How to specify Audio bit depth for test sources?
Hi. Different test sources have different bit depth: ffplay -f lavfi aevalsrc="0" # Audio: pcm_f64le ffplay -f lavfi anullsrc # Audio: pcm_u8 If I want to do some PCM math in the audio filter, I'll need to adapt the filter for each different input bit depth. There's no filter to change the depth. "aresample" only changes the sample rate. -acodec option won't do here, because it takes effect AFTER filtering and some tools don't have it (ffplay). -isf: I was unable to make it work The only workaround is to use /dev/zero as the source. Then you can specify the needed input format.
2017-03-02 16:24 GMT+01:00 Basin Ilya <basinilya@gmail.com>:
Hi. Different test sources have different bit depth:
ffplay -f lavfi aevalsrc="0" # Audio: pcm_f64le
Output of the aevalsrc filter is AV_SAMPLE_FMT_DBLP.
ffplay -f lavfi anullsrc # Audio: pcm_u8
Output of the anullsrc filter is AV_SAMPLE_FMT_U8 If that is not what you need, you have to use the aformat filter. Carl Eugen
2017-03-02 17:48 GMT+01:00 Carl Eugen Hoyos <ceffmpeg@gmail.com>:
2017-03-02 16:24 GMT+01:00 Basin Ilya <basinilya@gmail.com>:
Hi. Different test sources have different bit depth:
ffplay -f lavfi aevalsrc="0" # Audio: pcm_f64le
Output of the aevalsrc filter is AV_SAMPLE_FMT_DBLP.
ffplay -f lavfi anullsrc # Audio: pcm_u8
Output of the anullsrc filter is AV_SAMPLE_FMT_U8
If that is not what you need, you have to use the aformat filter.
Sorry, you have to use the aresample filter with the osf parameter. See https://ffmpeg.org/ffmpeg-resampler.html Carl Eugen
Thank you. On 02.03.2017 19:51, Carl Eugen Hoyos wrote:
2017-03-02 17:48 GMT+01:00 Carl Eugen Hoyos <ceffmpeg@gmail.com>:
2017-03-02 16:24 GMT+01:00 Basin Ilya <basinilya@gmail.com>:
Hi. Different test sources have different bit depth:
ffplay -f lavfi aevalsrc="0" # Audio: pcm_f64le
Output of the aevalsrc filter is AV_SAMPLE_FMT_DBLP.
ffplay -f lavfi anullsrc # Audio: pcm_u8
Output of the anullsrc filter is AV_SAMPLE_FMT_U8
If that is not what you need, you have to use the aformat filter.
Sorry, you have to use the aresample filter with the osf parameter.
See https://ffmpeg.org/ffmpeg-resampler.html
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participants (2)
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Basin Ilya -
Carl Eugen Hoyos