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- /*
- * Copyright (C) 2011 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"
-
- 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
- } AudioData;
-
- 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 rematrix_volume; ///< rematrixing volume coefficient
- 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)
- enum SwrDitherType dither_method;
- int dither_pos;
- float dither_scale;
- 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. 1.0 corresponds to half the output sample rate */
-
- int int_bps; ///< internal bytes per sample
- 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 dither; ///< noise used for dithering
- 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
-
- 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
-
- float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
- 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
-
- /* TODO: callbacks for ASM optimizations */
- };
-
- struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat);
- void swri_resample_free(struct ResampleContext **c);
- int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
- void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance);
- int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
- int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
- int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx);
- int swri_resample_double(struct ResampleContext *c,double *dst, const double *src, int *consumed, int src_size, int dst_size, int update_ctx);
-
- int swri_rematrix_init(SwrContext *s);
- int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
- void swri_sum2(enum AVSampleFormat format, void *dst, const void *src0, const void *src1, float coef0, float coef1, int len);
-
- void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
-
- void swri_audio_convert_init_x86(struct AudioConvert *ac,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels);
- #endif
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