कैसे पीसीएम WAV को 3GP फ़ाइल कन्वर्ट करने के लिए Ubuntu पर ffmpeg को कॉन्फ़िगर करें?


0

मैं Ubuntu 10.04 का उपयोग कर रहा हूं। मुझे एक .3gp फ़ाइल को PCM WAV में बदलने की आवश्यकता है। मैं उस के लिए ffmpeg का उपयोग कर रहा हूँ।

जब इसे रिपॉजिटरी से इंस्टॉल किया जाता है, तो aptitude install ffmpegइसका कुछ मूल संस्करण इंस्टॉल करके इसका उपयोग किया जाता है और मुझे इसकी आवश्यकता नहीं होती है।

मैंने नवीनतम यास्मीन ver.1.1.0 और नवीनतम x264 - 0.125.2208 स्थापित किया है। उसके बाद मैं से Git का उपयोग कर ffmpeg मिला आधिकारिक होमपेज के साथ git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg

मैंने स्वयं का उपयोग करके ffmpeg को कॉन्फ़िगर करने का प्रयास किया:

./configure --enable-gpl --enable-version3 --enable-postproc 
--enable-nonfree --enable-swscale --enable-pthreads --enable-libmp3lame 
--enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb

तब: time make && make install

इस समय तक सब कुछ ठीक था। के साथ रूपांतरण के बाद

ffmpeg -i audiotest.3gp -f s16le -ar 8000 -acodec pcm_s16le audio.wav

मैं इस PCM .wav फ़ाइल (ffmpeg -i audio.wav) के बारे में जानकारी जाँचना चाहता था और मुझे यह त्रुटि मिली:

~# ffmpeg -i audio.wav

ffmpeg version N-42619-g6b7849e Copyright (c) 2000-2012 the FFmpeg developers
  built on Jul 21 2012 00:50:52 with gcc 4.4.3
  configuration: --enable-gpl --enable-version3 --enable-postproc --enable-nonfree --enable-swscale --enable-pthreads --enable-libmp3lame --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb

  libavutil      51. 65.100 / 51. 65.100
  libavcodec     54. 41.100 / 54. 41.100
  libavformat    54. 17.100 / 54. 17.100
  libavdevice    54.  1.100 / 54.  1.100
  libavfilter     3.  2.100 /  3.  2.100
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[aac @ 0x943d4e0] Format aac detected only with low score of 1, misdetection possible!
[aac @ 0x9443740] channel element 0.0 is not allocated
    Last message repeated 2 times
[aac @ 0x9443740] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Number of bands (16) exceeds limit (4).
[aac @ 0x9443740] Number of bands (7) exceeds limit (2).
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] SSR not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x9443740] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x9443740] channel element 2.0 is not allocated
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Number of bands (31) exceeds limit (1).
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Number of bands (16) exceeds limit (2).
[aac @ 0x9443740] channel element 0.7 is not allocated
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Number of scalefactor bands in group (62) exceeds limit (41).
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.2 is not allocated
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] channel element 0.15 is not allocated
[aac @ 0x9443740] Pulse data corrupt or invalid.
[aac @ 0x9443740] Number of scalefactor bands in group (48) exceeds limit (41).
[aac @ 0x9443740] channel element 2.0 is not allocated
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Number of bands (16) exceeds limit (4).
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
    Last message repeated 1 times
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] channel element 2.0 is not allocated
[aac @ 0x9443740] Number of bands (31) exceeds limit (4).
[aac @ 0x9443740] Pulse data corrupt or invalid.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] SSR not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x9443740] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.3 is not allocated
[aac @ 0x9443740] Pulse data corrupt or invalid.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Number of bands (35) exceeds limit (16).
[aac @ 0x9443740] Number of scalefactor bands in group (63) exceeds limit (41).
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Number of bands (38) exceeds limit (10).
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] channel element 0.2 is not allocated
[aac @ 0x9443740] channel element 0.7 is not allocated
[aac @ 0x9443740] Reserved bit set.
    Last message repeated 2 times
[aac @ 0x9443740] channel element 0.2 is not allocated
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
    Last message repeated 1 times
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] decode_band_types: Input buffer exhausted before END element found
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Error decoding AAC frame header.
    Last message repeated 1 times
[aac @ 0x9443740] Reserved bit set.
    Last message repeated 1 times
[aac @ 0x9443740] Number of bands (4) exceeds limit (1).
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Reserved bit set.
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x9443740] Number of bands (31) exceeds limit (8).
[aac @ 0x9443740] Invalid Predictor Reset Group.
[aac @ 0x9443740] Number of bands (31) exceeds limit (2).
[aac @ 0x9443740] Number of bands (28) exceeds limit (1).
[aac @ 0x9443740] channel element 0.0 is not allocated
[aac @ 0x9443740] Input buffer exhausted before END element found
[aac @ 0x9443740] Number of bands (16) exceeds limit (2).
[aac @ 0x9443740] Error decoding AAC frame header.
[aac @ 0x943d4e0] decoding for stream 0 failed
[aac @ 0x943d4e0] Could not find codec parameters for stream 0 (Audio: aac, 4.0, s16, 383 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[aac @ 0x943d4e0] Estimating duration from bitrate, this may be inaccurate
audio.wav: could not find codec parameters

क्या कोई मुझे इस बारे में सहायता कर सकता है? मैं क्या गलत कर रहा हूँ?


कृपया हमें वास्तविक एन्कोडिंग प्रक्रिया के पूर्ण, काटा हुआ उत्पादन, दिखाने के नहीं अपने आउटपुट फ़ाइल से।
slhck

जवाबों:


1

जैसा कि पहली त्रुटि संदेश द्वारा सुझाया गया है:

Format aac detected only with low score of 1, misdetection possible!

यह गलत तरीके से इनपुट फ़ाइल प्रकार का पता लगा रहा है। -fविकल्प का उपयोग करके इनपुट फ़ाइल प्रारूप को निर्दिष्ट करें :

ffmpeg -f s16le -i input.wav

और यह बेहतर काम करना चाहिए।

हालाँकि, यदि आप केवल फ़ाइल के बारे में जानकारी प्राप्त करना चाहते हैं, तो आपको इसके बजाय FFprobe का उपयोग करना चाहिए । यह आमतौर पर एफएफएमपी के साथ पैक किया जाता है, समान विकल्प लेता है और प्रारूप को पार्स करने के लिए बहुत आसान जानकारी प्रदान करता है। -show_formatऔर -show_streamsविकल्प आप सबसे सभी जानकारी आप एक फ़ाइल के बारे में की जरूरत है देना चाहिए।

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