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author | Anton Khirnov <anton@khirnov.net> | 2014-03-03 22:26:57 +0100 |
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committer | Anton Khirnov <anton@khirnov.net> | 2014-04-11 16:26:36 +0200 |
commit | f20892eb67a7b4b5a3c080388a35a3380516a805 (patch) | |
tree | 179aaaf0886c0a48701f62abd7f23a63a465f002 /libavresample/resample.c | |
parent | b9dea23766f52b8e059e72d34980bb7b456efe8f (diff) | |
download | ffmpeg-f20892eb67a7b4b5a3c080388a35a3380516a805.tar.gz |
resample: split the nearest neighbour path into a separate function pointer
Diffstat (limited to 'libavresample/resample.c')
-rw-r--r-- | libavresample/resample.c | 25 |
1 files changed, 17 insertions, 8 deletions
diff --git a/libavresample/resample.c b/libavresample/resample.c index d9c4e019ec..904891258a 100644 --- a/libavresample/resample.c +++ b/libavresample/resample.c @@ -44,9 +44,11 @@ struct ResampleContext { int kaiser_beta; double factor; void (*set_filter)(void *filter, double *tab, int phase, int tap_count); - void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0, + void (*resample_one)(struct ResampleContext *c, void *dst0, int dst_index, const void *src0, int src_size, int index, int frac); + void (*resample_nearest)(void *dst0, int dst_index, + const void *src0, int index); int padding_size; }; @@ -178,18 +180,22 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr) switch (avr->internal_sample_fmt) { case AV_SAMPLE_FMT_DBLP: c->resample_one = resample_one_dbl; + c->resample_nearest = resample_nearest_dbl; c->set_filter = set_filter_dbl; break; case AV_SAMPLE_FMT_FLTP: c->resample_one = resample_one_flt; + c->resample_nearest = resample_nearest_flt; c->set_filter = set_filter_flt; break; case AV_SAMPLE_FMT_S32P: c->resample_one = resample_one_s32; + c->resample_nearest = resample_nearest_s32; c->set_filter = set_filter_s32; break; case AV_SAMPLE_FMT_S16P: c->resample_one = resample_one_s16; + c->resample_nearest = resample_nearest_s16; c->set_filter = set_filter_s16; break; } @@ -332,7 +338,8 @@ reinit_fail: } static int resample(ResampleContext *c, void *dst, const void *src, - int *consumed, int src_size, int dst_size, int update_ctx) + int *consumed, int src_size, int dst_size, int update_ctx, + int nearest_neighbour) { int dst_index; int index = c->index; @@ -344,8 +351,7 @@ static int resample(ResampleContext *c, void *dst, const void *src, if (!dst != !src) return AVERROR(EINVAL); - if (compensation_distance == 0 && c->filter_length == 1 && - c->phase_shift == 0) { + if (nearest_neighbour) { int64_t index2 = ((int64_t)index) << 32; int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr; dst_size = FFMIN(dst_size, @@ -354,7 +360,7 @@ static int resample(ResampleContext *c, void *dst, const void *src, if (dst) { for(dst_index = 0; dst_index < dst_size; dst_index++) { - c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0); + c->resample_nearest(dst, dst_index, src, index2 >> 32); index2 += incr; } } else { @@ -372,7 +378,7 @@ static int resample(ResampleContext *c, void *dst, const void *src, break; if (dst) - c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac); + c->resample_one(c, dst, dst_index, src, src_size, index, frac); frac += dst_incr_frac; index += dst_incr; @@ -412,6 +418,9 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src) { int ch, in_samples, in_leftover, consumed = 0, out_samples = 0; int ret = AVERROR(EINVAL); + int nearest_neighbour = (c->compensation_distance == 0 && + c->filter_length == 1 && + c->phase_shift == 0); in_samples = src ? src->nb_samples : 0; in_leftover = c->buffer->nb_samples; @@ -432,7 +441,7 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src) /* TODO: try to calculate this without the dummy resample() run */ if (!dst->read_only && dst->allow_realloc) { out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples, - INT_MAX, 0); + INT_MAX, 0, nearest_neighbour); ret = ff_audio_data_realloc(dst, out_samples); if (ret < 0) { av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n"); @@ -445,7 +454,7 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src) out_samples = resample(c, (void *)dst->data[ch], (const void *)c->buffer->data[ch], &consumed, c->buffer->nb_samples, dst->allocated_samples, - ch + 1 == c->buffer->channels); + ch + 1 == c->buffer->channels, nearest_neighbour); } if (out_samples < 0) { av_log(c->avr, AV_LOG_ERROR, "error during resampling\n"); |