mirror of https://git.ffmpeg.org/ffmpeg.git
swresample: improve Doxygen introduction
Signed-off-by: Timothy Gu <timothygu99@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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@ -38,10 +38,16 @@
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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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* The first thing you will need to do in order to use lswr is to allocate
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* SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts(). If you
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* are using the former, you must set options through the @ref avoptions API.
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* The latter function provides the same feature, but it allows you to set some
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* common options in the same statement.
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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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* matrix). This is using the swr_alloc() function.
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* @code
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* SwrContext *swr = swr_alloc();
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* av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
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@ -52,10 +58,24 @@
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* av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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* @endcode
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*
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* The same job can be done using swr_alloc_set_opts() as well:
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* @code
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* SwrContext *swr = swr_alloc_set_opts(NULL, // we're allocating a new context
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* AV_CH_LAYOUT_STEREO, // out_ch_layout
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* AV_SAMPLE_FMT_S16, // out_sample_fmt
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* 44100, // out_sample_rate
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* AV_CH_LAYOUT_5POINT1, // in_ch_layout
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* AV_SAMPLE_FMT_FLTP, // in_sample_fmt
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* 48000, // in_sample_rate
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* 0, // log_offset
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* NULL); // log_ctx
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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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* using @ref AVOptions, as described above in the first example; or by using
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* swr_alloc_set_opts(), but with the first argument the allocated context.
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* You must then call swr_init() 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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@ -65,6 +85,10 @@
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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 samples used in the conversion process can be managed with the libavutil
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* @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc()
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* function used in the following example.
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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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@ -89,6 +113,9 @@
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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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* A swr_close() function is also available, but it exists mainly for
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* compatibility with libavresample, and is not required to be called.
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*
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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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