|
- /*
- * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
- *
- * This file is part of libswresample
- *
- * libswresample is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * libswresample is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with libswresample; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
- #ifndef SWR_INTERNAL_H
- #define SWR_INTERNAL_H
-
- #include "swresample.h"
- #include "libavutil/channel_layout.h"
- #include "config.h"
-
- #define SWR_CH_MAX 32
-
- #define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
-
- #define NS_TAPS 20
-
- #if ARCH_X86_64
- typedef int64_t integer;
- #else
- typedef int integer;
- #endif
-
- typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
- typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
-
- typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
-
- typedef struct AudioData{
- uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
- uint8_t *data; ///< samples buffer
- int ch_count; ///< number of channels
- int bps; ///< bytes per sample
- int count; ///< number of samples
- int planar; ///< 1 if planar audio, 0 otherwise
- enum AVSampleFormat fmt; ///< sample format
- } AudioData;
-
- struct DitherContext {
- int method;
- int noise_pos;
- float scale;
- float noise_scale; ///< Noise scale
- int ns_taps; ///< Noise shaping dither taps
- float ns_scale; ///< Noise shaping dither scale
- float ns_scale_1; ///< Noise shaping dither scale^-1
- int ns_pos; ///< Noise shaping dither position
- float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients
- float ns_errors[SWR_CH_MAX][2*NS_TAPS];
- AudioData noise; ///< noise used for dithering
- AudioData temp; ///< temporary storage when writing into the input buffer isn't possible
- int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly
- };
-
- typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
- double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
- typedef void (* resample_free_func)(struct ResampleContext **c);
- typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
- typedef int (* resample_flush_func)(struct SwrContext *c);
- typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
- typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
- typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
-
- struct Resampler {
- resample_init_func init;
- resample_free_func free;
- multiple_resample_func multiple_resample;
- resample_flush_func flush;
- set_compensation_func set_compensation;
- get_delay_func get_delay;
- invert_initial_buffer_func invert_initial_buffer;
- };
-
- extern struct Resampler const swri_resampler;
-
- struct SwrContext {
- const AVClass *av_class; ///< AVClass used for AVOption and av_log()
- int log_level_offset; ///< logging level offset
- void *log_ctx; ///< parent logging context
- enum AVSampleFormat in_sample_fmt; ///< input sample format
- enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
- enum AVSampleFormat out_sample_fmt; ///< output sample format
- int64_t in_ch_layout; ///< input channel layout
- int64_t out_ch_layout; ///< output channel layout
- int in_sample_rate; ///< input sample rate
- int out_sample_rate; ///< output sample rate
- int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
- float slev; ///< surround mixing level
- float clev; ///< center mixing level
- float lfe_mix_level; ///< LFE mixing level
- float rematrix_volume; ///< rematrixing volume coefficient
- float rematrix_maxval; ///< maximum value for rematrixing output
- int matrix_encoding; /**< matrixed stereo encoding */
- const int *channel_map; ///< channel index (or -1 if muted channel) map
- int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
- int engine;
-
- struct DitherContext dither;
-
- int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
- int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
- int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
- double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
- int filter_type; /**< swr resampling filter type */
- int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
- double precision; /**< soxr resampling precision (in bits) */
- int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
-
- float min_compensation; ///< swr minimum below which no compensation will happen
- float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
- float soft_compensation_duration; ///< swr duration over which soft compensation is applied
- float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
- float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
- int64_t firstpts_in_samples; ///< swr first pts in samples
-
- int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
- int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
- int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
-
- AudioData in; ///< input audio data
- AudioData postin; ///< post-input audio data: used for rematrix/resample
- AudioData midbuf; ///< intermediate audio data (postin/preout)
- AudioData preout; ///< pre-output audio data: used for rematrix/resample
- AudioData out; ///< converted output audio data
- AudioData in_buffer; ///< cached audio data (convert and resample purpose)
- AudioData silence; ///< temporary with silence
- AudioData drop_temp; ///< temporary used to discard output
- int in_buffer_index; ///< cached buffer position
- int in_buffer_count; ///< cached buffer length
- int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
- int flushed; ///< 1 if data is to be flushed and no further input is expected
- int64_t outpts; ///< output PTS
- int64_t firstpts; ///< first PTS
- int drop_output; ///< number of output samples to drop
-
- struct AudioConvert *in_convert; ///< input conversion context
- struct AudioConvert *out_convert; ///< output conversion context
- struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
- struct ResampleContext *resample; ///< resampling context
- struct Resampler const *resampler; ///< resampler virtual function table
-
- float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
- uint8_t *native_matrix;
- uint8_t *native_one;
- uint8_t *native_simd_one;
- uint8_t *native_simd_matrix;
- int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
- uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
- mix_1_1_func_type *mix_1_1_f;
- mix_1_1_func_type *mix_1_1_simd;
-
- mix_2_1_func_type *mix_2_1_f;
- mix_2_1_func_type *mix_2_1_simd;
-
- mix_any_func_type *mix_any_f;
-
- /* TODO: callbacks for ASM optimizations */
- };
-
- int swri_realloc_audio(AudioData *a, int count);
-
- void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
- void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
- void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
- void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
-
- int swri_rematrix_init(SwrContext *s);
- void swri_rematrix_free(SwrContext *s);
- int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
- int swri_rematrix_init_x86(struct SwrContext *s);
-
- void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
- int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
-
- void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels);
- void swri_audio_convert_init_arm(struct AudioConvert *ac,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels);
- void swri_audio_convert_init_x86(struct AudioConvert *ac,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels);
- #endif
|
