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• M4a (mp4) audio file encoded with pydub+ffmpeg doesn't play on Android

  6 juillet 2020, par Evelyn

  I have a python script to split up some wav files and export to m4a using pydub. I'm able to get these files to play on several devices, but not on an Android device (using Google Pixel 3). When I try encoding with straight ffmpeg in terminal, that works fine on the Android device.

  


  What is the difference in these two files, and since pydub is using ffmpeg, what do I need to change to make it do exactly the same as the ffmpeg command ?

  


  Not working

  


  from pydub import AudioSegment
>>> audio = AudioSegment.from_wav("input.wav")
>>> slice = audio[1000:3000]
>>> slice.export("pydub_export.m4a", format="mp4", parameters=["-ac", "1", "-c:a", "libfdk_aac", "-profile:a", "aac_he", "-vbr", "2"])


  


  mediainfo output :

  


  General
Complete name                            : pydub_export.m4a
Format                                   : MPEG-4
Format profile                           : Base Media
Codec ID                                 : isom (isom/iso2/mp41)
File size                                : 11.7 KiB
Duration                                 : 2 s 115 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 45.1 kb/s
Writing application                      : Lavf58.29.100

Audio
ID                                       : 1
Format                                   : AAC LC SBR
Format/Info                              : Advanced Audio Codec Low Complexity with Spectral Band Replication
Commercial name                          : HE-AAC
Format settings                          : NBC
Codec ID                                 : mp4a-40-5
Duration                                 : 2 s 115 ms
Duration_LastFrame                       : -22 ms
Bit rate mode                            : Constant
Bit rate                                 : 41.4 kb/s
Channel(s)                               : 1 channel
Channel layout                           : C
Sampling rate                            : 44.1 kHz
Frame rate                               : 21.533 FPS (2048 SPF)
Compression mode                         : Lossy
Stream size                              : 10.7 KiB (92%)
Default                                  : Yes
Alternate group                          : 1


  


  Working

  


  $ffmpeg -i input.wav -acodec copy -ss 1 -to 3 input_slice.wav
$ffmpeg -i input_slice.wav -ac 1 -c:a libfdk_aac -profile:a aac_he -vbr 2 ffmpeg_export.m4a


  


  mediainfo output :

  


  General
Complete name                            : ffmpeg_export.m4a
Format                                   : MPEG-4
Format profile                           : Apple audio with iTunes info
Codec ID                                 : M4A  (isom/iso2)
File size                                : 11.2 KiB
Duration                                 : 2 s 112 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 43.6 kb/s
Writing application                      : Lavf58.29.100

Audio
ID                                       : 1
Format                                   : AAC LC SBR
Format/Info                              : Advanced Audio Codec Low Complexity with Spectral Band Replication
Commercial name                          : HE-AAC
Format settings                          : NBC
Codec ID                                 : mp4a-40-5
Duration                                 : 2 s 112 ms
Duration_LastFrame                       : -25 ms
Bit rate mode                            : Constant
Bit rate                                 : 39.9 kb/s
Channel(s)                               : 1 channel
Channel layout                           : C
Sampling rate                            : 44.1 kHz
Frame rate                               : 21.533 FPS (2048 SPF)
Compression mode                         : Lossy
Stream size                              : 10.3 KiB (91%)
Default                                  : Yes
Alternate group                          : 1


  


  I already tried moving metadata to the front with -movflags faststart on the broken file and it didn't make a difference.

  


• -use_wallclock_as_timestamps adds delay in live stream

  2 novembre 2020, par Arikael

  We have a livestream (MPEG-TS with RTP), which we currently, for testing purposes, replay with tcpreplay.

  


  Our mpeg-ts stream consists of 4 streams (codec details omitted for brevity).

  


  Stream #0:2: Video: h264
Stream #0:1: Audio: mp2
Stream #0:4: Data: bin_data ([6][0][0][0] / 0x0006)
Strean #0:3: Data: bin_data (FBID / 0x4494246)
Stream #0:0: Data: klv (KLVA / 0x41564C4B)


  


  Sometimes the stream indexes are different (like audio stream being stream 0 and so on, I don't know if thats normal behavior)

  


  What we currently try is just to get the stream and copy it with ffmpeg, like

  


  ffmpeg -nostdin -hide_banner -i rtp://239.0.0.2:3000 -map 0 -codec copy -f rtp_mpegts rtp://239.0.0.1:2000`


  


  This leads to the error Application provided invalid, non monotonically incereasing dts to muxer in stream 0: [NUMBER] >= 0

  


  It always says stream 0 no matter what stream 0 contains.

  


  if I add use_wallclock_as_timestamps it works but adds a delay (compared to a video directly streamed from 239.0.0.2:3000 of 10seconds which are never caught up.

  


  If I set the output format to mpegts instead of rpt_mpegts it works as expected, the same I if don't map the KLVA and FBID stream.

  


  Is this behavior expected (because of wallclock) or what can I do to either use_wallclock_as_timestamps without delay or get rid of the error above ?

  


• FFMPEG : Microphone capturing too much noise while using ffmpeg command

  29 octobre 2020, par Rupesh J

  I am listening audio from source IP address and trying to encode it into speex format and again sending it to the destination IP address using ffmpeg.

  


  My ffmpeg command is :

  


  ffmpeg -protocol_whitelist file,rtp,udp -i temp.sdp -c:a libspeex -f rtp rtp://:<port>&#xA;</port>

  


  SDP file content is(temp.sdp) :

  


  v=0 &#xA;c=IN IP4 &#xA;t=0 0&#xA;m=audio <port> RTP/AVP 98&#xA;a=rtpmap:98 L16/8000&#xA;</port>

  


  Issue : Whenever I am trying to run this command, I am getting too much background noise on speaker.
I could hear music(not clearly), but not human voice.

  


  Also, I have tried with highpass and lowpass filters are as follows :

  


  ffmpeg -protocol_whitelist file,rtp,udp -i temp.sdp -af "highpass=f=200, lowpass=f=3000" -c:a &#xA;    libspeex -f rtp rtp://:<port>&#xA;</port>