mirror of
https://github.com/FFmpeg/FFmpeg.git
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Kacper Michajłow
09f879b074
With a matrix, samples of more than 16 bits are mixed in floating point, which takes the lowest bits of s32. A matrix that only copies channels computes nothing, so select the internal format like without a matrix. Layouts that differ in the order of their channels or in unused ones get such a matrix. The matrix is built before the format is selected for this. The limit for its coefficients no longer looks at the selected format. For mixing that is an integer format only if the output has one or if it was set, which is what the limit looks at now. s64 still goes through double, the matrix has no functions for it. Bump the micro version, as this and the previous commits change which custom layouts can be used. Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
237 lines
14 KiB
C
237 lines
14 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 SQRT1_3 0.57735026918962576451 /* sqrt(1/3) */
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#define SQRT2_3 0.81649658092772603273 /* sqrt(2/3) */
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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, const 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, const void *coeffp, integer index1, integer index2, integer len);
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typedef void (mix_any_func_type)(uint8_t *const *out, const uint8_t *const *in1, const 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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union {
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float matrix_flt[SWR_CH_MAX][SWR_CH_MAX]; ///< single precision floating point rematrixing coefficients
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///< valid iff int_sample_fmt is AV_SAMPLE_FMT_FLTP
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int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
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///< valid iff int_sample_fmt is != AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP
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};
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union {
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int i;
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float f;
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double d;
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} native_one;
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uint8_t *native_matrix;
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union {
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int16_t i16[2];
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float f;
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} native_simd_one;
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uint8_t *native_simd_matrix;
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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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int swri_check_chlayout(struct SwrContext *s, const AVChannelLayout *chl, const char *name);
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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_build(SwrContext *s);
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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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