[Libav-user] Getting spikes in signal when reading mp3 data

[Mark McKay]

Just to add more data, this is an example of what I'm reading from my
mp3 file.  While many of the values appear to be correct (and a graph
I've made of the data appears to mostly match what I'm getting from the
same audio sample in .wav format), I'm getting a lot of byte sequences
that do not belong there - especially 0xeefeeefe. This data is taken
from channel 0, stepped by every 64 samples: 

f4 8f 94 bd : -0.072540 

a4 61 69 bd : -0.056978 

15 8e 77 3d : 0.060438 

07 31 32 be : -0.174015 

af c4 92 3d : 0.071664 

c0 dc 45 3d : 0.048306 

14 ee 38 bd : -0.045149 

67 88 a9 3e : 0.331119 

c0 04 c4 2f : 0.000000 

ab ab ab ab : -0.000000 

c5 76 7e 75 : 322571370690696392781073674665984.000000 

5c d4 fb 09 : 0.000000 

ee fe ee fe : -158839966806181746533156862930074992640.000000 

7b cb 3c 91 : -0.000000 

ee fe ee fe : -158839966806181746533156862930074992640.000000 

65 55 7e 6f : 78712428148303660770110996480.000000 

30 ef fc 6e : 39139691209729015717184929792.000000 

0e 2e 00 be : -0.125176 

19 8d 08 bb : -0.002084 

89 1f da bd : -0.106505 

70 c3 23 bb : -0.002499 

0e 2e 28 3d : 0.041060 

2d d3 28 bd : -0.041217 

c8 6c 30 be : -0.172290 

61 0f 17 3e : 0.147520 

75 a0 15 3d : 0.036530 

ee fe ee fe : -158839966806181746533156862930074992640.000000 

ee fe ee fe : -158839966806181746533156862930074992640.000000 

On 2019-12-03 07:56, Mark McKay wrote:

> I'm trying to use libav to read in audio PCM data.  I've got it to read in my .wav sample data correctly (the byte format is AV_SAMPLE_FMT_S16). However, my .mp3 sample is getting a lot of strange spikes in the data.  The sample format for the .mp3 is AV_SAMPLE_FMT_FLTP and while most of the samples appear to be correct, about 10% of them have very large magnitudes.  I'm wondering what's going wrong, and if this might have something to do with the exponent of the floating point number being translated incorrectly.  It's not clear to me what the difference between AV_SAMPLE_FMT_FLT and AV_SAMPLE_FMT_FLTP is.
> 
> This is a sample of the bit of my code that reads the data from the frame.  Am I reading it correctly? 
> 
> int bufferSize = av_samples_get_buffer_size(NULL,
> 
> aCodecCtx->channels,
> 
> aFrame->nb_samples,
> 
> aCodecCtx->sample_fmt,
> 
> 1);
> 
> uint8_t *dataBuf = aFrame->data[0];
> 
> for (int i = 0; i < aFrame->nb_samples; i++)
> 
> {
> 
> for (int ch = 0; ch < numChannels; ch++)
> 
> {
> 
> float val = 0;
> 
> switch (aCodecCtx->sample_fmt)
> 
> {
> ...
> 
> case AV_SAMPLE_FMT_FLT:
> 
> case AV_SAMPLE_FMT_FLTP:
> 
> {
> 
> val = ((float*)dataBuf)[ch + i * numChannels];
> 
> if (val > 1 || val < -1)
> 
> {
> 
> qDebug("Sample out of range %f", val);
> 
> val = 0;
> 
> }
> 
> break;
> 
> }
> 
> ---
> http://www.kitfox.com 
> 
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