virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
Code Issues Releases Activity

swr: general doxy text about swr and example code.

Browse Source 
Based on doxy from avr
Reviewed-by: Clément Bœsch
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-11-19 23:46:34 +01:00
parent 4293464705
commit 7cca237ddd
1 changed files with 72 additions and 2 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

74
libswresample/swresample.h
View File

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