[Libav-user] How to transcode mp3 file to G.711 encoded wav file

[Bob Kirnum]

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 <
libav-user at ffmpeg.org> 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 <
> bkirnum at gmail.com> 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 <
> libav-user at ffmpeg.org> 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 <
> bkirnum at gmail.com> 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 <
> libav-user at ffmpeg.org> 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 <
> bkirnum at gmail.com> 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 <
> libav-user at ffmpeg.org> wrote:
> >
> > pCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
>
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