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
[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?
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