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Médias (91)
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#3 The Safest Place
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#4 Emo Creates
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#2 Typewriter Dance
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#1 The Wires
11 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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ED-ME-5 1-DVD
11 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Audio
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Revolution of Open-source and film making towards open film making
6 octobre 2011, par
Mis à jour : Juillet 2013
Langue : English
Type : Texte
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Support de tous types de médias
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Sur d’autres sites (5397)
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Convert 2 channel mp4 to each mono wav file using FFMPEG or Python code
30 mai 2024, par Harish AlwalaI am new to audio files and its codecs.


I would like to convert a 2 channel mp4 file to a single mono wav files.


My understanding is a when I say 2 channel, it stores speech coming from each microphone in a separate channel. And when I split the channels to each individual mono wav files, I get speech of each microphone.


My intension here is to get the speech from each channel and convert them to text. This way I can set the name of the speaker based on channel.


I tried with ffmpeg and python code as well, unfortunately I get two files with same content.


Looking at the following details
can someone construct ffmpeg command or python script to convert the 2 channel mp4 file to 2 individual mono wav files.


FFprobe
ffprobe -i Two-Channel.mp4 -show_streams -select_streams a


Result


Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: isommp42
 encoder : Google
 Duration: 00:52:42.19, start: 0.000000, bitrate: 421 kb/s
 Stream #0:0[0x1](und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 640x360 [SAR 1:1 DAR 16:9], 322 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
 Metadata:
 handler_name : ISO Media file produced by Google Inc.
 vendor_id : [0][0][0][0]
 Stream #0:1[0x2](eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 96 kb/s (default)
 Metadata:
 handler_name : ISO Media file produced by Google Inc.
 vendor_id : [0][0][0][0]
[STREAM]
index=1
codec_name=aac
codec_long_name=AAC (Advanced Audio Coding)
profile=LC
codec_type=audio
codec_tag_string=mp4a
codec_tag=0x6134706d
sample_fmt=fltp
sample_rate=44100
channels=2
channel_layout=stereo
bits_per_sample=0
initial_padding=0
id=0x2
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/44100
start_pts=0
start_time=0.000000
duration_ts=139452416
duration=3162.186304
bit_rate=96000
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=136184
nb_read_frames=N/A
nb_read_packets=N/A
extradata_size=16
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
DISPOSITION:non_diegetic=0
DISPOSITION:captions=0
DISPOSITION:descriptions=0
DISPOSITION:metadata=0
DISPOSITION:dependent=0
DISPOSITION:still_image=0
TAG:language=eng
TAG:handler_name=ISO Media file produced by Google Inc.
TAG:vendor_id=[0][0][0][0]
[/STREAM] 



FFmpeg command


ffmpeg -i Two-Channel.mp4 -filter_complex "pan=mono|c0=0c0" left_channel.wav


python code
using FFPMEG I converted mp4 to wav and then tried below code





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Get PTS from raw H264 mdat generated by iOS AVAssetWriter
26 décembre 2012, par kolyuchiyI'm trying to simultaneously read and write H.264 mov file written by AVAssetWriter. I managed to extract individual NAL units, pack them into ffmpeg's AVPackets and write them into another video format using ffmpeg. It works and the resulting file plays well except the playback speed is not right. How do I calculate the correct PTS/DTS values from raw H.264 data ? Or maybe there exists some other way to get them ?
Here's what I've tried :
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Limit capture min/max frame rate to 30 and assume that the output file will be 30 fps. In fact its fps is always less than values that I set. And also, I think the fps is not constant from packet to packet.
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Remember each written sample's presentation timestamp and assume that samples map one-to-one to NALUs and apply saved timestamp to output packet. This doesn't work.
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Setting PTS to 0 or AV_NOPTS_VALUE. Doesn't work.
From googling about it I understand that raw H.264 data usually doesn't contain any timing info. It can sometimes have some timing info inside SEI, but the files that I use don't have it. On the other hand, there are some applications that do exactly what I'm trying to do, so I suppose it is possible somehow.
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lavc/h264dec : don't error out when receiving multiple IDR slices
26 septembre 2018, par Josh de Kocklavc/h264dec : don't error out when receiving multiple IDR slices
This error isn't particularly helpful as checking for mixed IDR/non-IDR
NALUs would need to be done at a higher level to actually be accurate.
Removing the error allows an API user to send individual slice NALUs
(i.e. incomplete frames) so they can take advantage of slice
threading. The ticket which this error was added for (#4408) no
longer segfaults after removing this error (as the bug was likely
fixed more properly elsewhere).