From 01b760190d32550683d7c790309acadea3fe0820 Mon Sep 17 00:00:00 2001 From: Anton Khirnov Date: Sun, 28 Oct 2012 22:52:54 +0100 Subject: lavr: add general API usage doxy Signed-off-by: Anton Khirnov --- libavresample/avresample.h | 71 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 71 insertions(+) (limited to 'libavresample/avresample.h') diff --git a/libavresample/avresample.h b/libavresample/avresample.h index ea93952..87134b3 100644 --- a/libavresample/avresample.h +++ b/libavresample/avresample.h @@ -23,9 +23,76 @@ /** * @file + * @ingroup lavr * external API header */ +/** + * @defgroup lavr Libavresample + * @{ + * + * Libavresample (lavr) is a library that handles audio resampling, sample + * format conversion and mixing. + * + * Interaction with lavr is done through AVAudioResampleContext, which is + * allocated with avresample_alloc_context(). It is opaque, so all parameters + * must be set with the @ref avoptions API. + * + * For example the following code will setup conversion from planar float sample + * format to interleaved signed 16-bit integer, downsampling from 48kHz to + * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing + * matrix): + * @code + * AVAudioResampleContext *avr = avresample_alloc_context(); + * av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0); + * av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0); + * av_opt_set_int(avr, "in_sample_rate", 48000, 0); + * av_opt_set_int(avr, "out_sample_rate", 44100, 0); + * av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); + * av_opt_set_int(avr, "out_sample_fmt, AV_SAMPLE_FMT_S16, 0); + * @endcode + * + * Once the context is initialized, it must be opened with avresample_open(). If + * you need to change the conversion parameters, you must close the context with + * avresample_close(), change the parameters as described above, then reopen it + * again. + * + * The conversion itself is done by repeatedly calling avresample_convert(). + * Note that the samples may get buffered in two places in lavr. The first one + * is the output FIFO, where the samples end up if the output buffer is not + * large enough. The data stored in there may be retrieved at any time with + * avresample_read(). The second place is the resampling delay buffer, + * applicable only when resampling is done. The samples in it require more input + * before they can be processed. Their current amount is returned by + * avresample_get_delay(). At the end of conversion the resampling buffer can be + * flushed by calling avresample_convert() with NULL input. + * + * The following code demonstrates the conversion loop assuming the parameters + * from above and caller-defined functions get_input() and handle_output(): + * @code + * uint8_t **input; + * int in_linesize, in_samples; + * + * while (get_input(&input, &in_linesize, &in_samples)) { + * uint8_t *output + * int out_linesize; + * int out_samples = avresample_available(avr) + + * av_rescale_rnd(avresample_get_delay(avr) + + * in_samples, 44100, 48000, AV_ROUND_UP); + * av_samples_alloc(&output, &out_linesize, 2, out_samples, + * AV_SAMPLE_FMT_S16, 0); + * out_samples = avresample_convert(avr, &output, out_linesize, out_samples, + * input, in_linesize, in_samples); + * handle_output(output, out_linesize, out_samples); + * av_freep(&output); + * } + * @endcode + * + * When the conversion is finished and the FIFOs are flushed if required, the + * conversion context and everything associated with it must be freed with + * avresample_free(). + */ + #include "libavutil/audioconvert.h" #include "libavutil/avutil.h" #include "libavutil/dict.h" @@ -289,4 +356,8 @@ int avresample_available(AVAudioResampleContext *avr); */ int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples); +/** + * @} + */ + #endif /* AVRESAMPLE_AVRESAMPLE_H */ -- cgit v1.1