On Mar 26, 2013, at 12:14 PM, Brad O'Hearne <[email protected]> wrote:
> Hello, > > I've noticed that several functions in samplefmt.h take an "align" parameter, > such as the function calls: > > int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, > int nb_samples, enum AVSampleFormat sample_fmt, int > align); > > int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, > enum AVSampleFormat sample_fmt, int align); > > int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, > const uint8_t *buf, > int nb_channels, int nb_samples, > enum AVSampleFormat sample_fmt, int align); > > This align parameter has the following description: > > * @param align buffer size alignment (0 = default, 1 = no > alignment) > > I want to make sure that I'm properly understanding the purpose and setting > of this parameter. As I understand it, a sample is "packed" if its sample > bits occupy the entire available bits for the channel. If a sample's bits do > not occupy the entire available bits for the channel it is not packed, and > then the data is either high or low-aligned within the channel. > > In the case of my app, my sample format of captured audio is: > > Linear PCM, 32 bit little-endian floating point, 2 channels, 44100 Hz > > and this data IS indeed packed, meaning that there is neither high nor low > alignment. In setting the appropriate align value for the aforementioned > functions, I have two questions: > > 1. What is "default" alignment according to the documentation? Is that high > or low, or something else? > > 2. Based on my captured sample data being packed, shouldn't this mean that > there is NO alignment, and therefore the value for these method invocations > be 1? This is essentially a "bump" given my reference to this issue in my post in another thread, but I'll add the detail that I revisited the resampling_audio.c example and I noticed that both 0 and 1 align parameter values are being used, and I wasn't completely clear as to why. If someone could speak to the questions I asked above, this would help to clear up the principle of the issue, plus it might have a bearing on the bigger audio problem I'm having. Thanks, Brad _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
