/**
* @file
+ * @ingroup lavr
* external API header
*/
-#include "libavutil/audioconvert.h"
+/**
+ * @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/avutil.h"
+#include "libavutil/channel_layout.h"
#include "libavutil/dict.h"
#include "libavutil/log.h"
AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
};
+/** Resampling Filter Types */
+enum AVResampleFilterType {
+ AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
+ AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
+ AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
+};
+
/**
* Return the LIBAVRESAMPLE_VERSION_INT constant.
*/
* the weight of input channel i in output channel o.
* @param stride distance between adjacent input channels in the
* matrix array
+ * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, int normalize, double *matrix,
- int stride);
+ int stride, enum AVMatrixEncoding matrix_encoding);
/**
* Get the current channel mixing matrix.
/**
* Convert input samples and write them to the output FIFO.
*
+ * The upper bound on the number of output samples is given by
+ * avresample_available() + (avresample_get_delay() + number of input samples) *
+ * output sample rate / input sample rate.
+ *
* The output data can be NULL or have fewer allocated samples than required.
* In this case, any remaining samples not written to the output will be added
* to an internal FIFO buffer, to be returned at the next call to this function
* not including converted samples added to the internal
* output FIFO
*/
-int avresample_convert(AVAudioResampleContext *avr, void **output,
- int out_plane_size, int out_samples, void **input,
+int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
+ int out_plane_size, int out_samples, uint8_t **input,
int in_plane_size, int in_samples);
/**
* @see avresample_convert()
*
* @param avr audio resample context
- * @param output output data pointers
+ * @param output output data pointers. May be NULL, in which case
+ * nb_samples of data is discarded from output FIFO.
* @param nb_samples number of samples to read from the FIFO
* @return the number of samples written to output
*/
-int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples);
+int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
+
+/**
+ * @}
+ */
#endif /* AVRESAMPLE_AVRESAMPLE_H */
projects / ffmpeg / blobdiff