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https://github.com/xenia-project/FFmpeg.git
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swr: handle initial negative sample index outside DSP function.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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@ -407,6 +407,51 @@ static int resample_flush(struct SwrContext *s) {
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return 0;
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}
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// in fact the whole handle multiple ridiculously small buffers might need more thinking...
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static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src,
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int in_count, int *out_idx, int *out_sz)
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{
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int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res;
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if (c->index >= 0)
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return 0;
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if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0)
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return res;
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// copy
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for (n = *out_sz; n < num; n++) {
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for (ch = 0; ch < src->ch_count; ch++) {
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memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
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src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size);
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}
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}
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// if not enough data is in, return and wait for more
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if (num < c->filter_length + 1) {
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*out_sz = num;
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*out_idx = c->filter_length;
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return INT_MAX;
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}
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// else invert
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for (n = 1; n <= c->filter_length; n++) {
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for (ch = 0; ch < src->ch_count; ch++) {
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memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size),
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dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
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c->felem_size);
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}
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}
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res = num - *out_sz;
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*out_idx = c->filter_length + (c->index >> c->phase_shift);
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*out_sz = 1 + c->filter_length * 2 - *out_idx;
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c->index &= c->phase_mask;
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assert(res > 0);
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return res;
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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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@ -414,4 +459,5 @@ struct Resampler const swri_resampler={
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resample_flush,
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set_compensation,
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get_delay,
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invert_initial_buffer,
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};
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@ -107,7 +107,10 @@
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#endif
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int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int *consumed, int src_size, int dst_size, int update_ctx){
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int dst_index, i;
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int dst_index;
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#if !defined(COMMON_CORE) || !defined(LINEAR_CORE)
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int i;
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#endif
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int index= c->index;
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int frac= c->frac;
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int dst_incr_frac= c->dst_incr % c->src_incr;
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@ -133,7 +136,7 @@ int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int
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av_assert2(index >= 0);
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*consumed= index;
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index = 0;
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} else if (index >= 0) {
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} else {
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int64_t end_index = (1LL + src_size - c->filter_length) << c->phase_shift;
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int64_t delta_frac = (end_index - index) * c->src_incr - c->frac;
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int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
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@ -195,54 +198,6 @@ int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int
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}
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*consumed = sample_index;
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} else {
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int sample_index = 0;
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for(dst_index=0; dst_index < dst_size; dst_index++){
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FELEM *filter;
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FELEM2 val=0;
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sample_index += index >> c->phase_shift;
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index &= c->phase_mask;
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filter = ((FELEM*)c->filter_bank) + c->filter_alloc*index;
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if(sample_index + c->filter_length > src_size || -sample_index >= src_size){
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break;
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}else if(sample_index < 0){
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for(i=0; i<c->filter_length; i++)
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val += src[FFABS(sample_index + i)] * (FELEM2)filter[i];
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OUT(dst[dst_index], val);
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}else if(c->linear){
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FELEM2 v2=0;
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#ifdef LINEAR_CORE
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LINEAR_CORE
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#else
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for(i=0; i<c->filter_length; i++){
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val += src[sample_index + i] * (FELEM2)filter[i];
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v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_alloc];
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}
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#endif
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val+=(v2-val)*(FELEML)frac / c->src_incr;
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OUT(dst[dst_index], val);
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}else{
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#ifdef COMMON_CORE
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COMMON_CORE
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#else
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for(i=0; i<c->filter_length; i++){
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val += src[sample_index + i] * (FELEM2)filter[i];
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}
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OUT(dst[dst_index], val);
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#endif
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}
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frac += dst_incr_frac;
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index += dst_incr;
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if(frac >= c->src_incr){
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frac -= c->src_incr;
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index++;
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}
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}
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*consumed= FFMAX(sample_index, 0);
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index += FFMIN(sample_index, 0) << c->phase_shift;
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}
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if(update_ctx){
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@ -87,7 +87,14 @@ static int64_t get_delay(struct SwrContext *s, int64_t base){
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return (int64_t)(delay_s * base + .5);
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}
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static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src,
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int in_count, int *out_idx, int *out_sz)
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{
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return 0;
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}
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struct Resampler const soxr_resampler={
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create, destroy, process, flush, NULL /* set_compensation */, get_delay,
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invert_initial_buffer,
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};
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@ -541,6 +541,12 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count,
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tmp=out=*out_param;
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in = *in_param;
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border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
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&in, in_count, &s->in_buffer_index, &s->in_buffer_count);
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if (border == INT_MAX) return 0;
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else if (border < 0) return border;
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else if (border) { buf_set(&in, &in, border); in_count -= border; s->resample_in_constraint = 0; }
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do{
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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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@ -157,6 +157,7 @@ typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData
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typedef int (* resample_flush_func)(struct SwrContext *c);
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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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typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
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struct Resampler {
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resample_init_func init;
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@ -165,6 +166,7 @@ struct Resampler {
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resample_flush_func flush;
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set_compensation_func set_compensation;
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get_delay_func get_delay;
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invert_initial_buffer_func invert_initial_buffer;
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};
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extern struct Resampler const swri_resampler;
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