8"
3. Command used to revert to 8 bit color :

ffmpeg -i "video.mkv" -c copy -pix_fmt yuv420p -f hls "plexTemp/out.m3u8"

- 

- I tried removing extra streams and just focusing on audio and video :




ffmpeg -i "out.mkv" -map 0:v:0 -map 0:a:1 -c copy -pix_fmt yuv420p10le -f hls "plexTemp/out.m3u8"

I also saw these warnings when running each command.
Stream HEVC is not hvc1, you should use tag:v hvc1 to set it.
And this

[matroska,webm @ 000001d7921803c0] Stream #12: not enough frames to estimate rate; consider increasing probesize
[matroska,webm @ 000001d7921803c0] Could not find codec parameters for stream 6 (Subtitle: hdmv_pgs_subtitle (pgssub)): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
[matroska,webm @ 000001d7921803c0] Could not find codec parameters for stream 7 (Subtitle: hdmv_pgs_subtitle (pgssub)): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
[matroska,webm @ 000001d7921803c0] Could not find codec parameters for stream 8 (Subtitle: hdmv_pgs_subtitle (pgssub)): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
[matroska,webm @ 000001d7921803c0] Could not find codec parameters for stream 9 (Subtitle: hdmv_pgs_subtitle (pgssub)): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
[matroska,webm @ 000001d7921803c0] Could not find codec parameters for stream 10 (Subtitle: hdmv_pgs_subtitle (pgssub)): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
[matroska,webm @ 000001d7921803c0] Could not find codec parameters for stream 11 (Subtitle: hdmv_pgs_subtitle (pgssub)): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options


So I tried increasing the
analyzedurationandprobesizeand adding the tag like so :
ffmpeg -analyzeduration 10000000 -probesize 10000000 -i "out.mkv" -c copy -tag:v hvc1 -f hls "plexTemp/out.m3u8"

To no avail. Here is what the output looks like on VLC. It's mostly black with a few lines of color that randomly change.


On my TV I see this :




One thing I see that stands out in ffmpegs output is this :


[hls @ 00000207239a9ec0] Opening 'plexTemp/out0.ts' for writing7 bitrate= -0.0kbits/s speed=N/A
[hls @ 00000207239a9ec0] Opening 'plexTemp/out.m3u8.tmp' for writing


Here is the audio and video info on the mkv :


General
Unique ID : 92280908398971492516286250889389584022 (0x456CA80EF29B1357B572719D6EC4AE96)
Complete name : I:\video.mkv
Format : Matroska
Format version : Version 2
File size : 49.4 GiB
Duration : 1 h 39 min
Overall bit rate mode : Variable
Overall bit rate : 71.2 Mb/s
Frame rate : 23.976 FPS
Movie name : video
Encoded date : 2023-04-06 22:39:53 UTC
Writing application : MakeMKV v1.16.7 win(x64-release)
Writing library : libmakemkv v1.16.7 (1.3.10/1.5.2) win(x64-release)
Cover : Yes
Attachments : cover.jpg

Video
ID : 1
ID in the original source medium : 4113 (0x1011)
Format : HEVC
Format/Info : High Efficiency Video Coding
Format profile : Main 10@L5.1@High
HDR format : SMPTE ST 2086, HDR10 compatible
Codec ID : V_MPEGH/ISO/HEVC
Duration : 1 h 39 min
Bit rate : 63.6 Mb/s
Width : 3 840 pixels
Height : 2 160 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 23.976 (24000/1001) FPS
Color space : YUV
Chroma subsampling : 4:2:0 (Type 2)
Bit depth : 10 bits
Bits/(Pixel*Frame) : 0.320
Stream size : 44.1 GiB (89%)
Writing library : ATEME Titan File 3.9.6 (4.9.6.2) 
Language : English
Default : No
Forced : No
Color range : Limited
Color primaries : BT.2020
Transfer characteristics : PQ
Matrix coefficients : BT.2020 non-constant
Mastering display color primaries : Display P3
Mastering display luminance : min: 0.0050 cd/m2, max: 1000 cd/m2
Maximum Content Light Level : 1000 cd/m2
Maximum Frame-Average Light Level : 140 cd/m2
Original source medium : Blu-ray

Audio #1
ID : 2
ID in the original source medium : 4352 (0x1100)
Format : DTS XLL X
Format/Info : Digital Theater Systems
Commercial name : DTS:X
Codec ID : A_DTS
Duration : 1 h 39 min
Bit rate mode : Variable
Bit rate : 4 174 kb/s
Channel(s) : 8 channels
Channel layout : C L R LFE Lb Rb Lss Rss
Sampling rate : 48.0 kHz
Frame rate : 93.750 FPS (512 SPF)
Bit depth : 24 bits
Stream size : 2.89 GiB (6%)
Title : Surround 7.1
Language : English
Default : Yes
Forced : No
Original source medium : Blu-ray
Here is information on the HLS output:
General
Complete name : I:\out.m3u8
Format : HLS
Format profile : Media
File size : 67.4 MiB
Duration : 8 s 138 ms
Overall bit rate mode : Variable
Overall bit rate : 69.4 Mb/s
Frame rate : 23.976 FPS


And my
Here is the output of my HLS stream :


Video
ID : 256 (0x100)
Menu ID : 1 (0x1)
Format : HEVC
Format/Info : High Efficiency Video Coding
Format profile : Main 10@L5.1@High
HDR format : SMPTE ST 2086, HDR10 compatible
Muxing mode : MPEG-TS
Codec ID : 36
Duration : 8 s 49 ms
Width : 3 840 pixels
Height : 2 160 pixels
Display aspect ratio : 16:9
Frame rate : 23.976 (24000/1001) FPS
Color space : YUV
Chroma subsampling : 4:2:0 (Type 2)
Bit depth : 10 bits
Writing library : ATEME Titan File 3.9.6 (4.9.6.2) 
Color range : Limited
Color primaries : BT.2020
Transfer characteristics : PQ
Matrix coefficients : BT.2020 non-constant
Mastering display color primaries : Display P3
Mastering display luminance : min: 0.0050 cd/m2, max: 1000 cd/m2
Maximum Content Light Level : 1000 cd/m2
Maximum Frame-Average Light Level : 140 cd/m2
Source : out92.ts

Audio
ID : 257 (0x101)
Menu ID : 1 (0x1)
Format : DTS XLL X
Format/Info : Digital Theater Systems
Commercial name : DTS:X
Muxing mode : MPEG-TS
Codec ID : 130
Duration : 8 s 138 ms
Bit rate mode : Variable
Channel(s) : 8 channels
Channel layout : C L R LFE Lb Rb Lss Rss
Sampling rate : 48.0 kHz
Frame rate : 93.750 FPS (512 SPF)
Bit depth : 24 bits
Delay relative to video : -125 ms
Language : English
Source : out92.ts


-
ffmpeg failed to load audio file
14 avril 2024, par Vaishnav GhengeFailed to load audio: ffmpeg version 5.1.4-0+deb12u1 Copyright (c) Failed to load audio: ffmpeg version 5.1.4-0+deb12u1 Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 12 (Debian 12.2.0-14)
 configuration: --prefix=/usr --extra-version=0+deb12u1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --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-sdl2 --disable-sndio --enable-libjxl --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --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
/tmp/tmpjlchcpdm.wav: Invalid data found when processing input


backend :



@app.route("/transcribe", methods=["POST"])
def transcribe():
 # Check if audio file is present in the request
 if 'audio_file' not in request.files:
 return jsonify({"error": "No file part"}), 400
 
 audio_file = request.files.get('audio_file')

 # Check if audio_file is sent in files
 if not audio_file:
 return jsonify({"error": "`audio_file` is missing in request.files"}), 400

 # Check if the file is present
 if audio_file.filename == '':
 return jsonify({"error": "No selected file"}), 400

 # Save the file with a unique name
 filename = secure_filename(audio_file.filename)
 unique_filename = os.path.join("uploads", str(uuid.uuid4()) + '_' + filename)
 # audio_file.save(unique_filename)
 
 # Read the contents of the audio file
 contents = audio_file.read()

 max_file_size = 500 * 1024 * 1024
 if len(contents) > max_file_size:
 return jsonify({"error": "File is too large"}), 400

 # Check if the file extension suggests it's a WAV file
 if not filename.lower().endswith('.wav'):
 # Delete the file if it's not a WAV file
 os.remove(unique_filename)
 return jsonify({"error": "Only WAV files are supported"}), 400

 print(f"\033[92m{filename}\033[0m")

 # Call Celery task asynchronously
 result = transcribe_audio.delay(contents)

 return jsonify({
 "task_id": result.id,
 "status": "pending"
 })


@celery_app.task
def transcribe_audio(contents):
 # Transcribe the audio
 try:
 # Create a temporary file to save the audio data
 with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_audio:
 temp_path = temp_audio.name
 temp_audio.write(contents)

 print(f"\033[92mFile temporary path: {temp_path}\033[0m")
 transcribe_start_time = time.time()

 # Transcribe the audio
 transcription = transcribe_with_whisper(temp_path)
 
 transcribe_end_time = time.time()
 print(f"\033[92mTranscripted text: {transcription}\033[0m")

 return transcription, transcribe_end_time - transcribe_start_time

 except Exception as e:
 print(f"\033[92mError: {e}\033[0m")
 return str(e)


frontend :


useEffect(() => {
 const init = () => {
 navigator.mediaDevices.getUserMedia({audio: true})
 .then((audioStream) => {
 const recorder = new MediaRecorder(audioStream);

 recorder.ondataavailable = e => {
 if (e.data.size > 0) {
 setChunks(prevChunks => [...prevChunks, e.data]);
 }
 };

 recorder.onerror = (e) => {
 console.log("error: ", e);
 }

 recorder.onstart = () => {
 console.log("started");
 }

 recorder.start();

 setStream(audioStream);
 setRecorder(recorder);
 });
 }

 init();

 return () => {
 if (recorder && recorder.state === 'recording') {
 recorder.stop();
 }

 if (stream) {
 stream.getTracks().forEach(track => track.stop());
 }
 }
 }, []);

 useEffect(() => {
 // Send chunks of audio data to the backend at regular intervals
 const intervalId = setInterval(() => {
 if (recorder && recorder.state === 'recording') {
 recorder.requestData(); // Trigger data available event
 }
 }, 8000); // Adjust the interval as needed


 return () => {
 if (intervalId) {
 console.log("Interval cleared");
 clearInterval(intervalId);
 }
 };
 }, [recorder]);

 useEffect(() => {
 const processAudio = async () => {
 if (chunks.length > 0) {
 // Send the latest chunk to the server for transcription
 const latestChunk = chunks[chunks.length - 1];

 const audioBlob = new Blob([latestChunk]);
 convertBlobToAudioFile(audioBlob);
 }
 };

 void processAudio();
 }, [chunks]);

 const convertBlobToAudioFile = useCallback((blob: Blob) => {
 // Convert Blob to audio file (e.g., WAV)
 // This conversion may require using a third-party library or service
 // For example, you can use the MediaRecorder API to record audio in WAV format directly
 // Alternatively, you can use a library like recorderjs to perform the conversion
 // Here's a simplified example using recorderjs:

 const reader = new FileReader();
 reader.onload = () => {
 const audioBuffer = reader.result; // ArrayBuffer containing audio data

 // Send audioBuffer to Flask server or perform further processing
 sendAudioToFlask(audioBuffer as ArrayBuffer);
 };

 reader.readAsArrayBuffer(blob);
 }, []);

 const sendAudioToFlask = useCallback((audioBuffer: ArrayBuffer) => {
 const formData = new FormData();
 formData.append('audio_file', new Blob([audioBuffer]), `speech_audio.wav`);

 console.log(formData.get("audio_file"));

 fetch('http://34.87.75.138:8000/transcribe', {
 method: 'POST',
 body: formData
 })
 .then(response => response.json())
 .then((data: { task_id: string, status: string }) => {
 pendingTaskIdsRef.current.push(data.task_id);
 })
 .catch(error => {
 console.error('Error sending audio to Flask server:', error);
 });
 }, []);


I was trying to pass the audio from frontend to whisper model which is in flask app


Syndiquer tout le site
