| /* |
| * 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; |
| extern struct Resampler const swri_soxr_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; |
| |
| int user_in_ch_count; ///< User set input channel count |
| int user_out_ch_count; ///< User set output channel count |
| int user_used_ch_count; ///< User set used channel count |
| int64_t user_in_ch_layout; ///< User set input channel layout |
| int64_t user_out_ch_layout; ///< User set output channel layout |
| |
| 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 |
