On Apr 28, 2013, at 12:22 PM, Bruce Wheaton <[email protected]> wrote:
> Why don't you just check the addresses of the data planes and know it? It's 2
> minutes to check a few times.
Bruce...thanks for the advice. The data in both channel 1 and channel 2 (and as
a whole), planar or not, and both in the captured sample buffer and in the
source data buffer is 32-byte aligned (has been all along). I tested both.
Beyond that, just for good measure, I went through the exercise of moving
sample values anyway from the capture buffer to the FFmpeg-allocated source
data buffer created with av_samples_alloc_array_and_samples -- so by definition
it should have been aligned (checked it still though)....same result, data
aligned just as before, but same distortion.
It appears however I've found the problem. First, after reading that aac is
apparently a dead codec, I dumped it and went to libmp3lame. Ran it, same
problem. Did some further debugging...revisited this. In avcodec.h, there is
this doc above the frame_rate variable:
* Number of samples per channel in an audio frame.
*
* - encoding: set by libavcodec in avcodec_open2(). Each submitted frame
* except the last must contain exactly frame_size samples per channel.
* May be 0 when the codec has CODEC_CAP_VARIABLE_FRAME_SIZE set, then the
* frame size is not restricted.
In creating the destination data buffer for resampling and the eventual frame
to be streamed, I'd been using the frame_size value computed by libavcodec in
avcodec_open2 as my destination number of samples (which the doc would suggest
is what should be done -- and in my journeys through code snippets and blogs
encountered in the series of tubes, it was suggested this was a proper
approach). I otherwise would not have thought anything of it had I not been
checking this computed value and noticed that libmp3lame results in a value of
1152, which was different than what I'd seen coming off other codecs. So I
ditched using the frame_size, went back to using the number of samples as
received by capture. Don't know what to make of the doc above. My frames
contain 512 samples, while the frame_size of the codec context is 1152.
Result, clear audio, both alone and combined with video, both written to a
local file and streamed. More tests are needed to verify consistently
problem-free function, but seems to work as desired.
Brad
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