ffmpeg で mp3 の速度を変える
ffmpeg で mp3 の速度を変えたいのですが、
下記のようになりなぜかうまく行っていません。
mp3 の速度を変更するのは Audacity で成功しましたが
このやり方は少し面倒なので
大量のファイルを処理したい時などは ffmpeg を使いたいです。
OSは debian wheezy です。
どうぞご教授下さい。
よろしくお願いします。
$ ffmpeg -i 02.mp3 -af "atempo=0.8" 02_1.mp3
ffmpeg version 0.8.16-6:0.8.16-1, Copyright (c) 2000-2014 the Libav developers
built on Sep 16 2014 23:10:48 with gcc 4.7.2
The ffmpeg program is only provided for script compatibility and will be removed
in a future release. It has been deprecated in the Libav project to allow for
incompatible command line syntax improvements in its replacement called avconv
(see Changelog for details). Please use avconv instead.
[mp3 @ 0x112b460] max_analyze_duration reached
Input #0, mp3, from '02.mp3':
Metadata:
date : 1900
track : 1
artist : Unknown Artist
album : Unknown Album
title : Track 1
genre : Unknown
TLEN : 97906
encoder : Lavf53.21.1
Duration: 00:01:37.95, start: 0.000000, bitrate: 192 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 192 kb/s
Unrecognized option 'af'
Failed to set value 'atempo=0.8' for option 'af'
$ ffmpeg -i 02.mp3 -filter:a "atempo=0.8" 02_1.mp3
ffmpeg version 0.8.16-6:0.8.16-1, Copyright (c) 2000-2014 the Libav developers
built on Sep 16 2014 23:10:48 with gcc 4.7.2
The ffmpeg program is only provided for script compatibility and will be removed
in a future release. It has been deprecated in the Libav project to allow for
incompatible command line syntax improvements in its replacement called avconv
(see Changelog for details). Please use avconv instead.
[mp3 @ 0x10f3460] max_analyze_duration reached
Input #0, mp3, from '02.mp3':
Metadata:
date : 1900
track : 1
artist : Unknown Artist
album : Unknown Album
title : Track 1
genre : Unknown
TLEN : 97906
encoder : Lavf53.21.1
Duration: 00:01:37.95, start: 0.000000, bitrate: 192 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 192 kb/s
Unrecognized option 'filter:a'
Failed to set value 'atempo=0.8' for option 'filter:a'
補足
ffmpeg.exe -f dshow -i audio="ステレオ ミキサー (Realtek(R) Audio)" -acodec aac "audio.mp3" でやってみたのですが、以下のようになります。 [aist#0:0/pcm_s16le @ 0000019914dc1ec0] Guessed Channel Layout: stereo Input #0, dshow, from 'audio=ステレオ ミキサー (Realtek(R) Audio)': Duration: N/A, start: 176.076000, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s File 'audio.mp3' already exists. Overwrite? [y/N] [dshow @ 0000019912386d40] real-time buffer [ステレオ ミキサー (Realtek(R) Audio)] [audio input] too full or near too full (63% of size: 3041280 [rtbufsize parameter])! frame dropped! [dshow @ 0000019912386d40] real-time buffer [ステレオ ミキサー (Realtek(R) Audio)] [audio input] too full or near too full (72% of size: 3041280 [rtbufsize parameter])! frame dropped!