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pydub.exceptions.CouldntDecodeError : Decoding failed. ffmpeg returned error code : 1
9 avril, par azail765This script will work on a 30 second wav file but not a 10 minutes phone call also in wav format. Any help would be appreciated
I've downloaded ffmpeg.
# Import necessary libraries from pydub import AudioSegment import speech_recognition as sr import os import pydub chunk_count = 0 directory = os.fsencode(r'C:\Users\zach.blair\Downloads\speechRecognition\New folder') # Text file to write the recognized audio fh = open("recognized.txt", "w+") for file in os.listdir(directory): filename = os.fsdecode(file) if filename.endswith(".wav"): chunk_count += 1 # Input audio file to be sliced audio = AudioSegment.from_file(filename,format="wav") ''' Step #1 - Slicing the audio file into smaller chunks. ''' # Length of the audiofile in milliseconds n = len(audio) # Variable to count the number of sliced chunks counter = 1 # Interval length at which to slice the audio file. interval = 20 * 1000 # Length of audio to overlap. overlap = 1 * 1000 # Initialize start and end seconds to 0 start = 0 end = 0 # Flag to keep track of end of file. # When audio reaches its end, flag is set to 1 and we break flag = 0 # Iterate from 0 to end of the file, # with increment = interval for i in range(0, 2 * n, interval): # During first iteration, # start is 0, end is the interval if i == 0: start = 0 end = interval # All other iterations, # start is the previous end - overlap # end becomes end + interval else: start = end - overlap end = start + interval # When end becomes greater than the file length, # end is set to the file length # flag is set to 1 to indicate break. if end >= n: end = n flag = 1 # Storing audio file from the defined start to end chunk = audio[start:end] # Filename / Path to store the sliced audio filename = str(chunk_count)+'chunk'+str(counter)+'.wav' # Store the sliced audio file to the defined path chunk.export(filename, format ="wav") # Print information about the current chunk print(str(chunk_count)+str(counter)+". Start = " +str(start)+" end = "+str(end)) # Increment counter for the next chunk counter = counter + 1 AUDIO_FILE = filename # Initialize the recognizer r = sr.Recognizer() # Traverse the audio file and listen to the audio with sr.AudioFile(AUDIO_FILE) as source: audio_listened = r.listen(source) # Try to recognize the listened audio # And catch expections. try: rec = r.recognize_google(audio_listened) # If recognized, write into the file. fh.write(rec+" ") # If google could not understand the audio except sr.UnknownValueError: print("Empty Value") # If the results cannot be requested from Google. # Probably an internet connection error. except sr.RequestError as e: print("Could not request results.") # Check for flag. # If flag is 1, end of the whole audio reached. # Close the file and break. fh.close()I get this error on
audio = AudioSegment.from_file(filename,format="wav"):Traceback (most recent call last): File "C:\Users\zach.blair\Downloads\speechRecognition\New folder\speechRecognition3.py", line 17, inaudio = AudioSegment.from_file(filename,format="wav") File "C:\Users\zach.blair\AppData\Local\Programs\Python\Python37-32\lib\site-packages\pydub\audio_segment.py", line 704, in from_file p.returncode, p_err)) pydub.exceptions.CouldntDecodeError: Decoding failed. ffmpeg returned error code: 1 Output from ffmpeg/avlib:
ffmpeg version N-95027-g8c90bb8ebb Copyright (c) 2000-2019 the FFmpeg developers built with gcc 9.2.1 (GCC) 20190918 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf libavutil 56. 35.100 / 56. 35.100 libavcodec 58. 58.101 / 58. 58.101 libavformat 58. 33.100 / 58. 33.100 libavdevice 58. 9.100 / 58. 9.100 libavfilter 7. 58.102 / 7. 58.102 libswscale 5. 6.100 / 5. 6.100 libswresample 3. 6.100 / 3. 6.100 libpostproc 55. 6.100 / 55. 6.100 Guessed Channel Layout for Input Stream #0.0 : mono Input #0, wav, from '2a.wav.wav': Duration: 00:09:52.95, bitrate: 64 kb/s Stream #0:0: Audio: pcm_mulaw ([7][0][0][0] / 0x0007), 8000 Hz, mono, s16, 64 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_mulaw (native) -> pcm_s8 (native)) Press [q] to stop, [?] for help [wav @ 0000024307974400] pcm_s8 codec not supported in WAVE format Could not write header for output file #0 (incorrect codec parameters ?): Function not implemented Error initializing output stream 0:0 -- Conversion failed! -
How to dump ALL metadata from a media file, including cover image title ? [closed]
9 avril, par UnidealI have an MP3 song:
# ffprobe -hide_banner -i filename.mp3 Input #0, mp3, from 'filename.mp3': Metadata: composer : Music Author title : Song Name artist : Singer encoder : Lavf61.7.100 genre : Rock date : 2025 Duration: 00:03:14.04, start: 0.023021, bitrate: 208 kb/s Stream #0:0: Audio: mp3 (mp3float), 48000 Hz, stereo, fltp, 192 kb/s Metadata: encoder : Lavc61.19 Stream #0:1: Video: png, rgb24(pc, gbr/unknown/unknown), 600x600 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn (attached pic) Metadata: title : Cover comment : Cover (front)The task is to save its metadata to a text file and restore from that file later. Both goals should be accomplished with ffmpeg.
The simpliest method is to run:
# ffmpeg -i filename.mp3 -f ffmetadata metadata.txtAfter that,
metadata.txtcontains:;FFMETADATA1 composer=Music Author title=Song Name artist=Singer date=2025 genre=Rock encoder=Lavf61.7.100I got global metadata only, but stream-specific info (cover image title and comment in my case) are missing.
Google suggested a more complex form of the command above to extract all metadata fields without any exclusions:
# ffmpeg -y -i filename.mp3 -c copy -map_metadata 0 -map_metadata:s:v 0:s:v -map_metadata:s:a 0:s:a -f ffmetadata metadata.txtBut the output is exactly the same:
;FFMETADATA1 composer=Music Author title=Song Name artist=Singer date=2025 genre=Rock encoder=Lavf61.7.100Again, no info about the attached image.
Please explain what am I doing wrong.
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some codecs (ie. libx264) cannot be reused after draining
7 avril, par NeddieAfter draining an encoder (by sending it a null frame and then receiving packets until EOF) it enters draining mode after which
avcodec_send_framewill fail, returning EOF. You're supposed to callavcodec_flush_bufferswhich among other things sets the internaldrainingflag to 0, allowingavcodec_send_frameto work again. Unfortunately it checksAV_CODEC_CAP_ENCODER_FLUSHfirst and if not set it returns without resetting thedrainingflag.So the only way to reuse the codec is to close and reopen it, which is wasteful, plus
avcodec_closeis deprecated. Am I missing something?int attribute_align_arg avcodec_send_frame(AVCodecContext *avctx, const AVFrame *frame) { AVCodecInternal *avci = avctx->internal; int ret; if (!avcodec_is_open(avctx) || !av_codec_is_encoder(avctx->codec)) return AVERROR(EINVAL); if (avci->draining) return AVERROR_EOF; if (avci->buffer_frame->buf[0]) return AVERROR(EAGAIN); if (!frame) { avci->draining = 1; } else { ret = encode_send_frame_internal(avctx, frame); if (ret < 0) return ret; } if (!avci->buffer_pkt->data && !avci->buffer_pkt->side_data) { ret = encode_receive_packet_internal(avctx, avci->buffer_pkt); if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) return ret; } avctx->frame_num++; return 0; } void avcodec_flush_buffers(AVCodecContext *avctx) { AVCodecInternal *avci = avctx->internal; if (av_codec_is_encoder(avctx->codec)) { int caps = avctx->codec->capabilities; if (!(caps & AV_CODEC_CAP_ENCODER_FLUSH)) { // Only encoders that explicitly declare support for it can be // flushed. Otherwise, this is a no-op. av_log(avctx, AV_LOG_WARNING, "Ignoring attempt to flush encoder " "that doesn't support it\n"); return; } ff_encode_flush_buffers(avctx); } else ff_decode_flush_buffers(avctx); avci->draining = 0; avci->draining_done = 0; if (avci->buffer_frame) av_frame_unref(avci->buffer_frame); if (avci->buffer_pkt) av_packet_unref(avci->buffer_pkt); if (HAVE_THREADS && avctx->active_thread_type & FF_THREAD_FRAME && !avci->is_frame_mt) ff_thread_flush(avctx); else if (ffcodec(avctx->codec)->flush) ffcodec(avctx->codec)->flush(avctx); } -
How to automatically rotate video based on camera orientation while recording ?
7 avril, par jestrabikrI am developing Mediasoup SFU and client web app, on the server I am recording client's stream by sending it to FFmpeg as plain RTP. FFmpeg creates HLS recording (.m3u8 and .ts files), because I need to be able to switch between WebRTC live stream and HLS recording before live stream ends.
My problem is that, when I am testing the app and I rotate my phone 90 degrees, the recording's aspect ratio stays the same but the image is rotated (as shown on images 1.1, 1.2 and 1.3 below). I need for it to change aspect ratio dynamically according to camera orientation. How can I do that using FFmpeg?
On the live stream it works perfectly fine (as shown on the images - 2.1 and 2.2 below), when the phone is rotated, the aspect ration is changed and video is shown correctly. I think it works on live stream because maybe WebRTC is signaling orientation changes somehow (but it does not project to the recording).
These are my ffmpeg command arguments for recording (version 6.1.1-3ubuntu5):
let commandArgs = [ "-loglevel", "info", "-protocol_whitelist", "pipe,udp,rtp", "-fflags", "+genpts+discardcorrupt", "-reinit_filter", "1", "-strict", "-2", "-f", "sdp", "-i", "pipe:0", "-map", "0:v:0", "-c:v", "libx264", "-b:v", "1500k", "-profile:v", "high", "-level:v", "4.1", "-pix_fmt", "yuv420p", "-g", "30", "-map", "0:a:0", "-c:a", "aac", "-b:a", "128k", "-movflags", "+frag_keyframe+empty_moov", "-f", "hls", "-hls_time", "4", "-hls_list_size", "0", "-hls_flags", "split_by_time", `${filePath}.m3u8` ];Image 1.1 - Portrait mode in recording:
Image 1.2 - Landscape mode in recording (rotated 90deg to my left side - front camera is on my left side):
Image 1.3 - Landscape mode in recording (rotated 90deg to my right side):
Image 2.1 - Portrait mode in live stream (correct behavior):
Image 2.2 - Landscape mode in live stream (correct behavior):
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ffmpeg : error while loading shared libraries : libnppig.so.12 : cannot open shared object file on CentOS [closed]
7 avril, par Rahib RasheedI'm trying to install ffmpeg on a CentOS server, but I'm running into a shared library error when I try to run it:
ffmpeg: error while loading shared libraries: libnppig.so.12: cannot open shared object file: No such file or directoryRunning ldd shows some missing dependencies:
ldd $(which ffmpeg) | grep libnpp libnppig.so.12 => not found libnppicc.so.12 => /lib64/libnppicc.so.12 libnppidei.so.12 => /lib64/libnppidei.so.12 libnppif.so.12 => not found libnppc.so.12 => /lib64/libnppc.so.12I’ve tried searching for the missing libraries but couldn’t find the correct packages for CentOS. What’s the best way to resolve this issue?
