get white noise after convert it into mp3 format

[henry]

Summary of the bug:
i used ffmpeg 0.8.7 convert a wav file into mp3 format , but only get white noise in target mp3 format.

How to reproduce:

 ffmpeg -i 01.wav  -y /home/share/01.mp3
ffmpeg version 0.8.7, Copyright (c) 2000-2011 the FFmpeg developers
  built on Jan 16 2012 15:16:29 with gcc 4.4.5 20101112 (Red Hat 4.4.5-2)
  configuration: --prefix=/usr/local/enctools --enable-postproc --
enable-avfilter --enable-libopencore-amrnb --enable-libopencore-amrwb -
-enable-version3 --enable-libmp3lame --disable-shared --enable-libx264 
--enable-pthreads --enable-libfaac --enable-nonfree --enable-gpl --
enable-libgsm --enable-libnut --enable-libspeex --enable-libtheora --
enable-libvorbis --enable-libopenjpeg --enable-libxvid --enable-bzlib -
-enable-libschroedinger --enable-zlib --enable-small --disable-network 
--disable-debug --extra-cflags=-I/usr/local/enctools/include --extra-ldflags=-L/usr/local/enctools/lib
  libavutil    51.  9. 1 / 51.  9. 1
  libavcodec   53.  8. 0 / 53.  8. 0
  libavformat  53.  5. 0 / 53.  5. 0
  libavdevice  53.  1. 1 / 53.  1. 1
  libavfilter   2. 23. 0 /  2. 23. 0
  libswscale    2.  0. 0 /  2.  0. 0
  libpostproc  51.  2. 0 / 51.  2. 0
[wav @ 0x175f400] max_analyze_duration 5000000 reached at 5015510
Input #0, wav, from '01.wav':
  Duration: 00:02:05.59, bitrate: 1411 kb/s
    Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
Output #0, mp3, to '/home/share/01.mp3':
  Metadata:
    TSSE            : Lavf53.5.0
    Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
Press [q] to stop, [?] for help
size=     982kB time=00:02:05.62 bitrate=  64.0kbits/s read=   21635kB    
video:0kB audio:981kB global headers:0kB muxing overhead 0.018707%

[Carl Eugen Hoyos]

Duplicate of ticket #1747, please update to current git head.

[Carl Eugen Hoyos]

Work-around is to use:
$ ffmpeg -f dts -i 01.wav 01.mp3

[Carl Eugen Hoyos]

Replying to cehoyos:

Work-around is to use:
$ ffmpeg -f dts -i 01.wav 01.mp3

Scratch this, the correct work-around is:
$ ffmpeg -acodec dts -i 01.wav 01.mp3

[henry]

Replying to cehoyos:

Replying to cehoyos:

Work-around is to use:
$ ffmpeg -f dts -i 01.wav 01.mp3

Scratch this, the correct work-around is:
$ ffmpeg -acodec dts -i 01.wav 01.mp3

but i would like to let ffmpeg detect codec automatically and set in scripts , for this file(dca) it works , but for other normal files (like,ac3,aac,etc), it will failed.

[henry]

any prompt or info in ffmpeg i should use "-acodec dts " ? because seems all info from ffmpeg as same as other normal files.

[Carl Eugen Hoyos]

Replying to henry:

$ ffmpeg -acodec dts -i 01.wav 01.mp3

but i would like to let ffmpeg detect codec automatically

Then please update to current FFmpeg.

[henry]

right , upgrade to 1.0 and it get correct info :

ffmpeg  -i 01.wav 
ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers
  built on Sep 29 2012 15:36:05 with gcc 4.4.6 (GCC) 20120305 (Red Hat 4.4.6-4)
  libavutil      51. 73.101 / 51. 73.101
  libavcodec     54. 59.100 / 54. 59.100
  libavformat    54. 29.104 / 54. 29.104
  libavdevice    54.  2.101 / 54.  2.101
  libavfilter     3. 17.100 /  3. 17.100
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[dca @ 0x1d56760] Not a valid DCA frame
[dca @ 0x1d56760] Number of channels changed in DCA decoder (2 -> 6)
[wav @ 0x1d50240] max_analyze_duration 5000000 reached at 5015510
Input #0, wav, from '01.wav':
  Duration: 00:00:10.00, bitrate: 1411 kb/s
    Stream #0:0: Audio: dts (DTS) ([1][0][0][0] / 0x0001), 44100 Hz, 5.1(side), s16, 1411 kb/s
At least one output file must be specified