mirror of https://git.ffmpeg.org/ffmpeg.git
swr: general doxy text about swr and example code.
Based on doxy from avr Reviewed-by: Clément Bœsch Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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@ -18,13 +18,79 @@
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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*/
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#ifndef SWRESAMPLE_SWRESAMPLE_H
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#define SWRESAMPLE_SWRESAMPLE_H
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/**
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/**
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* @file
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* @file
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* libswresample public header
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* libswresample public header
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*/
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*/
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#ifndef SWRESAMPLE_SWRESAMPLE_H
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/**
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#define SWRESAMPLE_SWRESAMPLE_H
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* @defgroup lswr Libswresample
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* @{
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*
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* Libswresample (lswr) 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 lswr is done through SwrContext, which is
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* allocated with swr_alloc() or swr_alloc_set_opts(). 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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* SwrContext *swr = swr_alloc();
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* av_opt_set_int(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
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* av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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* av_opt_set_int(swr, "in_sample_rate", 48000, 0);
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* av_opt_set_int(swr, "out_sample_rate", 44100, 0);
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* av_opt_set_int(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
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* av_opt_set_int(swr, "out_sample_fmt, AV_SAMPLE_FMT_S16, 0);
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* @endcode
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*
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* Once all values have been set, it must be initialized with swr_init(). If
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* you need to change the conversion parameters, you can change the parameters
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* as described above, or by using swr_alloc_set_opts(), then call swr_init()
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* again.
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*
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* The conversion itself is done by repeatedly calling swr_convert().
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* Note that the samples may get buffered in swr if you provide insufficient
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* output space or if sample rate conversion is done, which requires "future"
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* samples. Samples that do not require future input can be retrieved at any
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* time by using swr_convert() (in_count can be set to 0).
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* At the end of conversion the resampling buffer can be flushed by calling
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* swr_convert() with NULL in and 0 in_count.
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*
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* The delay between input and output, can at any time be found by using
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* swr_get_delay().
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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_samples;
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*
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* while (get_input(&input, &in_samples)) {
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* uint8_t *output
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* int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
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* in_samples, 44100, 48000, AV_ROUND_UP);
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* av_samples_alloc(&output, NULL, 2, out_samples,
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* AV_SAMPLE_FMT_S16, 0);
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* out_samples = swr_convert(swr, &output, out_samples,
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* input, in_samples);
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* handle_output(output, 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, the conversion
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* context and everything associated with it must be freed with swr_free().
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* There will be no memory leak if the data is not completely flushed before
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* swr_free().
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*/
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#include <inttypes.h>
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#include <inttypes.h>
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#include "libavutil/samplefmt.h"
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#include "libavutil/samplefmt.h"
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@ -217,4 +283,8 @@ const char *swresample_configuration(void);
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*/
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*/
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const char *swresample_license(void);
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const char *swresample_license(void);
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/**
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* @}
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*/
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#endif /* SWRESAMPLE_SWRESAMPLE_H */
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#endif /* SWRESAMPLE_SWRESAMPLE_H */
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