410 lines
16 KiB
C
410 lines
16 KiB
C
/*
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVRESAMPLE_AVRESAMPLE_H
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#define AVRESAMPLE_AVRESAMPLE_H
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/**
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* @file
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* @ingroup lavr
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* external API header
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*/
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/**
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* @defgroup lavr Libavresample
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* @{
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*
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* Libavresample (lavr) 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 lavr is done through AVAudioResampleContext, which is
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* allocated with avresample_alloc_context(). 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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* AVAudioResampleContext *avr = avresample_alloc_context();
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* av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
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* av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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* av_opt_set_int(avr, "in_sample_rate", 48000, 0);
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* av_opt_set_int(avr, "out_sample_rate", 44100, 0);
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* av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
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* av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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* @endcode
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*
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* Once the context is initialized, it must be opened with avresample_open(). If
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* you need to change the conversion parameters, you must close the context with
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* avresample_close(), change the parameters as described above, then reopen it
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* again.
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*
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* The conversion itself is done by repeatedly calling avresample_convert().
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* Note that the samples may get buffered in two places in lavr. The first one
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* is the output FIFO, where the samples end up if the output buffer is not
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* large enough. The data stored in there may be retrieved at any time with
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* avresample_read(). The second place is the resampling delay buffer,
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* applicable only when resampling is done. The samples in it require more input
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* before they can be processed. Their current amount is returned by
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* avresample_get_delay(). At the end of conversion the resampling buffer can be
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* flushed by calling avresample_convert() with NULL input.
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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_linesize, in_samples;
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*
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* while (get_input(&input, &in_linesize, &in_samples)) {
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* uint8_t *output
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* int out_linesize;
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* int out_samples = avresample_available(avr) +
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* av_rescale_rnd(avresample_get_delay(avr) +
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* in_samples, 44100, 48000, AV_ROUND_UP);
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* av_samples_alloc(&output, &out_linesize, 2, out_samples,
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* AV_SAMPLE_FMT_S16, 0);
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* out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
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* input, in_linesize, in_samples);
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* handle_output(output, out_linesize, 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 and the FIFOs are flushed if required, the
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* conversion context and everything associated with it must be freed with
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* avresample_free().
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*/
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#include "libavutil/avutil.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/dict.h"
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#include "libavutil/log.h"
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#include "libavresample/version.h"
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#define AVRESAMPLE_MAX_CHANNELS 32
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typedef struct AVAudioResampleContext AVAudioResampleContext;
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/** Mixing Coefficient Types */
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enum AVMixCoeffType {
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AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
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AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
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AV_MIX_COEFF_TYPE_FLT, /** floating-point */
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AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
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};
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/** Resampling Filter Types */
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enum AVResampleFilterType {
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AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
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AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
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AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
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};
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enum AVResampleDitherMethod {
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AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */
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AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */
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AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/
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AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */
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AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */
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AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */
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};
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/**
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* Return the LIBAVRESAMPLE_VERSION_INT constant.
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*/
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unsigned avresample_version(void);
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/**
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* Return the libavresample build-time configuration.
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* @return configure string
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*/
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const char *avresample_configuration(void);
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/**
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* Return the libavresample license.
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*/
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const char *avresample_license(void);
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/**
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* Get the AVClass for AVAudioResampleContext.
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*
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* Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
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* without allocating a context.
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*
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* @see av_opt_find().
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*
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* @return AVClass for AVAudioResampleContext
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*/
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const AVClass *avresample_get_class(void);
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/**
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* Allocate AVAudioResampleContext and set options.
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*
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* @return allocated audio resample context, or NULL on failure
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*/
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AVAudioResampleContext *avresample_alloc_context(void);
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/**
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* Initialize AVAudioResampleContext.
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*
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* @param avr audio resample context
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_open(AVAudioResampleContext *avr);
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/**
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* Close AVAudioResampleContext.
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*
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* This closes the context, but it does not change the parameters. The context
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* can be reopened with avresample_open(). It does, however, clear the output
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* FIFO and any remaining leftover samples in the resampling delay buffer. If
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* there was a custom matrix being used, that is also cleared.
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*
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* @see avresample_convert()
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* @see avresample_set_matrix()
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*
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* @param avr audio resample context
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*/
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void avresample_close(AVAudioResampleContext *avr);
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/**
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* Free AVAudioResampleContext and associated AVOption values.
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*
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* This also calls avresample_close() before freeing.
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*
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* @param avr audio resample context
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*/
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void avresample_free(AVAudioResampleContext **avr);
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/**
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* Generate a channel mixing matrix.
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*
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* This function is the one used internally by libavresample for building the
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* default mixing matrix. It is made public just as a utility function for
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* building custom matrices.
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*
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* @param in_layout input channel layout
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* @param out_layout output channel layout
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* @param center_mix_level mix level for the center channel
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* @param surround_mix_level mix level for the surround channel(s)
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* @param lfe_mix_level mix level for the low-frequency effects channel
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* @param normalize if 1, coefficients will be normalized to prevent
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* overflow. if 0, coefficients will not be
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* normalized.
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* @param[out] matrix mixing coefficients; matrix[i + stride * o] is
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* the weight of input channel i in output channel o.
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* @param stride distance between adjacent input channels in the
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* matrix array
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* @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
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double center_mix_level, double surround_mix_level,
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double lfe_mix_level, int normalize, double *matrix,
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int stride, enum AVMatrixEncoding matrix_encoding);
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/**
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* Get the current channel mixing matrix.
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*
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* If no custom matrix has been previously set or the AVAudioResampleContext is
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* not open, an error is returned.
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*
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* @param avr audio resample context
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* @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
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* input channel i in output channel o.
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* @param stride distance between adjacent input channels in the matrix array
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
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int stride);
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/**
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* Set channel mixing matrix.
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*
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* Allows for setting a custom mixing matrix, overriding the default matrix
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* generated internally during avresample_open(). This function can be called
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* anytime on an allocated context, either before or after calling
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* avresample_open(), as long as the channel layouts have been set.
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* avresample_convert() always uses the current matrix.
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* Calling avresample_close() on the context will clear the current matrix.
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*
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* @see avresample_close()
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*
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* @param avr audio resample context
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* @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
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* input channel i in output channel o.
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* @param stride distance between adjacent input channels in the matrix array
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
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int stride);
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/**
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* Set a customized input channel mapping.
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*
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* This function can only be called when the allocated context is not open.
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* Also, the input channel layout must have already been set.
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*
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* Calling avresample_close() on the context will clear the channel mapping.
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*
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* The map for each input channel specifies the channel index in the source to
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* use for that particular channel, or -1 to mute the channel. Source channels
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* can be duplicated by using the same index for multiple input channels.
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*
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* Examples:
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*
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* Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):
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* { 1, 2, 0, 5, 3, 4 }
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*
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* Muting the 3rd channel in 4-channel input:
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* { 0, 1, -1, 3 }
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*
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* Duplicating the left channel of stereo input:
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* { 0, 0 }
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*
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* @param avr audio resample context
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* @param channel_map customized input channel mapping
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_set_channel_mapping(AVAudioResampleContext *avr,
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const int *channel_map);
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/**
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* Set compensation for resampling.
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*
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* This can be called anytime after avresample_open(). If resampling is not
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* automatically enabled because of a sample rate conversion, the
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* "force_resampling" option must have been set to 1 when opening the context
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* in order to use resampling compensation.
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*
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* @param avr audio resample context
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* @param sample_delta compensation delta, in samples
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* @param compensation_distance compensation distance, in samples
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* @return 0 on success, negative AVERROR code on failure
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*/
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int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
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int compensation_distance);
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/**
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* Convert input samples and write them to the output FIFO.
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*
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* The upper bound on the number of output samples is given by
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* avresample_available() + (avresample_get_delay() + number of input samples) *
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* output sample rate / input sample rate.
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*
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* The output data can be NULL or have fewer allocated samples than required.
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* In this case, any remaining samples not written to the output will be added
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* to an internal FIFO buffer, to be returned at the next call to this function
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* or to avresample_read().
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*
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* If converting sample rate, there may be data remaining in the internal
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* resampling delay buffer. avresample_get_delay() tells the number of remaining
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* samples. To get this data as output, call avresample_convert() with NULL
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* input.
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*
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* At the end of the conversion process, there may be data remaining in the
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* internal FIFO buffer. avresample_available() tells the number of remaining
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* samples. To get this data as output, either call avresample_convert() with
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* NULL input or call avresample_read().
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*
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* @see avresample_available()
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* @see avresample_read()
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* @see avresample_get_delay()
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*
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* @param avr audio resample context
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* @param output output data pointers
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* @param out_plane_size output plane size, in bytes.
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* This can be 0 if unknown, but that will lead to
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* optimized functions not being used directly on the
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* output, which could slow down some conversions.
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* @param out_samples maximum number of samples that the output buffer can hold
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* @param input input data pointers
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* @param in_plane_size input plane size, in bytes
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* This can be 0 if unknown, but that will lead to
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* optimized functions not being used directly on the
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* input, which could slow down some conversions.
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* @param in_samples number of input samples to convert
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* @return number of samples written to the output buffer,
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* not including converted samples added to the internal
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* output FIFO
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*/
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int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
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int out_plane_size, int out_samples, uint8_t **input,
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int in_plane_size, int in_samples);
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/**
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* Return the number of samples currently in the resampling delay buffer.
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*
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* When resampling, there may be a delay between the input and output. Any
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* unconverted samples in each call are stored internally in a delay buffer.
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* This function allows the user to determine the current number of samples in
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* the delay buffer, which can be useful for synchronization.
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*
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* @see avresample_convert()
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*
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* @param avr audio resample context
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* @return number of samples currently in the resampling delay buffer
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*/
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int avresample_get_delay(AVAudioResampleContext *avr);
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/**
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* Return the number of available samples in the output FIFO.
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*
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* During conversion, if the user does not specify an output buffer or
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* specifies an output buffer that is smaller than what is needed, remaining
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* samples that are not written to the output are stored to an internal FIFO
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* buffer. The samples in the FIFO can be read with avresample_read() or
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* avresample_convert().
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*
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* @see avresample_read()
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* @see avresample_convert()
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*
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* @param avr audio resample context
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* @return number of samples available for reading
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*/
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int avresample_available(AVAudioResampleContext *avr);
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/**
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* Read samples from the output FIFO.
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*
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* During conversion, if the user does not specify an output buffer or
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* specifies an output buffer that is smaller than what is needed, remaining
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* samples that are not written to the output are stored to an internal FIFO
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* buffer. This function can be used to read samples from that internal FIFO.
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*
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* @see avresample_available()
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* @see avresample_convert()
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*
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* @param avr audio resample context
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* @param output output data pointers. May be NULL, in which case
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* nb_samples of data is discarded from output FIFO.
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* @param nb_samples number of samples to read from the FIFO
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* @return the number of samples written to output
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*/
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int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
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/**
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* @}
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*/
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#endif /* AVRESAMPLE_AVRESAMPLE_H */
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