mirror of https://git.ffmpeg.org/ffmpeg.git
swr: Add API to make resample engine selectable.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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e8e575633f
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5a5d70748c
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@ -195,7 +195,7 @@ static int build_filter(ResampleContext *c, void *filter, double factor, int tap
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return 0;
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}
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ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
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static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
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double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta){
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double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
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int phase_count= 1<<phase_shift;
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@ -259,28 +259,14 @@ error:
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return NULL;
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}
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void swri_resample_free(ResampleContext **c){
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static void resample_free(ResampleContext **c){
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if(!*c)
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return;
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av_freep(&(*c)->filter_bank);
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av_freep(c);
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}
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int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
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ResampleContext *c;
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int ret;
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if (!s || compensation_distance < 0)
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return AVERROR(EINVAL);
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if (!compensation_distance && sample_delta)
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return AVERROR(EINVAL);
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if (!s->resample) {
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s->flags |= SWR_FLAG_RESAMPLE;
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ret = swr_init(s);
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if (ret < 0)
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return ret;
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}
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c= s->resample;
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static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){
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c->compensation_distance= compensation_distance;
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if (compensation_distance)
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c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
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@ -322,7 +308,7 @@ int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensatio
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#endif // HAVE_MMXEXT_INLINE
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int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
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static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
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int i, ret= -1;
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int av_unused mm_flags = av_get_cpu_flags();
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int need_emms= 0;
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@ -348,17 +334,20 @@ int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, Aud
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return ret;
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}
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int64_t swr_get_delay(struct SwrContext *s, int64_t base){
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static int64_t get_delay(struct SwrContext *s, int64_t base){
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ResampleContext *c = s->resample;
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if(c){
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int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
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num <<= c->phase_shift;
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num -= c->index;
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num *= c->src_incr;
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num -= c->frac;
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return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift);
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}else{
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return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
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}
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int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
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num <<= c->phase_shift;
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num -= c->index;
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num *= c->src_incr;
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num -= c->frac;
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return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift);
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}
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struct Resampler const swri_resampler={
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resample_init,
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resample_free,
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multiple_resample,
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set_compensation,
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get_delay,
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};
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@ -84,6 +84,8 @@ static const AVOption options[]={
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{"phase_shift" , "set resampling phase shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
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{"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
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{"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM },
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{"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"},
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{"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"},
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{"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
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, OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
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{"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
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@ -205,7 +207,8 @@ av_cold void swr_free(SwrContext **ss){
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swri_audio_convert_free(&s-> in_convert);
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swri_audio_convert_free(&s->out_convert);
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swri_audio_convert_free(&s->full_convert);
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swri_resample_free(&s->resample);
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if (s->resampler)
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s->resampler->free(&s->resample);
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swri_rematrix_free(s);
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}
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@ -258,13 +261,20 @@ av_cold int swr_init(struct SwrContext *s){
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return AVERROR(EINVAL);
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}
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switch(s->engine){
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case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
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default:
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av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
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return AVERROR(EINVAL);
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}
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set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
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set_audiodata_fmt(&s->out, s->out_sample_fmt);
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if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
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s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta);
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s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta);
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}else
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swri_resample_free(&s->resample);
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s->resampler->free(&s->resample);
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if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
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&& s->int_sample_fmt != AV_SAMPLE_FMT_S32P
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&& s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
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@ -463,7 +473,7 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count,
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int ret, size, consumed;
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if(!s->resample_in_constraint && s->in_buffer_count){
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buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
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ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
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ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
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out_count -= ret;
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ret_sum += ret;
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buf_set(&out, &out, ret);
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@ -483,7 +493,7 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count,
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if(in_count && !s->in_buffer_count){
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s->in_buffer_index=0;
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ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
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ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
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out_count -= ret;
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ret_sum += ret;
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buf_set(&out, &out, ret);
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@ -771,6 +781,34 @@ int swr_inject_silence(struct SwrContext *s, int count){
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return ret;
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}
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int64_t swr_get_delay(struct SwrContext *s, int64_t base){
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if (s->resampler && s->resample){
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return s->resampler->get_delay(s, base);
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}else{
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return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
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}
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}
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int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
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int ret;
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if (!s || compensation_distance < 0)
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return AVERROR(EINVAL);
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if (!compensation_distance && sample_delta)
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return AVERROR(EINVAL);
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if (!s->resample) {
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s->flags |= SWR_FLAG_RESAMPLE;
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ret = swr_init(s);
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if (ret < 0)
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return ret;
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}
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if (!s->resampler->set_compensation){
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return AVERROR(EINVAL);
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}else{
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return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
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}
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}
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int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
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if(pts == INT64_MIN)
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return s->outpts;
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@ -114,6 +114,12 @@ enum SwrDitherType {
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SWR_DITHER_NB, ///< not part of API/ABI
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};
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/** Resampling Engines */
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enum SwrEngine {
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SWR_ENGINE_SWR, /**< SW Resampler */
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SWR_ENGINE_NB, ///< not part of API/ABI
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};
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/** Resampling Filter Types */
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enum SwrFilterType {
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SWR_FILTER_TYPE_CUBIC, /**< Cubic */
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@ -67,6 +67,7 @@ struct SwrContext {
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enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
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const int *channel_map; ///< channel index (or -1 if muted channel) map
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int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
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enum SwrEngine engine;
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enum SwrDitherType dither_method;
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int dither_pos;
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float dither_scale;
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@ -104,6 +105,7 @@ struct SwrContext {
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struct AudioConvert *out_convert; ///< output conversion context
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struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
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struct ResampleContext *resample; ///< resampling context
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struct Resampler const *resampler; ///< resampler virtual function table
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float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
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uint8_t *native_matrix;
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@ -122,10 +124,23 @@ struct SwrContext {
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/* TODO: callbacks for ASM optimizations */
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};
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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, enum SwrFilterType, int kaiser_beta);
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void swri_resample_free(struct ResampleContext **c);
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int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
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void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance);
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typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
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double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta);
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typedef void (* resample_free_func)(struct ResampleContext **c);
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typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
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typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
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typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
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struct Resampler {
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resample_init_func init;
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resample_free_func free;
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multiple_resample_func multiple_resample;
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set_compensation_func set_compensation;
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get_delay_func get_delay;
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};
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extern struct Resampler const swri_resampler;
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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);
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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);
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int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx);
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