diff options
author | Rob Sykes <aquegg@yahoo.co.uk> | 2012-12-11 18:36:58 +0100 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2012-12-11 22:04:00 +0100 |
commit | 5a5d70748c5d606b055fedce30a84e31790d6d15 (patch) | |
tree | 4996c759f4f30c590611424f3f0a547fa09ff35a | |
parent | e8e575633faf19711910cf9caf59f7db300a9ccd (diff) | |
download | ffmpeg-5a5d70748c5d606b055fedce30a84e31790d6d15.tar.gz |
swr: Add API to make resample engine selectable.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
-rw-r--r-- | libswresample/resample.c | 49 | ||||
-rw-r--r-- | libswresample/swresample.c | 48 | ||||
-rw-r--r-- | libswresample/swresample.h | 6 | ||||
-rw-r--r-- | libswresample/swresample_internal.h | 23 |
4 files changed, 87 insertions, 39 deletions
diff --git a/libswresample/resample.c b/libswresample/resample.c index 2096e43db6..f3881cdd57 100644 --- a/libswresample/resample.c +++ b/libswresample/resample.c @@ -195,7 +195,7 @@ static int build_filter(ResampleContext *c, void *filter, double factor, int tap return 0; } -ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, +static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta){ double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); int phase_count= 1<<phase_shift; @@ -259,28 +259,14 @@ error: return NULL; } -void swri_resample_free(ResampleContext **c){ +static void resample_free(ResampleContext **c){ if(!*c) return; av_freep(&(*c)->filter_bank); av_freep(c); } -int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ - ResampleContext *c; - int ret; - - if (!s || compensation_distance < 0) - return AVERROR(EINVAL); - if (!compensation_distance && sample_delta) - return AVERROR(EINVAL); - if (!s->resample) { - s->flags |= SWR_FLAG_RESAMPLE; - ret = swr_init(s); - if (ret < 0) - return ret; - } - c= s->resample; +static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){ c->compensation_distance= compensation_distance; if (compensation_distance) c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; @@ -322,7 +308,7 @@ int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensatio #endif // HAVE_MMXEXT_INLINE -int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ +static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ int i, ret= -1; int av_unused mm_flags = av_get_cpu_flags(); int need_emms= 0; @@ -348,17 +334,20 @@ int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, Aud return ret; } -int64_t swr_get_delay(struct SwrContext *s, int64_t base){ +static int64_t get_delay(struct SwrContext *s, int64_t base){ ResampleContext *c = s->resample; - if(c){ - int64_t num = s->in_buffer_count - (c->filter_length-1)/2; - num <<= c->phase_shift; - num -= c->index; - num *= c->src_incr; - num -= c->frac; - - return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift); - }else{ - return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; - } + int64_t num = s->in_buffer_count - (c->filter_length-1)/2; + num <<= c->phase_shift; + num -= c->index; + num *= c->src_incr; + num -= c->frac; + return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift); } + +struct Resampler const swri_resampler={ + resample_init, + resample_free, + multiple_resample, + set_compensation, + get_delay, +}; diff --git a/libswresample/swresample.c b/libswresample/swresample.c index c1668dabc7..1eaa415c1a 100644 --- a/libswresample/swresample.c +++ b/libswresample/swresample.c @@ -84,6 +84,8 @@ static const AVOption options[]={ {"phase_shift" , "set resampling phase shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM }, {"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM }, {"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM }, +{"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"}, +{"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"}, {"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied" , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM }, {"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data." @@ -205,7 +207,8 @@ av_cold void swr_free(SwrContext **ss){ swri_audio_convert_free(&s-> in_convert); swri_audio_convert_free(&s->out_convert); swri_audio_convert_free(&s->full_convert); - swri_resample_free(&s->resample); + if (s->resampler) + s->resampler->free(&s->resample); swri_rematrix_free(s); } @@ -258,13 +261,20 @@ av_cold int swr_init(struct SwrContext *s){ return AVERROR(EINVAL); } + switch(s->engine){ + case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break; + default: + av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n"); + return AVERROR(EINVAL); + } + set_audiodata_fmt(&s-> in, s-> in_sample_fmt); set_audiodata_fmt(&s->out, s->out_sample_fmt); if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){ - 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); + 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); }else - swri_resample_free(&s->resample); + s->resampler->free(&s->resample); if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P && s->int_sample_fmt != AV_SAMPLE_FMT_S32P && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP @@ -463,7 +473,7 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count, int ret, size, consumed; if(!s->resample_in_constraint && s->in_buffer_count){ buf_set(&tmp, &s->in_buffer, s->in_buffer_index); - ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed); + ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed); out_count -= ret; ret_sum += ret; buf_set(&out, &out, ret); @@ -483,7 +493,7 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count, if(in_count && !s->in_buffer_count){ s->in_buffer_index=0; - ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed); + ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed); out_count -= ret; ret_sum += ret; buf_set(&out, &out, ret); @@ -771,6 +781,34 @@ int swr_inject_silence(struct SwrContext *s, int count){ return ret; } +int64_t swr_get_delay(struct SwrContext *s, int64_t base){ + if (s->resampler && s->resample){ + return s->resampler->get_delay(s, base); + }else{ + return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; + } +} + +int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ + int ret; + + if (!s || compensation_distance < 0) + return AVERROR(EINVAL); + if (!compensation_distance && sample_delta) + return AVERROR(EINVAL); + if (!s->resample) { + s->flags |= SWR_FLAG_RESAMPLE; + ret = swr_init(s); + if (ret < 0) + return ret; + } + if (!s->resampler->set_compensation){ + return AVERROR(EINVAL); + }else{ + return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance); + } +} + int64_t swr_next_pts(struct SwrContext *s, int64_t pts){ if(pts == INT64_MIN) return s->outpts; diff --git a/libswresample/swresample.h b/libswresample/swresample.h index 8d9f77d3c8..356fb61488 100644 --- a/libswresample/swresample.h +++ b/libswresample/swresample.h @@ -114,6 +114,12 @@ enum SwrDitherType { SWR_DITHER_NB, ///< not part of API/ABI }; +/** Resampling Engines */ +enum SwrEngine { + SWR_ENGINE_SWR, /**< SW Resampler */ + SWR_ENGINE_NB, ///< not part of API/ABI +}; + /** Resampling Filter Types */ enum SwrFilterType { SWR_FILTER_TYPE_CUBIC, /**< Cubic */ diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h index 459b1b0868..6d607e5d18 100644 --- a/libswresample/swresample_internal.h +++ b/libswresample/swresample_internal.h @@ -67,6 +67,7 @@ struct SwrContext { enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */ const int *channel_map; ///< channel index (or -1 if muted channel) map int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count) + enum SwrEngine engine; enum SwrDitherType dither_method; int dither_pos; float dither_scale; @@ -104,6 +105,7 @@ struct SwrContext { struct AudioConvert *out_convert; ///< output conversion context struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output) struct ResampleContext *resample; ///< resampling context + struct Resampler const *resampler; ///< resampler virtual function table float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients uint8_t *native_matrix; @@ -122,10 +124,23 @@ struct SwrContext { /* TODO: callbacks for ASM optimizations */ }; -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); -void swri_resample_free(struct ResampleContext **c); -int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); -void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance); +typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, + double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta); +typedef void (* resample_free_func)(struct ResampleContext **c); +typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); +typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance); +typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base); + +struct Resampler { + resample_init_func init; + resample_free_func free; + multiple_resample_func multiple_resample; + set_compensation_func set_compensation; + get_delay_func get_delay; +}; + +extern struct Resampler const swri_resampler; + 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); 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); int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx); |