mirror of https://git.ffmpeg.org/ffmpeg.git
221 lines
13 KiB
C
221 lines
13 KiB
C
/*
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* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
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*
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* This file is part of libswresample
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*
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* libswresample 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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* libswresample 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 libswresample; 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 SWRESAMPLE_SWRESAMPLE_INTERNAL_H
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#define SWRESAMPLE_SWRESAMPLE_INTERNAL_H
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#include "swresample.h"
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#include "libavutil/channel_layout.h"
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#include "config.h"
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#define SWR_CH_MAX 64
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#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
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#define NS_TAPS 20
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#if ARCH_X86_64
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typedef int64_t integer;
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#else
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typedef int integer;
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#endif
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typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
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typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
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typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
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typedef struct AudioData{
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uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
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uint8_t *data; ///< samples buffer
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int ch_count; ///< number of channels
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int bps; ///< bytes per sample
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int count; ///< number of samples
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int planar; ///< 1 if planar audio, 0 otherwise
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enum AVSampleFormat fmt; ///< sample format
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} AudioData;
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struct DitherContext {
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int method;
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int noise_pos;
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float scale;
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float noise_scale; ///< Noise scale
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int ns_taps; ///< Noise shaping dither taps
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float ns_scale; ///< Noise shaping dither scale
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float ns_scale_1; ///< Noise shaping dither scale^-1
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int ns_pos; ///< Noise shaping dither position
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float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients
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float ns_errors[SWR_CH_MAX][2*NS_TAPS];
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AudioData noise; ///< noise used for dithering
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AudioData temp; ///< temporary storage when writing into the input buffer isn't possible
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int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly
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};
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typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
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double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational);
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typedef void (* resample_free_func)(struct ResampleContext **c);
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typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
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typedef int (* resample_flush_func)(struct SwrContext *c);
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typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
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typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
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typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
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typedef int64_t (* get_out_samples_func)(struct SwrContext *s, int in_samples);
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struct Resampler {
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resample_init_func init;
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resample_free_func free;
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multiple_resample_func multiple_resample;
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resample_flush_func flush;
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set_compensation_func set_compensation;
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get_delay_func get_delay;
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invert_initial_buffer_func invert_initial_buffer;
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get_out_samples_func get_out_samples;
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};
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extern struct Resampler const swri_resampler;
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extern struct Resampler const swri_soxr_resampler;
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struct SwrContext {
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const AVClass *av_class; ///< AVClass used for AVOption and av_log()
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int log_level_offset; ///< logging level offset
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void *log_ctx; ///< parent logging context
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enum AVSampleFormat in_sample_fmt; ///< input sample format
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enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
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enum AVSampleFormat out_sample_fmt; ///< output sample format
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AVChannelLayout used_ch_layout; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
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AVChannelLayout in_ch_layout; ///< input channel layout
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AVChannelLayout out_ch_layout; ///< output channel layout
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int in_sample_rate; ///< input sample rate
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int out_sample_rate; ///< output sample rate
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int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
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float slev; ///< surround mixing level
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float clev; ///< center mixing level
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float lfe_mix_level; ///< LFE mixing level
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float rematrix_volume; ///< rematrixing volume coefficient
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float rematrix_maxval; ///< maximum value for rematrixing output
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int matrix_encoding; /**< matrixed stereo encoding */
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const int *channel_map; ///< channel index (or -1 if muted channel) map
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int engine;
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AVChannelLayout user_used_chlayout; ///< User set used channel layout
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AVChannelLayout user_in_chlayout; ///< User set input channel layout
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AVChannelLayout user_out_chlayout; ///< User set output channel layout
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enum AVSampleFormat user_int_sample_fmt; ///< User set internal sample format
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int user_dither_method; ///< User set dither method
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struct DitherContext dither;
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int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
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int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
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int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
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int exact_rational; /**< if 1 then enable non power of 2 phase_count */
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double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
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int filter_type; /**< swr resampling filter type */
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double kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
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double precision; /**< soxr resampling precision (in bits) */
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int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
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float min_compensation; ///< swr minimum below which no compensation will happen
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float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
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float soft_compensation_duration; ///< swr duration over which soft compensation is applied
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float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
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float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
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int64_t firstpts_in_samples; ///< swr first pts in samples
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int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
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int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
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int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
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AudioData in; ///< input audio data
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AudioData postin; ///< post-input audio data: used for rematrix/resample
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AudioData midbuf; ///< intermediate audio data (postin/preout)
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AudioData preout; ///< pre-output audio data: used for rematrix/resample
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AudioData out; ///< converted output audio data
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AudioData in_buffer; ///< cached audio data (convert and resample purpose)
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AudioData silence; ///< temporary with silence
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AudioData drop_temp; ///< temporary used to discard output
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int in_buffer_index; ///< cached buffer position
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int in_buffer_count; ///< cached buffer length
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int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
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int flushed; ///< 1 if data is to be flushed and no further input is expected
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int64_t outpts; ///< output PTS
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int64_t firstpts; ///< first PTS
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int drop_output; ///< number of output samples to drop
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double delayed_samples_fixup; ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
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struct AudioConvert *in_convert; ///< input conversion context
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struct AudioConvert *out_convert; ///< output conversion context
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struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
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struct ResampleContext *resample; ///< resampling context
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struct Resampler const *resampler; ///< resampler virtual function table
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double matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
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float matrix_flt[SWR_CH_MAX][SWR_CH_MAX]; ///< single precision floating point rematrixing coefficients
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uint8_t *native_matrix;
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uint8_t *native_one;
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uint8_t *native_simd_one;
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uint8_t *native_simd_matrix;
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int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
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uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
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mix_1_1_func_type *mix_1_1_f;
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mix_1_1_func_type *mix_1_1_simd;
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mix_2_1_func_type *mix_2_1_f;
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mix_2_1_func_type *mix_2_1_simd;
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mix_any_func_type *mix_any_f;
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/* TODO: callbacks for ASM optimizations */
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};
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av_warn_unused_result
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int swri_realloc_audio(AudioData *a, int count);
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void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
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void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
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void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
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void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
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av_warn_unused_result
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int swri_rematrix_init(SwrContext *s);
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void swri_rematrix_free(SwrContext *s);
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int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
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int swri_rematrix_init_x86(struct SwrContext *s);
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av_warn_unused_result
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int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
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av_warn_unused_result
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int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
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void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
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enum AVSampleFormat out_fmt,
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enum AVSampleFormat in_fmt,
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int channels);
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void swri_audio_convert_init_arm(struct AudioConvert *ac,
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enum AVSampleFormat out_fmt,
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enum AVSampleFormat in_fmt,
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int channels);
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void swri_audio_convert_init_x86(struct AudioConvert *ac,
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enum AVSampleFormat out_fmt,
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enum AVSampleFormat in_fmt,
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int channels);
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#endif
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