FFmpeg
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SwrContext Struct Reference

Public Attributes

const AVClassav_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.
 
enum AVSampleFormat user_int_sample_fmt
 User set internal sample format.
 
int user_dither_method
 User set dither method.
 
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
 
int exact_rational
 if 1 then enable non power of 2 phase_count
 
double cutoff
 resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). More...
 
int filter_type
 swr resampling filter type
 
double 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
 
double delayed_samples_fixup
 soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
 
struct AudioConvertin_convert
 input conversion context
 
struct AudioConvertout_convert
 output conversion context
 
struct AudioConvertfull_convert
 full conversion context (single conversion for input and output)
 
struct ResampleContextresample
 resampling context
 
struct Resampler const * resampler
 resampler virtual function table
 
double matrix [SWR_CH_MAX][SWR_CH_MAX]
 floating point rematrixing coefficients
 
float matrix_flt [SWR_CH_MAX][SWR_CH_MAX]
 single precision 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
 

Member Data Documentation

§ cutoff

double SwrContext::cutoff

resampling cutoff frequency (swr: 6dB point; soxr: 0dB point).

1.0 corresponds to half the output sample rate


The documentation for this struct was generated from the following file: