[Libav-user] filtering_audio.c example not working

[Stefano Sabatini]

On date Monday 2012-11-12 14:50:24 -0600, Ron Woods wrote:
> I am trying out the filtering_audio.c example provided with the ffmpeg libraries for Windows to extract the audio from a MP4 file, resample to 8 KHz and convert from stereo to mono. The example pipes the audio output via stdout to ffplay. I am using Visual Studio 2010 and the example successfully builds and runs but the result is clearly not the desired result. At the end of init_filters I added a call to avfilter_graph_dump() and it all looks correct and also the pipe to ffplay as in this trace:
> 
> abuffer filter args: time_base=1/24000:sample_rate=24000:sample_fmt=s16:channel_layout=0x4
> Output: srate:8000Hz fmt:s16 chlayout:mono
> +-----------+
> |    in     |default--[24000Hz s16:mono]--Parsed_aconvert_0:default
> | (abuffer) |
> +-----------+
> 
>                                                       +-----------------+
> Parsed_aresample_1:default--[8000Hz s16:mono]--default|       out       |
>                                                       | (ffabuffersink) |
>                                                       +-----------------+
> 
>                                        +-------------------+
> in:default--[24000Hz s16:mono]--default| Parsed_aconvert_0 |default--[24000Hz s16:mono]-Parsed_aresample_1:default
>                                        |    (aconvert)     |
>                                        +-------------------+
> 
>                                                       +--------------------+
> Parsed_aconvert_0:default--[24000Hz s16:mono]--default| Parsed_aresample_1 |default--[8000Hz s16:mono]--out:default
>                                                       |    (aresample)     |
>                                                       +--------------------+

> [s16le @ 003edda0] Invalid sample rate 0 specified using default of 44100

This is fishy.

> [s16le @ 003edda0] Estimating duration from bitrate, this may be inaccurate
> Input #0, s16le, from 'pipe:':
>   Duration: N/A, start: 0.000000, bitrate: 128 kb/s
>     Stream #0:0: Audio: pcm_s16le, 8000 Hz, 1 channels, s16, 128 kb/s
> 
> If you have made this example run properly on Windows in VS 2010, would you please provide any tips or changes you made for it to work?

I just tried latest git and seems to work fine here (Linux). Does the
problem depend on the input file?
-- 
FFmpeg = Funny and Fierce Merciless Practical Extended Goblin