Commit 01b76019 authored by Anton Khirnov's avatar Anton Khirnov

lavr: add general API usage doxy

Signed-off-by: default avatarAnton Khirnov <anton@khirnov.net>
parent bff5e5f8
......@@ -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 */
......@@ -39,6 +39,7 @@
* @li @ref libavf "libavformat" I/O and muxing/demuxing library
* @li @ref lavd "libavdevice" special devices muxing/demuxing library
* @li @ref lavu "libavutil" common utility library
* @li @ref lavr "libavresample" audio resampling, format conversion and mixing
* @li @subpage libswscale color conversion and scaling library
*/
......
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