[FFmpeg-cvslog] swr: Add API to make resample engine selectable.
Rob Sykes
git at videolan.org
Tue Dec 11 22:17:27 CET 2012
ffmpeg | branch: master | Rob Sykes <aquegg at yahoo.co.uk> | Tue Dec 11 18:36:58 2012 +0100| [5a5d70748c5d606b055fedce30a84e31790d6d15] | committer: Michael Niedermayer
swr: Add API to make resample engine selectable.
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=5a5d70748c5d606b055fedce30a84e31790d6d15
---
libswresample/resample.c | 49 ++++++++++++++---------------------
libswresample/swresample.c | 48 ++++++++++++++++++++++++++++++----
libswresample/swresample.h | 6 +++++
libswresample/swresample_internal.h | 23 +++++++++++++---
4 files changed, 87 insertions(+), 39 deletions(-)
diff --git a/libswresample/resample.c b/libswresample/resample.c
index 2096e43..f3881cd 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 c1668da..1eaa415 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 8d9f77d..356fb61 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 459b1b0..6d607e5 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);
More information about the ffmpeg-cvslog
mailing list