I assume you are talking about encoding?
On Wednesday, December 8, 2021, 02:26:22 PM EST, Bob Kirnum
<[email protected]> wrote:
I assume there is a sample format for 32 bit float, try that instead of
AV_SAMPLE_FMT_S16. If that does not work you may need to convert samples from
32 bit float to 16 bit integer.
On Wed, Dec 8, 2021 at 2:21 PM Robert Smith via Libav-user
<[email protected]> wrote:
Dumped raw data to binary file, opened it in Audacity, found correct sampling
rate and correctly played audio clip.
I noticed that Audacity shows "32 bit float", as you mentioned in your previous
E-mail that could be the reason why transcoding to uLaw fails.
Could you clarify what is proper way to resolve the issue? Should I correct
decoding? Thanks
On Wednesday, December 8, 2021, 12:41:41 PM EST, Bob Kirnum
<[email protected]> wrote:
To verify the decoded audio from the MP3 I would simply write the decoded
frames to file (fwrite or similar to a binary file). The resulting file is a
raw PCM file which can be imported with Audacity. You specify the sample rate,
sample format (i.e. 16 bit integer or 32 bit float), etc. When you get the
right settings the audio should play good.
On Wed, Dec 8, 2021 at 12:01 PM Robert Smith via Libav-user
<[email protected]> wrote:
Hi,
Thank you for your response. All my attempts to sort it out failed so far.
Could you clarify how can I verify that I decoded the MP3 to linear PCM
properly?
I mean as result of decoding I'm getting frames:
avcodec_decode_audio4(pMp3CodecCtx, frame, &gotFrame, &packet)
So what is next? Write frame to wav file? How? The function
av_interleaved_write_frame despite its name accept packets not a frames?
How to check validity of wav container? I usually run "ffprobe -i Output.wav"
to see details. Is there other way? Thanks
On Monday, December 6, 2021, 11:43:17 AM EST, Bob Kirnum
<[email protected]> wrote:
Perhaps the AV_SAMPLE_FMT_S16 is referring to the sample format input to the
G.711 uLaw encoder (input should be 16 bit linear PCM)? Then this would be
correct. Perhaps split the effort and verify that you can decode the MP3 to
linear PCM properly, and encode linear to uLaw properly for WAV separately?
Using the wrong sample rate (not downsampling to 8 kHz) would not result in
noise (what you reported), would just result in the wrong audio speed. One
additional thought, does decoding the MP3 result in 16 bit integer linear PCM
or 32 bit float PCM? If float, that would be a mismatch and result in noise
when encoding to uLaw.
On Mon, Dec 6, 2021 at 10:40 AM Robert Smith via Libav-user
<[email protected]> wrote:
If I change
pCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
to
pCodecCtx->sample_fmt = AV_SAMPLE_FMT_U8;
Then I get the following error:
[pcm_mulaw @ 0x2d33450] Specified sample_fmt is not supported.
Any idea? Thanks
On Friday, December 3, 2021, 06:15:54 PM EST, Bob Kirnum
<[email protected]> wrote:
G.711 is 8 bits, could be an issue. Also, not sure if you need to downsample
to 8kHz manually or not.
> On Dec 3, 2021, at 5:36 PM, Robert Smith via Libav-user
> <[email protected]> wrote:
>
> pCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
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