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lavr: add general API usage doxy
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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@ -23,9 +23,76 @@
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/**
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* @file
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* @ingroup lavr
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* external API header
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*/
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/**
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* @defgroup lavr Libavresample
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* @{
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*
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* Libavresample (lavr) is a library that handles audio resampling, sample
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* format conversion and mixing.
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*
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* Interaction with lavr is done through AVAudioResampleContext, which is
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* allocated with avresample_alloc_context(). It is opaque, so all parameters
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* must be set with the @ref avoptions API.
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*
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* For example the following code will setup conversion from planar float sample
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* format to interleaved signed 16-bit integer, downsampling from 48kHz to
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* 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
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* matrix):
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* @code
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* AVAudioResampleContext *avr = avresample_alloc_context();
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* av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
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* av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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* av_opt_set_int(avr, "in_sample_rate", 48000, 0);
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* av_opt_set_int(avr, "out_sample_rate", 44100, 0);
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* av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
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* av_opt_set_int(avr, "out_sample_fmt, AV_SAMPLE_FMT_S16, 0);
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* @endcode
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*
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* Once the context is initialized, it must be opened with avresample_open(). If
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* you need to change the conversion parameters, you must close the context with
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* avresample_close(), change the parameters as described above, then reopen it
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* again.
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*
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* The conversion itself is done by repeatedly calling avresample_convert().
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* Note that the samples may get buffered in two places in lavr. The first one
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* is the output FIFO, where the samples end up if the output buffer is not
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* large enough. The data stored in there may be retrieved at any time with
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* avresample_read(). The second place is the resampling delay buffer,
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* applicable only when resampling is done. The samples in it require more input
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* before they can be processed. Their current amount is returned by
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* avresample_get_delay(). At the end of conversion the resampling buffer can be
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* flushed by calling avresample_convert() with NULL input.
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*
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* The following code demonstrates the conversion loop assuming the parameters
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* from above and caller-defined functions get_input() and handle_output():
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* @code
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* uint8_t **input;
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* int in_linesize, in_samples;
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*
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* while (get_input(&input, &in_linesize, &in_samples)) {
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* uint8_t *output
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* int out_linesize;
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* int out_samples = avresample_available(avr) +
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* av_rescale_rnd(avresample_get_delay(avr) +
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* in_samples, 44100, 48000, AV_ROUND_UP);
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* av_samples_alloc(&output, &out_linesize, 2, out_samples,
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* AV_SAMPLE_FMT_S16, 0);
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* out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
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* input, in_linesize, in_samples);
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* handle_output(output, out_linesize, out_samples);
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* av_freep(&output);
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* }
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* @endcode
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*
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* When the conversion is finished and the FIFOs are flushed if required, the
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* conversion context and everything associated with it must be freed with
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* avresample_free().
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/avutil.h"
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#include "libavutil/dict.h"
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@ -289,4 +356,8 @@ int avresample_available(AVAudioResampleContext *avr);
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*/
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int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
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/**
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* @}
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*/
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#endif /* AVRESAMPLE_AVRESAMPLE_H */
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@ -39,6 +39,7 @@
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* @li @ref libavf "libavformat" I/O and muxing/demuxing library
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* @li @ref lavd "libavdevice" special devices muxing/demuxing library
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* @li @ref lavu "libavutil" common utility library
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* @li @ref lavr "libavresample" audio resampling, format conversion and mixing
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* @li @subpage libswscale color conversion and scaling library
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*/
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