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• How to record the bluetooth audio from my phone in my pc using ffmpeg ?

  12 septembre 2023, par Juan Inzunza

  I 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



  


• ffmpeg doesn't record RTSP stream

  7 août 2023, par zcoder

  I'm trying to record RTSP stream from Dahua IP camera which is located on 10.10.10.10 (for example purpose), on Windows it works well and on previous ubuntu version I had (18) it was also working well.
However on Ubuntu Server 22.04 it runs but ends without any error.

  


  My command :
ffmpeg -loglevel debug -i rtsp://admin:admin@10.10.10.10:554/cam/realmonitor?channel=1"&"subtype=1 -movflags +frag_keyframe+separate_moof+omit_tfhd_offset+empty_moov -acodec copy -vcodec copy out.mp4

  


  And debug output of the ffmpeg is :

  


  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
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-i' ... matched as input url with argument 'rtsp://admin:admin@10.10.10.10:554/cam/realmonitor?channel=1&subtype=1'.
Reading option '-movflags' ... matched as AVOption 'movflags' with argument '+frag_keyframe+separate_moof+omit_tfhd_offset+empty_moov'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'copy'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'copy'.
Reading option '123.mp4' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input url rtsp://admin:admin@10.10.10.10:554/cam/realmonitor?channel=1&subtype=1.
Successfully parsed a group of options.
Opening an input file: rtsp://admin:admin@10.10.10.10:554/cam/realmonitor?channel=1&subtype=1.
[tcp @ 0x55c812986800] No default whitelist set
[tcp @ 0x55c812986800] Original list of addresses:
[tcp @ 0x55c812986800] Address 10.10.10.10 port 554
[tcp @ 0x55c812986800] Interleaved list of addresses:
[tcp @ 0x55c812986800] Address 10.10.10.10 port 554
[tcp @ 0x55c812986800] Starting connection attempt to 10.10.10.10 port 554
[tcp @ 0x55c812986800] Successfully connected to 10.10.10.10 port 554
[rtsp @ 0x55c812983740] SDP:
v=0
o=- 2252669512 2252669512 IN IP4 0.0.0.0
s=Media Server
c=IN IP4 0.0.0.0
t=0 0
a=control:*
a=packetization-supported:DH
a=rtppayload-supported:DH
a=range:npt=now-
m=video 0 RTP/AVP 98
a=control:trackID=0
a=framerate:25.000000
a=rtpmap:98 H265/90000
a=fmtp:98 profile-id=1;sprop-sps=QgEBAUAAAAMAAAMAAAMAAAMAmaAFggCQf5a7kbBrlUE=;sprop-pps=RAHAc8BMkA==;sprop-vps=QAEMAf//AUAAAAMAAAMAAAMAAAMAmawJ
a=recvonly

[rtsp @ 0x55c812983740] video codec set to: hevc
[rtp @ 0x55c8129899c0] No default whitelist set
[udp @ 0x55c81298a480] No default whitelist set
[udp @ 0x55c81298a480] end receive buffer size reported is 425984
[udp @ 0x55c812989e00] No default whitelist set
[udp @ 0x55c812989e00] end receive buffer size reported is 425984
[rtsp @ 0x55c812983740] setting jitter buffer size to 500
[rtsp @ 0x55c812983740] hello state=0
Failed to parse interval end specification ''
[hevc @ 0x55c812989100] nal_unit_type: 32(VPS), nuh_layer_id: 0, temporal_id: 0
[hevc @ 0x55c812989100] nal_unit_type: 33(SPS), nuh_layer_id: 0, temporal_id: 0
[hevc @ 0x55c812989100] nal_unit_type: 34(PPS), nuh_layer_id: 0, temporal_id: 0
[hevc @ 0x55c812989100] Decoding VPS
[hevc @ 0x55c812989100] Main profile bitstream
[hevc @ 0x55c812989100] Decoding SPS
[hevc @ 0x55c812989100] Main profile bitstream
[hevc @ 0x55c812989100] Decoding PPS
[hevc @ 0x55c812989100] nal_unit_type: 32(VPS), nuh_layer_id: 0, temporal_id: 0
[hevc @ 0x55c812989100] nal_unit_type: 33(SPS), nuh_layer_id: 0, temporal_id: 0
[hevc @ 0x55c812989100] nal_unit_type: 34(PPS), nuh_layer_id: 0, temporal_id: 0
[hevc @ 0x55c812989100] Decoding VPS
[hevc @ 0x55c812989100] Main profile bitstream
[hevc @ 0x55c812989100] Decoding SPS
[hevc @ 0x55c812989100] Main profile bitstream
[hevc @ 0x55c812989100] Decoding PPS
Input #0, rtsp, from 'rtsp://admin:admin@10.10.10.10:554/cam/realmonitor?channel=1&subtype=1':
  Metadata:
    title           : Media Server
  Duration: N/A, bitrate: N/A
  Stream #0:0, 0, 1/90000: Video: hevc (Main), 1 reference frame, yuv420p(tv, left), 704x576, 0/1, 90k tbr, 90k tbn, 90k tbc
Successfully opened the file.
Parsing a group of options: output url out.mp4.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument copy.
Applying option vcodec (force video codec ('copy' to copy stream)) with argument copy.
Successfully parsed a group of options.
Opening an output file: out.mp4.
File '123.mp4' already exists. Overwrite? [y/N] y
[file @ 0x55c8129ba9c0] Setting default whitelist 'file,crypto,data'
Successfully opened the file.
[mp4 @ 0x55c8129bb800] Empty MOOV enabled; disabling automatic bitstream filtering
Output #0, mp4, to 'out.mp4':
  Metadata:
    title           : Media Server
    encoder         : Lavf58.76.100
  Stream #0:0, 0, 1/90000: Video: hevc (Main), 1 reference frame (hev1 / 0x31766568), yuv420p(tv, left), 704x576 (0x0), 0/1, q=2-31, 90k tbr, 90k tbn, 90k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
No more output streams to write to, finishing.e=00:00:00.00 bitrate=N/A speed=   0x
frame=    0 fps=0.0 q=-1.0 Lsize=       1kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Input file #0 (rtsp://admin:admin@10.10.10.10:554/cam/realmonitor?channel=1&subtype=1):
  Input stream #0:0 (video): 0 packets read (0 bytes);
  Total: 0 packets (0 bytes) demuxed
Output file #0 (123.mp4):
  Output stream #0:0 (video): 0 packets muxed (0 bytes);
  Total: 0 packets (0 bytes) muxed
0 frames successfully decoded, 0 decoding errors



  


  I would like to record stream to mp4 file. Any ideas ?

  


• record every step of selenium test case to a video proplem

  16 février 2017, par user7574253

  i use pytest and selenium to test web site, i am writing a base class, every test case run should be recording to a video, so i can see the fail test case in the video and know what’s happing. below is my code

  ffmpeg_command = os.environ['ffmpeg_command']
  
  video_base_path = os.environ['video_base_path']
  
  
  
  
  
  class BaseRecordTest(BaseTest):
  
      def set_up_00(self, method):
  
  
  
          # video save path should be "base path" + "test case module path" +      "test case class" + "test case name"
  
          module_name = self.__module__
  
          if module_name == '__main__':
  
               filename = sys.modules[self.__module__].__file__
  
               module_name = os.path.splitext(os.path.basename(filename))[0]
  
          module_path = module_name.replace('.', '\\')
  
          dir_name = video_base_path + module_path
  
          if not os.path.exists(dir_name):
  
              os.makedirs(dir_name)
  
  
  
          video_path = dir_name + '\{}-{}.avi'.format(self.__class__.__name__,   method.__name__)
  
  
  
          if os.path.exists(video_path):
  
              os.remove(video_path)
  
  
  
          process = Popen(
  
          "{} {}".format(ffmpeg_command, video_path),
  
          shell=True, stdout=PIPE, stderr=PIPE)
  
          self.__process = process
  
  
  
      def tear_down_99(self, method):
  
          # self.__process.terminate()
  
          # os.killpg(self.__process.pid, signal.CTRL_C_EVENT)
  
          os.kill(self.__process.pid, signal.CTRL_C_EVENT)
  
          # terminate(self.__process)

  my system is windows 10 , i have try send signal to kill the recording process, but if use Popen terminate or kill method , the ffmpeg_command does not real stop, if send ctrl_c_event signal, video can be saved but the next test case not execute forever, so i want to ask there is a way to resolve this, that can execute each case and each case execute process can be recording to a vedio, thanks !

  ffmpeg cammand like that :

  D:\\ffmpeg-20170214-8fa18e0-win64-static\\bin\\ffmpeg -video_size 1366x768 -f gdigrab -offset_x 0 -offset_y 0 -framerate 60 -i desktop