#include "swresample.h"
+typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, int index, int len);
+typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, int index1, int index2, int len);
+
+typedef void (mix_any_func_type)(void **out, const void **in1, void *coeffp, int len);
+
typedef struct AudioData{
uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
uint8_t *data; ///< samples buffer
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 SwrContext {
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_FLT or AV_SAMPLE_FMT_S16)
+ 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 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
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 */
+
+ float min_compensation; ///< minimum below which no compensation will happen
+ float min_hard_compensation; ///< minimum below which no silence inject / sample drop will happen
+ float soft_compensation_duration; ///< duration over which soft compensation is applied
+ float max_soft_compensation; ///< maximum soft compensation in seconds over soft_compensation_duration
- 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 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
+ int64_t outpts; ///< output 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 ResampleContext *resample; ///< resampling context
float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
+ uint8_t *native_matrix;
+ uint8_t *native_one;
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_2_1_func_type *mix_2_1_f;
+
+ mix_any_func_type *mix_any_f;
/* 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);
+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(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
+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);
+void swri_rematrix_free(SwrContext *s);
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
+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
projects / ffmpeg / blobdiff