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Sur d’autres sites (11059)
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How to record the bluetooth audio from my phone in my pc using ffmpeg ?
12 septembre 2023, par Juan InzunzaI have my phone connected via Bluetooth to my PC and playing audio in my phone so i can hear that audio in my pc and i was trying to record that audio in my PC using ffmpeg this is the command i got so far


ffmpeg -f pulse -ac 1 -ar 44100 -i alsa_input.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.mono-fallback -f pulse -ac 2 -ar 44100 -i bluez_source.84_5F_04_75_B9_F6.a2dp_source -filter_complex amix=inputs=2 -i /dev/video5 -s 640x480 -vcodec libx264 -preset veryfast -crf 18 -acodec libmp3lame -ar 44100 -q:a 1 -pix_fmt yuv420p -aq 0 ~/Videos


"bluez_source.84_5F_04_75_B9_F6.a2dp_source" is the name of the bluetooth audio source (my phone) but the problem is, if i am not playing audio on the device, just disappears when i execute this command "pactl list sources", and appears again when i play audio, so is listed when i run "pactl list sources" and i have to keep the phone playing audio so ffmpeg recognizes that device and not fail when executed.


Finally my question is how can i keep that device in running state or suspended state without need to play audio all the time.


gandalf@Mordor:~$ pactl list sources
Source #64
 State: IDLE
 Name: alsa_output.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.analog-stereo.monitor
 Description: Monitor of KT USB Audio Analog Stereo
 Driver: module-alsa-card.c
 Sample Specification: s16le 2ch 44100Hz
 Channel Map: front-left,front-right
 Owner Module: 85
 Mute: no
 Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
 balance 0.00
 Base Volume: 65536 / 100% / 0.00 dB
 Monitor of Sink: alsa_output.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.analog-stereo
 Latency: 0 usec, configured 2000000 usec
 Flags: DECIBEL_VOLUME LATENCY 
 Properties:
 device.description = "Monitor of KT USB Audio Analog Stereo"
 device.class = "monitor"
 alsa.card = "1"
 alsa.card_name = "KT USB Audio"
 alsa.long_card_name = "KTMicro KT USB Audio at usb-0000:03:00.3-3, full speed"
 alsa.driver_name = "snd_usb_audio"
 device.bus_path = "pci-0000:03:00.3-usb-0:3:1.0"
 sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:03:00.3/usb1/1-3/1-3:1.0/sound/card1"
 udev.id = "usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00"
 device.bus = "usb"
 device.vendor.id = "12d1"
 device.vendor.name = "Huawei Technologies Co., Ltd."
 device.product.id = "0010"
 device.product.name = "KT USB Audio"
 device.serial = "KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000-"
 device.string = "1"
 module-udev-detect.discovered = "1"
 device.icon_name = "audio-card-usb"
 Formats:
 pcm

Source #65
 State: SUSPENDED
 Name: alsa_input.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.mono-fallback
 Description: KT USB Audio Mono
 Driver: module-alsa-card.c
 Sample Specification: s16le 1ch 44100Hz
 Channel Map: mono
 Owner Module: 85
 Mute: no
 Volume: mono: 65536 / 100% / 0.00 dB
 balance 0.00
 Base Volume: 65536 / 100% / 0.00 dB
 Monitor of Sink: n/a
 Latency: 0 usec, configured 0 usec
 Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY 
 Properties:
 alsa.resolution_bits = "16"
 device.api = "alsa"
 device.class = "sound"
 alsa.class = "generic"
 alsa.subclass = "generic-mix"
 alsa.name = "USB Audio"
 alsa.id = "USB Audio"
 alsa.subdevice = "0"
 alsa.subdevice_name = "subdevice #0"
 alsa.device = "0"
 alsa.card = "1"
 alsa.card_name = "KT USB Audio"
 alsa.long_card_name = "KTMicro KT USB Audio at usb-0000:03:00.3-3, full speed"
 alsa.driver_name = "snd_usb_audio"
 device.bus_path = "pci-0000:03:00.3-usb-0:3:1.0"
 sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:03:00.3/usb1/1-3/1-3:1.0/sound/card1"
 udev.id = "usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00"
 device.bus = "usb"
 device.vendor.id = "12d1"
 device.vendor.name = "Huawei Technologies Co., Ltd."
 device.product.id = "0010"
 device.product.name = "KT USB Audio"
 device.serial = "KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000-"
 device.string = "hw:1"
 device.buffering.buffer_size = "176400"
 device.buffering.fragment_size = "88200"
 device.access_mode = "mmap+timer"
 device.profile.name = "mono-fallback"
 device.profile.description = "Mono"
 device.description = "KT USB Audio Mono"
 module-udev-detect.discovered = "1"
 device.icon_name = "audio-card-usb"
 Ports:
 analog-input-mic: Microphone (type: Mic, priority: 8700, availability unknown)
 Active Port: analog-input-mic
 Formats:
 pcm

Source #68
 State: RUNNING
 Name: bluez_source.84_5F_04_75_B9_F6.a2dp_source
 Description: Juan's S20 FE
 Driver: module-bluez5-device.c
 Sample Specification: s16le 2ch 44100Hz
 Channel Map: front-left,front-right
 Owner Module: 23
 Mute: no
 Volume: front-left: 53151 / 81% / -5.46 dB, front-right: 53151 / 81% / -5.46 dB
 balance 0.00
 Base Volume: 65536 / 100% / 0.00 dB
 Monitor of Sink: n/a
 Latency: 68928 usec, configured 68537 usec
 Flags: HARDWARE DECIBEL_VOLUME LATENCY 
 Properties:
 bluetooth.protocol = "a2dp_source"
 bluetooth.codec = "sbc"
 device.description = "Juan's S20 FE"
 device.string = "84:5F:04:75:B9:F6"
 device.api = "bluez"
 device.class = "sound"
 device.bus = "bluetooth"
 device.form_factor = "phone"
 bluez.path = "/org/bluez/hci0/dev_84_5F_04_75_B9_F6"
 bluez.class = "0x5a020c"
 bluez.alias = "Juan's S20 FE"
 device.icon_name = "audio-card-bluetooth"
 Ports:
 phone-input: Phone (type: Phone, priority: 0, available)
 Active Port: phone-input
 Formats:
 pcm


This is the error i am getting when there's no playing audio in my phone.


ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
 built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
 configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --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-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --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-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 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
 libavutil 56. 70.100 / 56. 70.100
 libavcodec 58.134.100 / 58.134.100
 libavformat 58. 76.100 / 58. 76.100
 libavdevice 58. 13.100 / 58. 13.100
 libavfilter 7.110.100 / 7.110.100
 libswscale 5. 9.100 / 5. 9.100
 libswresample 3. 9.100 / 3. 9.100
 libpostproc 55. 9.100 / 55. 9.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, pulse, from 'alsa_input.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.mono-fallback':
 Duration: N/A, start: 1694557964.268337, bitrate: 705 kb/s
 Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
bluez_source.84_5F_04_75_B9_F6.a2dp_source: Input/output error



-
using ffmpeg libraray to write to a mp4, ffprobe shows there are 100 frames and 100 packets, but av_interleaved_write_frame only called 50 times
2 mai 2023, par ollydbg23here is my code to generate a mp4 file by using ffmpeg and opencv library. The opencv library is only try to generate 100 images(frames), and ffmpeg library is to compress the images to a mp4 files.


Here is the working code :


#include <iostream>
#include <vector>
#include <cstring>
#include <fstream>
#include <sstream>
#include <stdexcept>
#include <opencv2></opencv2>opencv.hpp>
extern "C" {
#include <libavutil></libavutil>imgutils.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libavutil></libavutil>opt.h>
}

#include<cstdlib> // to generate time stamps

using namespace std;
using namespace cv;

int main()
{
 // Set up input frames as BGR byte arrays
 vector<mat> frames;

 int width = 640;
 int height = 480;
 int num_frames = 100;
 Scalar black(0, 0, 0);
 Scalar white(255, 255, 255);
 int font = FONT_HERSHEY_SIMPLEX;
 double font_scale = 1.0;
 int thickness = 2;

 for (int i = 0; i < num_frames; i++) {
 Mat frame = Mat::zeros(height, width, CV_8UC3);
 putText(frame, std::to_string(i), Point(width / 2 - 50, height / 2), font, font_scale, white, thickness);
 frames.push_back(frame);
 }

 // generate a serial of time stamps which is used to set the PTS value
 // suppose they are in ms unit, the time interval is between 30ms to 59ms
 vector<int> timestamps;

 for (int i = 0; i < num_frames; i++) {
 int timestamp;
 if (i == 0)
 timestamp = 0;
 else
 {
 int random = 30 + (rand() % 30);
 timestamp = timestamps[i-0] + random;
 }

 timestamps.push_back(timestamp);
 }

 // Populate frames with BGR byte arrays

 // Initialize FFmpeg
 //av_register_all();

 // Set up output file
 AVFormatContext* outFormatCtx = nullptr;
 //AVCodec* outCodec = nullptr;
 AVCodecContext* outCodecCtx = nullptr;
 //AVStream* outStream = nullptr;
 //AVPacket outPacket;

 const char* outFile = "output.mp4";
 int outWidth = frames[0].cols;
 int outHeight = frames[0].rows;
 int fps = 25;

 // Open the output file context
 avformat_alloc_output_context2(&outFormatCtx, nullptr, nullptr, outFile);
 if (!outFormatCtx) {
 cerr << "Error: Could not allocate output format context" << endl;
 return -1;
 }

 // Open the output file
 if (avio_open(&outFormatCtx->pb, outFile, AVIO_FLAG_WRITE) < 0) {
 cerr << "Error opening output file" << std::endl;
 return -1;
 }

 // Set up output codec
 const AVCodec* outCodec = avcodec_find_encoder(AV_CODEC_ID_H264);
 if (!outCodec) {
 cerr << "Error: Could not find H.264 codec" << endl;
 return -1;
 }

 outCodecCtx = avcodec_alloc_context3(outCodec);
 if (!outCodecCtx) {
 cerr << "Error: Could not allocate output codec context" << endl;
 return -1;
 }
 outCodecCtx->codec_id = AV_CODEC_ID_H264;
 outCodecCtx->codec_type = AVMEDIA_TYPE_VIDEO;
 outCodecCtx->pix_fmt = AV_PIX_FMT_YUV420P;
 outCodecCtx->width = outWidth;
 outCodecCtx->height = outHeight;
 //outCodecCtx->time_base = { 1, fps*1000 }; // 25000
 outCodecCtx->time_base = { 1, fps}; // 25000
 outCodecCtx->framerate = {fps, 1}; // 25
 outCodecCtx->bit_rate = 4000000;

 //https://github.com/leandromoreira/ffmpeg-libav-tutorial
 //We set the flag AV_CODEC_FLAG_GLOBAL_HEADER which tells the encoder that it can use the global headers.
 if (outFormatCtx->oformat->flags & AVFMT_GLOBALHEADER)
 {
 outCodecCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; //
 }

 // Open output codec
 if (avcodec_open2(outCodecCtx, outCodec, nullptr) < 0) {
 cerr << "Error: Could not open output codec" << endl;
 return -1;
 }

 // Create output stream
 AVStream* outStream = avformat_new_stream(outFormatCtx, outCodec);
 if (!outStream) {
 cerr << "Error: Could not allocate output stream" << endl;
 return -1;
 }

 // Configure output stream parameters (e.g., time base, codec parameters, etc.)
 // ...

 // Connect output stream to format context
 outStream->codecpar->codec_id = outCodecCtx->codec_id;
 outStream->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
 outStream->codecpar->width = outCodecCtx->width;
 outStream->codecpar->height = outCodecCtx->height;
 outStream->codecpar->format = outCodecCtx->pix_fmt;
 outStream->time_base = outCodecCtx->time_base;

 int ret = avcodec_parameters_from_context(outStream->codecpar, outCodecCtx);
 if (ret < 0) {
 cerr << "Error: Could not copy codec parameters to output stream" << endl;
 return -1;
 }

 outStream->avg_frame_rate = outCodecCtx->framerate;
 //outStream->id = outFormatCtx->nb_streams++; <--- We shouldn't modify outStream->id

 ret = avformat_write_header(outFormatCtx, nullptr);
 if (ret < 0) {
 cerr << "Error: Could not write output header" << endl;
 return -1;
 }

 // Convert frames to YUV format and write to output file
 int frame_count = -1;
 for (const auto& frame : frames) {
 frame_count++;
 AVFrame* yuvFrame = av_frame_alloc();
 if (!yuvFrame) {
 cerr << "Error: Could not allocate YUV frame" << endl;
 return -1;
 }
 av_image_alloc(yuvFrame->data, yuvFrame->linesize, outWidth, outHeight, AV_PIX_FMT_YUV420P, 32);

 yuvFrame->width = outWidth;
 yuvFrame->height = outHeight;
 yuvFrame->format = AV_PIX_FMT_YUV420P;

 // Convert BGR frame to YUV format
 Mat yuvMat;
 cvtColor(frame, yuvMat, COLOR_BGR2YUV_I420);
 memcpy(yuvFrame->data[0], yuvMat.data, outWidth * outHeight);
 memcpy(yuvFrame->data[1], yuvMat.data + outWidth * outHeight, outWidth * outHeight / 4);
 memcpy(yuvFrame->data[2], yuvMat.data + outWidth * outHeight * 5 / 4, outWidth * outHeight / 4);

 // Set up output packet
 //av_init_packet(&outPacket); //error C4996: 'av_init_packet': was declared deprecated
 AVPacket* outPacket = av_packet_alloc();
 memset(outPacket, 0, sizeof(outPacket)); //Use memset instead of av_init_packet (probably unnecessary).
 //outPacket->data = nullptr;
 //outPacket->size = 0;

 // set the frame pts, do I have to set the package pts?

 // yuvFrame->pts = av_rescale_q(timestamps[frame_count]*25, outCodecCtx->time_base, outStream->time_base); //Set PTS timestamp
 yuvFrame->pts = av_rescale_q(frame_count*frame_count, outCodecCtx->time_base, outStream->time_base); //Set PTS timestamp

 // Encode frame and write to output file
 int ret = avcodec_send_frame(outCodecCtx, yuvFrame);
 if (ret < 0) {
 cerr << "Error: Could not send frame to output codec" << endl;
 return -1;
 }
 while (ret >= 0)
 {
 ret = avcodec_receive_packet(outCodecCtx, outPacket);

 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
 {
 int abc;
 abc++;
 break;
 }
 else if (ret < 0)
 {
 cerr << "Error: Could not receive packet from output codec" << endl;
 return -1;
 }

 //av_packet_rescale_ts(&outPacket, outCodecCtx->time_base, outStream->time_base);

 outPacket->stream_index = outStream->index;

 outPacket->duration = av_rescale_q(1, outCodecCtx->time_base, outStream->time_base); // Set packet duration

 ret = av_interleaved_write_frame(outFormatCtx, outPacket);

 static int call_write = 0;

 call_write++;
 printf("av_interleaved_write_frame %d\n", call_write);

 av_packet_unref(outPacket);
 if (ret < 0) {
 cerr << "Error: Could not write packet to output file" << endl;
 return -1;
 }
 }

 av_frame_free(&yuvFrame);
 }

 // Flush the encoder
 ret = avcodec_send_frame(outCodecCtx, nullptr);
 if (ret < 0) {
 std::cerr << "Error flushing encoder: " << std::endl;
 return -1;
 }

 while (ret >= 0) {
 AVPacket* pkt = av_packet_alloc();
 if (!pkt) {
 std::cerr << "Error allocating packet" << std::endl;
 return -1;
 }
 ret = avcodec_receive_packet(outCodecCtx, pkt);

 // Write the packet to the output file
 if (ret == 0)
 {
 pkt->stream_index = outStream->index;
 pkt->duration = av_rescale_q(1, outCodecCtx->time_base, outStream->time_base); // <---- Set packet duration
 ret = av_interleaved_write_frame(outFormatCtx, pkt);
 av_packet_unref(pkt);
 if (ret < 0) {
 std::cerr << "Error writing packet to output file: " << std::endl;
 return -1;
 }
 }
 }


 // Write output trailer
 av_write_trailer(outFormatCtx);

 // Clean up
 avcodec_close(outCodecCtx);
 avcodec_free_context(&outCodecCtx);
 avformat_free_context(outFormatCtx);

 return 0;
}

</int></mat></cstdlib></stdexcept></sstream></fstream></cstring></vector></iostream>

Note that I have used the
ffprobetool(one of the tool from ffmpeg) to inspect the generated mp4 files.

I see that the mp4 file has 100 frames and 100 packets, but in my code, I have such lines :


static int call_write = 0;

 call_write++;
 printf("av_interleaved_write_frame %d\n", call_write);


I just see that the
av_interleaved_write_framefunction is only called 50 times, not the expected 100 times, anyone can explain it ?

Thanks.


BTW, from the ffmpeg document( see here : For video, it should typically contain one compressed frame ), I see that a
packetmainly has one videoframe, so theffprobe's result looks correct.

Here is the command I used to inspect the mp4 file :


ffprobe -show_frames output.mp4 >> frames.txt
ffprobe -show_packets output.mp4 >> packets.txt


My testing code is derived from an answer in another question here : avformat_write_header() function call crashed when I try to save several RGB data to a output.mp4 file


-
How to fix laggy ffmpeg screen and audio capture ?
26 juillet 2022, par Wh0r00tI am using
ffmpegto capture the screen along with audio.

The
ffmpegcommand that i tried is

ffmpeg -y \
 -f x11grab \
 -framerate 60 \
 -s 1366x768 \
 -i :0.0 \
 -f alsa -i default -ac 2 \
 -r 30 \
 -c:v h264 -crf 0 -preset ultrafast -c:a vorbis -strict experimental \
 "$HOME/Videos/$fname-$(date '+%y%m%d-%H%M-%S').mkv"


The stdout of the
ffmpeghttps://pastebin.com/Qmi5TMKv

ffmpeg version n5.0.1 Copyright (c) 2000-2022 the FFmpeg developers
 built with gcc 12.1.0 (GCC)
 configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-shared --enable-version3
 libavutil 57. 17.100 / 57. 17.100
 libavcodec 59. 18.100 / 59. 18.100
 libavformat 59. 16.100 / 59. 16.100
 libavdevice 59. 4.100 / 59. 4.100
 libavfilter 8. 24.100 / 8. 24.100
 libswscale 6. 4.100 / 6. 4.100
 libswresample 4. 3.100 / 4. 3.100
 libpostproc 56. 3.100 / 56. 3.100
[x11grab @ 0x561faf77eb00] Stream #0: not enough frames to estimate rate; consider increasing probesize
Input #0, x11grab, from ':0.0':
 Duration: N/A, start: 1658814267.169414, bitrate: 2014248 kb/s
 Stream #0:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1366x768, 2014248 kb/s, 60 fps, 1000k tbr, 1000k tbn
Guessed Channel Layout for Input Stream #1.0 : stereo
Input #1, alsa, from 'default':
 Duration: N/A, start: 1658814267.230653, bitrate: 1536 kb/s
 Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
 Stream #1:0 -> #0:1 (pcm_s16le (native) -> vorbis (native))
Press [q] to stop, [?] for help
[libx264 @ 0x561faf7d4300] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX XOP FMA3 BMI1
[libx264 @ 0x561faf7d4300] profile High 4:4:4 Predictive, level 3.2, 4:4:4, 8-bit
[libx264 @ 0x561faf7d4300] 264 - core 164 r3081 19856cc - H.264/MPEG-4 AVC codec - Copyleft 2003-2021 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=0 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=0 chroma_qp_offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=cqp mbtree=0 qp=0
[alsa @ 0x561faf78a940] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
Output #0, matroska, to '/home/earth/Videos/-220726-1114-27.mkv':
 Metadata:
 encoder : Lavf59.16.100
 Stream #0:0: Video: h264 (H264 / 0x34363248), yuv444p(tv, progressive), 1366x768, q=2-31, 30 fps, 1k tbn
 Metadata:
 encoder : Lavc59.18.100 libx264
 Side data:
 cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
 Stream #0:1: Audio: vorbis (oV[0][0] / 0x566F), 48000 Hz, stereo, fltp
 Metadata:
 encoder : Lavc59.18.100 vorbis
[vorbis @ 0x561faf7d5500] Queue input is backward in time0 bitrate=N/A speed= 0x
frame= 153 fps= 31 q=-1.0 Lsize= 2295kB time=00:00:05.06 bitrate=3709.5kbits/s dup=0 drop=150 speed=1.01x
video:2282kB audio:7kB subtitle:0kB other streams:0kB global headers:3kB muxing overhead: 0.281689%
[libx264 @ 0x561faf7d4300] frame I:1 Avg QP: 0.00 size:381729
[libx264 @ 0x561faf7d4300] frame P:152 Avg QP: 0.00 size: 12857
[libx264 @ 0x561faf7d4300] mb I I16..4: 100.0% 0.0% 0.0%
[libx264 @ 0x561faf7d4300] mb P I16..4: 56.3% 0.0% 0.0% P16..4: 0.1% 0.0% 0.0% 0.0% 0.0% skip:43.6%
[libx264 @ 0x561faf7d4300] coded y,u,v intra: 1.6% 1.6% 1.6% inter: 0.2% 0.2% 0.2%
[libx264 @ 0x561faf7d4300] i16 v,h,dc,p: 99% 1% 0% 0%
[libx264 @ 0x561faf7d4300] kb/s:3664.27
Exiting normally, received signal 15.


I am using the preset ultrafast because I read that it helps not to compress the video too much.
The output of the recorded test file using ffmpeg is as below.


(+) Video --vid=1 (h264 1366x768 30.000fps)
 (+) Audio --aid=1 (vorbis 2ch 48000Hz)
AO: [pulse] 48000Hz stereo 2ch float
VO: [gpu] 1366x768 yuv444p
AV: 00:00:03 / 00:00:19 (17%) A-V: 0.000
[mkv] Discarding potentially broken or useless index.
AV: 00:00:14 / 00:00:19 (73%) A-V: 0.000

Exiting... (Quit)


The recording works but there is a audio lag. If I record the same using
simplescreenrecorderwith the same settings like,

audio backend - alsa


source - default


audio codec - vorbis


video codec - h.264


container - matroska


preset - superfast


The
simplescreenrecorderlog https://pastebin.com/83hMMRQF

[PageRecord::StartPage] Starting page ...
[PageRecord::StartPage] Started page.
[PageRecord::StartOutput] Starting output ...
[PageRecord::StartOutput] Output file: /home/earth/Videos/simplescreenrecorder-2022-07-26_11.18.13.mkv
[Muxer::Init] Using format matroska (Matroska).
[Muxer::AddStream] Using codec libx264 (libx264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10).
[VideoEncoder::PrepareStream] Using pixel format nv12.
[libx264 @ 0x563436cbfd40] using SAR=1/1
[libx264 @ 0x563436cbfd40] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX XOP FMA3 BMI1
[libx264 @ 0x563436cbfd40] profile High, level 3.2, 4:2:0, 8-bit
[libx264 @ 0x563436cbfd40] 264 - core 164 r3081 19856cc - H.264/MPEG-4 AVC codec - Copyleft 2003-2021 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x3 me=dia subme=1 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=4 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 pb_ratio=1.30 aq=1:1.00
[Muxer::AddStream] Using codec libvorbis (libvorbis).
[BaseEncoder::EncoderThread] Encoder thread started.
[AudioEncoder::PrepareStream] Using sample format f32p.
[BaseEncoder::EncoderThread] Encoder thread started.
[Muxer::MuxerThread] Muxer thread started.
[PageRecord::StartOutput] Started output.
[Synchronizer::SynchronizerThread] Synchronizer thread started.
[PageRecord::StartInput] Starting input ...
[X11Input::Init] Using X11 shared memory.
[X11Input::Init] Detecting screen configuration ...
[X11Input::Init] Screen 0: x1 = 0, y1 = 0, x2 = 1366, y2 = 768
[X11Input::InputThread] Input thread started.
[ALSAInput::InputThread] Using sample format s16.
[PageRecord::StartInput] Started input.
[ALSAInput::InputThread] Input thread started.
[FastResampler::Resample] Resample ratio is 1.0000 (was 0.0000).
[PageRecord::StopOutput] Stopping output ...
[PageRecord::StopOutput] Stopped output.
[PageRecord::StopInput] Stopping input ...
[X11Input::~X11Input] Stopping input thread ...
[X11Input::InputThread] Input thread stopped.
[ALSAInput::~ALSAInput] Stopping input thread ...
[ALSAInput::InputThread] Input thread stopped.
[PageRecord::StopInput] Stopped input.


It works perfectly without any lag whatsoever. The output of the recorded test file using simplescreenrecorder is as below.


(+) Video --vid=1 (h264 1366x768)
 (+) Audio --aid=1 (vorbis 2ch 48000Hz)
AO: [pulse] 48000Hz stereo 2ch float
VO: [gpu] 1366x768 yuv420p
AV: 00:00:01 / 00:00:17 (7%) A-V: 0.000
[mkv] Discarding potentially broken or useless index.
AV: 00:00:08 / 00:00:17 (47%) A-V: 0.000

Exiting... (Quit)


The only difference that I saw between these two recordings is
VO: [gpu] 1366x768 yuv444p
VO: [gpu] 1366x768 yuv420pfor ffmpeg and simplescreenrecorder receptively.
I do not know if this matters but is there something that I could tweak to makeffmpegto capture the screen and audio without any lag.
Like answered here https://unix.stackexchange.com/questions/675436/ffmpeg-recording-slows-down-when-audio-inputs-are-added
I do open pavucontrol but its not much of a help.

The reason that I going with
ffmpegis because I can kill the process usingpidat a particular time using cronjobs.
These are my system information, in case if it helps

System:
 Host: taco Kernel: 5.18.12-arch1-1 arch: x86_64 bits: 64 Desktop: dwm
 v: 6.2 Distro: Arch Linux
Machine:
 Type: Desktop Mobo: Acer model: A75F2-M v: P21-A1 serial: N/A BIOS: Acer
 v: P21-A1 date: 02/07/2014
CPU:
 Info: quad core model: AMD A8-5500B APU with Radeon HD Graphics bits: 64
 type: MT MCP cache: L2: 4 MiB
 Speed (MHz): avg: 1400 min/max: 1400/3200 cores: 1: 1400 2: 1400 3: 1400
 4: 1400
Graphics:
 Device-1: AMD Trinity [Radeon HD 7560D] driver: radeon v: kernel
 Display: server: X.Org v: 21.1.4 driver: X: loaded: modesetting
 gpu: radeon resolution: 1366x768~60Hz
 OpenGL: renderer: AMD ARUBA (DRM 2.50.0 / 5.18.12-arch1-1 LLVM 14.0.6)
 v: 4.3 Mesa 22.1.3
Audio:
 Device-1: AMD FCH Azalia driver: snd_hda_intel
 Sound Server-1: ALSA v: k5.18.12-arch1-1 running: yes
 Sound Server-2: PulseAudio v: 16.1 running: yes
 Sound Server-3: PipeWire v: 0.3.56 running: yes


Any help is much appreciated.


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