diff options
Diffstat (limited to 'libavresample/resample.c')
| -rw-r--r-- | libavresample/resample.c | 46 |
1 files changed, 32 insertions, 14 deletions
diff --git a/libavresample/resample.c b/libavresample/resample.c index 904891258a..c02bba472d 100644 --- a/libavresample/resample.c +++ b/libavresample/resample.c @@ -33,7 +33,7 @@ struct ResampleContext { int filter_length; int ideal_dst_incr; int dst_incr; - int index; + unsigned int index; int frac; int src_incr; int compensation_distance; @@ -45,11 +45,13 @@ struct ResampleContext { double factor; void (*set_filter)(void *filter, double *tab, int phase, int tap_count); void (*resample_one)(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, int src_size, - int index, int frac); + int dst_index, const void *src0, + unsigned int index, int frac); void (*resample_nearest)(void *dst0, int dst_index, - const void *src0, int index); + const void *src0, unsigned int index); int padding_size; + int initial_padding_filled; + int initial_padding_samples; }; @@ -220,15 +222,18 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr) c->ideal_dst_incr = c->dst_incr; c->padding_size = (c->filter_length - 1) / 2; - c->index = -phase_count * ((c->filter_length - 1) / 2); + c->initial_padding_filled = 0; + c->index = 0; c->frac = 0; /* allocate internal buffer */ - c->buffer = ff_audio_data_alloc(avr->resample_channels, 0, + c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size, avr->internal_sample_fmt, "resample buffer"); if (!c->buffer) goto error; + c->buffer->nb_samples = c->padding_size; + c->initial_padding_samples = c->padding_size; av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n", av_get_sample_fmt_name(avr->internal_sample_fmt), @@ -342,7 +347,7 @@ static int resample(ResampleContext *c, void *dst, const void *src, int nearest_neighbour) { int dst_index; - int index = c->index; + unsigned int index = c->index; int frac = c->frac; int dst_incr_frac = c->dst_incr % c->src_incr; int dst_incr = c->dst_incr / c->src_incr; @@ -352,7 +357,7 @@ static int resample(ResampleContext *c, void *dst, const void *src, return AVERROR(EINVAL); if (nearest_neighbour) { - int64_t index2 = ((int64_t)index) << 32; + uint64_t index2 = ((uint64_t)index) << 32; int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr; dst_size = FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / @@ -373,12 +378,11 @@ static int resample(ResampleContext *c, void *dst, const void *src, for (dst_index = 0; dst_index < dst_size; dst_index++) { int sample_index = index >> c->phase_shift; - if (sample_index + c->filter_length > src_size || - -sample_index >= src_size) + if (sample_index + c->filter_length > src_size) break; if (dst) - c->resample_one(c, dst, dst_index, src, src_size, index, frac); + c->resample_one(c, dst, dst_index, src, index, frac); frac += dst_incr_frac; index += dst_incr; @@ -394,11 +398,10 @@ static int resample(ResampleContext *c, void *dst, const void *src, } } if (consumed) - *consumed = FFMAX(index, 0) >> c->phase_shift; + *consumed = index >> c->phase_shift; if (update_ctx) { - if (index >= 0) - index &= c->phase_mask; + index &= c->phase_mask; if (compensation_distance) { compensation_distance -= dst_index; @@ -437,6 +440,20 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src) /* TODO: pad buffer to flush completely */ } + if (!c->initial_padding_filled) { + int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt); + int i; + + if (c->buffer->nb_samples < 2 * c->padding_size) + return 0; + + for (i = 0; i < c->padding_size; i++) + for (ch = 0; ch < c->buffer->channels; ch++) + memcpy(c->buffer->data[ch] + bps * i, + c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps); + c->initial_padding_filled = 1; + } + /* calculate output size and reallocate output buffer if needed */ /* TODO: try to calculate this without the dummy resample() run */ if (!dst->read_only && dst->allow_realloc) { @@ -463,6 +480,7 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src) /* drain consumed samples from the internal buffer */ ff_audio_data_drain(c->buffer, consumed); + c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0); av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n", in_samples, in_leftover, out_samples, c->buffer->nb_samples); |