[Libav-user] QTKit -> Libav: has it ever been done?

Paul B Mahol onemda at gmail.com
Wed Mar 27 21:08:37 CET 2013

On 3/27/13, Brad O'Hearne <brado at bighillsoftware.com> wrote:
> On Mar 27, 2013, at 4:44 AM, Rene J.V. Bertin <rjvbertin at gmail.com> wrote:
>> So if the QTSampleBuffers contain non-native endian data the
>> FFmpeg-encoded output will inevitably be the "wrong way around" unless
>> it is converted before being encoded. Not?
> Thanks for the replies everyone, as they all raise new ideas. I want to
> answer a few questions, and then revisit the situation asking a few further
> questions....perhaps we can get closer.
> First, regarding endian-ness. QTKit is using native-endian (which in OS X on
> Intel is little-endian), but more than that, I'm explicitly checking
> endian-ness on the sample buffer, and it is little endian. This is the
> reason I was asking if there was a function in FFmpeg which would output the
> endian-ness explicitly, just so I could verify that. But with what
> information I have, it appears we are going from little-endian to
> little-endian, so endian-ness shouldn't be an issue.
> Second, Paul said:
>> Provide raw output given by QTKit whatever, and I'm sure someone will
>> give you solution for your problem.
> I'll take you up on that. You'll have to give me a little bit to create
> this, but I'm going to provide two files, the first just the raw bytes of a
> QTKit sample buffer, and the second, a compressed FLV file created by FFmpeg
> containing a short encoded audio stream so that you can see what I'm working
> with. I'll post that later today.
> Finally, I want to revisit the scenario one more time with a few
> questions...maybe there's something in there that will turn on a light bulb
> (my own) somewhere. So here goes:
> When creating an AVOutputFormatContext using av_guess_format passing it an
> extension of "flv" and a MIME type of "video/x-flv" configures the context
> with the "adpcm_swf"  audio codec, which requires a sample format of
> AV_SAMPLE_FMT_S16. The QTSampleBuffer format being captured from QTKIt is as
> follows:
> Linear PCM, 32 bit little-endian floating point, 2 channels, 44100 Hz
> This format would appear to map to the FFMpeg sample format of
> AV_SAMPLE_FMT_FLT. However, there's a difference in how QTKit is delivering
> the sample buffer data -- it isn't interleaved. In other words, channel 1
> samples come before all channel 2 samples. So I then interleave this data
> (you can see this in the QTFFAVStreamer streamAudioFrame method of my sample
> app) to put it into AV_SAMPLE_FMT_FLT, prior to attempting any resampling,
> so that the resampling converts from AV_SAMPLE_FMT_FLT to AV_SAMPLE_FMT_S16.

Than use AV_SAMPLE_FMT_FLTP, you do not need to manually interleave samples.
Each channel samples are put into separate frame->data[X] where X is channel
number starting from 0.

> There's a very similar handling example I was referred to a while back by
> the QuickTime API mailing list which does this, you can see that here:
> http://git.videolan.org/?p=vlc.git;a=blob;f=modules/access/qtsound.m;h=4ff12309927591b749e40ccca9227fe6ba293711;hb=74a3b3f19f3f15843e913ce347c237eb23375f6f
> Unfortunately, it doesn't proceed with resampling or encoding with FFmpeg,
> so that's as far as I can follow the example. So if I understand the
> resampling process, here is what should be happening:
> decompressed audio samples in AV_SAMPLE_FMT_FLT -> [FFmpeg resample] ->
> decompressed audio samples in AV_SAMPLE_FMT_S16 -> [FFmpeg encoding] -> FLV
> file
> If there's any part of that which is inaccurate, please let me know.
> However, assuming that is accurate, I'm wondering if the resampling step is
> the problem, specifically the conversion of floats to signed 16-bits. I
> could perform the resampling manually, if I knew exactly how that conversion
> is occurring. This raises a couple of decent questions:
> 1. Regarding sample formats, what is the difference between
> AV_SAMPLE_FMT_S32 and AV_SAMPLE_FMT_FLT? Both are signed, both are 32
> bits...?
> 2. How is a 32 bit float being converted to signed 16 bits? Once I know
> this, I'll write this manually and eliminate that from the equation too.
> 3. I have posted another message to the mailing list which hasn't been
> responded to, but I had several questions about packed samples and the align
> parameter in several libswresample function calls. In reading through the
> resampling_audio.c example, it wasn't clear to me the setting of this
> parameter to 0 vs. 1. I'll bump this message again in hopes of directing
> dialog on that topic there.
> Thanks again for all the discussion and help. I'll get those files posted
> later today, but in the meantime, the answers to the above questions would
> really help.
> Cheers,
> Brad
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