/**
* @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_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
};
+enum AVResampleDitherMethod {
+ AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */
+ AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */
+ AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/
+ AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */
+ AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */
+ AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */
+};
+
/**
* Return the LIBAVRESAMPLE_VERSION_INT constant.
*/
/**
* Get the current channel mixing matrix.
*
+ * If no custom matrix has been previously set or the AVAudioResampleContext is
+ * not open, an error is returned.
+ *
* @param avr audio resample context
* @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
* input channel i in output channel o.
* Allows for setting a custom mixing matrix, overriding the default matrix
* generated internally during avresample_open(). This function can be called
* anytime on an allocated context, either before or after calling
- * avresample_open(). avresample_convert() always uses the current matrix.
+ * avresample_open(), as long as the channel layouts have been set.
+ * avresample_convert() always uses the current matrix.
* Calling avresample_close() on the context will clear the current matrix.
*
* @see avresample_close()
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
int stride);
+/**
+ * Set a customized input channel mapping.
+ *
+ * This function can only be called when the allocated context is not open.
+ * Also, the input channel layout must have already been set.
+ *
+ * Calling avresample_close() on the context will clear the channel mapping.
+ *
+ * The map for each input channel specifies the channel index in the source to
+ * use for that particular channel, or -1 to mute the channel. Source channels
+ * can be duplicated by using the same index for multiple input channels.
+ *
+ * Examples:
+ *
+ * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to Libav order (L,R,C,LFE,Ls,Rs):
+ * { 1, 2, 0, 5, 3, 4 }
+ *
+ * Muting the 3rd channel in 4-channel input:
+ * { 0, 1, -1, 3 }
+ *
+ * Duplicating the left channel of stereo input:
+ * { 0, 0 }
+ *
+ * @param avr audio resample context
+ * @param channel_map customized input channel mapping
+ * @return 0 on success, negative AVERROR code on failure
+ */
+int avresample_set_channel_mapping(AVAudioResampleContext *avr,
+ const int *channel_map);
+
/**
* Set compensation for resampling.
*
- * This can be called anytime after avresample_open(). If resampling was not
- * being done previously, the AVAudioResampleContext is closed and reopened
- * with resampling enabled. In this case, any samples remaining in the output
- * FIFO and the current channel mixing matrix will be restored after reopening
- * the context.
+ * This can be called anytime after avresample_open(). If resampling is not
+ * automatically enabled because of a sample rate conversion, the
+ * "force_resampling" option must have been set to 1 when opening the context
+ * in order to use resampling compensation.
*
* @param avr audio resample context
* @param sample_delta compensation delta, in samples
/**
* 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);
/**
* @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