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https://github.com/librempeg/librempeg
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resample: add initial padding explicitly
This simplifies the code, since we do not have to deal with a possibly negative source index anymore.
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@ -33,7 +33,7 @@ struct ResampleContext {
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int filter_length;
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int ideal_dst_incr;
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int dst_incr;
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int index;
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unsigned int index;
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int frac;
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int src_incr;
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int compensation_distance;
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@ -45,11 +45,13 @@ struct ResampleContext {
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double factor;
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void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
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void (*resample_one)(struct ResampleContext *c, void *dst0,
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int dst_index, const void *src0, int src_size,
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int index, int frac);
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int dst_index, const void *src0,
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unsigned int index, int frac);
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void (*resample_nearest)(void *dst0, int dst_index,
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const void *src0, int index);
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const void *src0, unsigned int index);
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int padding_size;
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int initial_padding_filled;
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int initial_padding_samples;
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};
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@ -220,15 +222,18 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
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c->ideal_dst_incr = c->dst_incr;
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c->padding_size = (c->filter_length - 1) / 2;
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c->index = -phase_count * ((c->filter_length - 1) / 2);
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c->initial_padding_filled = 0;
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c->index = 0;
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c->frac = 0;
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/* allocate internal buffer */
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c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,
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c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,
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avr->internal_sample_fmt,
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"resample buffer");
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if (!c->buffer)
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goto error;
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c->buffer->nb_samples = c->padding_size;
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c->initial_padding_samples = c->padding_size;
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av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
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av_get_sample_fmt_name(avr->internal_sample_fmt),
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@ -342,7 +347,7 @@ static int resample(ResampleContext *c, void *dst, const void *src,
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int nearest_neighbour)
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{
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int dst_index;
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int index = c->index;
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unsigned 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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int dst_incr = c->dst_incr / c->src_incr;
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@ -352,7 +357,7 @@ static int resample(ResampleContext *c, void *dst, const void *src,
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return AVERROR(EINVAL);
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if (nearest_neighbour) {
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int64_t index2 = ((int64_t)index) << 32;
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uint64_t index2 = ((uint64_t)index) << 32;
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int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
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dst_size = FFMIN(dst_size,
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(src_size-1-index) * (int64_t)c->src_incr /
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@ -373,12 +378,11 @@ static int resample(ResampleContext *c, void *dst, const void *src,
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for (dst_index = 0; dst_index < dst_size; dst_index++) {
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int sample_index = index >> c->phase_shift;
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if (sample_index + c->filter_length > src_size ||
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-sample_index >= src_size)
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if (sample_index + c->filter_length > src_size)
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break;
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if (dst)
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c->resample_one(c, dst, dst_index, src, src_size, index, frac);
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c->resample_one(c, dst, dst_index, src, index, frac);
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frac += dst_incr_frac;
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index += dst_incr;
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@ -394,11 +398,10 @@ static int resample(ResampleContext *c, void *dst, const void *src,
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}
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}
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if (consumed)
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*consumed = FFMAX(index, 0) >> c->phase_shift;
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*consumed = index >> c->phase_shift;
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if (update_ctx) {
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if (index >= 0)
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index &= c->phase_mask;
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index &= c->phase_mask;
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if (compensation_distance) {
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compensation_distance -= dst_index;
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@ -437,6 +440,20 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
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/* TODO: pad buffer to flush completely */
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}
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if (!c->initial_padding_filled) {
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int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
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int i;
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if (c->buffer->nb_samples < 2 * c->padding_size)
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return 0;
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for (i = 0; i < c->padding_size; i++)
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for (ch = 0; ch < c->buffer->channels; ch++)
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memcpy(c->buffer->data[ch] + bps * i,
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c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
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c->initial_padding_filled = 1;
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}
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/* calculate output size and reallocate output buffer if needed */
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/* TODO: try to calculate this without the dummy resample() run */
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if (!dst->read_only && dst->allow_realloc) {
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@ -463,6 +480,7 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
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/* drain consumed samples from the internal buffer */
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ff_audio_data_drain(c->buffer, consumed);
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c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);
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av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
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in_samples, in_leftover, out_samples, c->buffer->nb_samples);
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@ -54,7 +54,7 @@
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#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
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#endif
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static void SET_TYPE(resample_nearest)(void *dst0, int dst_index, const void *src0, int index)
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static void SET_TYPE(resample_nearest)(void *dst0, int dst_index, const void *src0, unsigned int index)
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{
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FELEM *dst = dst0;
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const FELEM *src = src0;
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@ -63,21 +63,17 @@ static void SET_TYPE(resample_nearest)(void *dst0, int dst_index, const void *sr
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static void SET_TYPE(resample_one)(ResampleContext *c,
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void *dst0, int dst_index, const void *src0,
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int src_size, int index, int frac)
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unsigned int index, int frac)
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{
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FELEM *dst = dst0;
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const FELEM *src = src0;
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int i;
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int sample_index = index >> c->phase_shift;
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unsigned int sample_index = index >> c->phase_shift;
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FELEM2 val = 0;
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FELEM *filter = ((FELEM *)c->filter_bank) +
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c->filter_length * (index & c->phase_mask);
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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) % src_size] *
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(FELEM2)filter[i];
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} else if (c->linear) {
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if (c->linear) {
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FELEM2 v2 = 0;
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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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