[FFmpeg-devel] [PATCH] swresample: Add support for clipping float/double to -1.0..1.0 range
wm4
nfxjfg at googlemail.com
Fri Oct 23 19:18:06 CEST 2015
On Fri, 23 Oct 2015 18:08:00 +0200
Michael Niedermayer <michaelni at gmx.at> wrote:
> From: Michael Niedermayer <michael at niedermayer.cc>
>
> Signed-off-by: Michael Niedermayer <michael at niedermayer.cc>
> ---
> libswresample/aarch64/audio_convert_init.c | 8 ++++-
> libswresample/arm/audio_convert_init.c | 8 ++++-
> libswresample/audioconvert.c | 44 ++++++++++++++++++++++++++--
> libswresample/options.c | 1 +
> libswresample/swresample.c | 6 ++--
> libswresample/swresample.h | 1 +
> libswresample/swresample_internal.h | 6 ++--
> libswresample/x86/audio_convert_init.c | 8 ++++-
> 8 files changed, 71 insertions(+), 11 deletions(-)
>
> diff --git a/libswresample/aarch64/audio_convert_init.c b/libswresample/aarch64/audio_convert_init.c
> index 60e24ad..dedb1aa 100644
> --- a/libswresample/aarch64/audio_convert_init.c
> +++ b/libswresample/aarch64/audio_convert_init.c
> @@ -48,12 +48,18 @@ static void conv_fltp_to_s16_nch_neon(uint8_t **dst, const uint8_t **src, int le
> av_cold void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
> enum AVSampleFormat out_fmt,
> enum AVSampleFormat in_fmt,
> - int channels)
> + int channels, int flags)
> {
> int cpu_flags = av_get_cpu_flags();
>
> ac->simd_f= NULL;
>
> + if ( (flags & SWR_FLAG_CLIP)
> + && av_get_packed_sample_fmt(in_fmt) == AV_SAMPLE_FMT_FLT
> + && av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_FLT) {
> + return;
> + }
> +
> if (have_neon(cpu_flags)) {
> if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLTP)
> ac->simd_f = conv_flt_to_s16_neon;
> diff --git a/libswresample/arm/audio_convert_init.c b/libswresample/arm/audio_convert_init.c
> index ec9e62e..f39978d 100644
> --- a/libswresample/arm/audio_convert_init.c
> +++ b/libswresample/arm/audio_convert_init.c
> @@ -48,12 +48,18 @@ static void conv_fltp_to_s16_nch_neon(uint8_t **dst, const uint8_t **src, int le
> av_cold void swri_audio_convert_init_arm(struct AudioConvert *ac,
> enum AVSampleFormat out_fmt,
> enum AVSampleFormat in_fmt,
> - int channels)
> + int channels, int flags)
> {
> int cpu_flags = av_get_cpu_flags();
>
> ac->simd_f= NULL;
>
> + if ( (flags & SWR_FLAG_CLIP)
> + && av_get_packed_sample_fmt(in_fmt) == AV_SAMPLE_FMT_FLT
> + && av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_FLT) {
> + return;
> + }
> +
> if (have_neon(cpu_flags)) {
> if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLTP)
> ac->simd_f = conv_flt_to_s16_neon;
> diff --git a/libswresample/audioconvert.c b/libswresample/audioconvert.c
> index 58b0bf3..dc6734a 100644
> --- a/libswresample/audioconvert.c
> +++ b/libswresample/audioconvert.c
> @@ -77,6 +77,27 @@ CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*
> CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
> CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
>
> +#define CONV_FUNC_NAME_CLIP(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt ## _clip
> +#define CONV_FUNC_CLIP(ofmt, otype, ifmt, expr)\
> +static void CONV_FUNC_NAME_CLIP(ofmt, ifmt)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)\
> +{\
> + uint8_t *end2 = end - 3*os;\
> + while(po < end2){\
> + *(otype*)po = expr; pi += is; po += os;\
> + *(otype*)po = expr; pi += is; po += os;\
> + *(otype*)po = expr; pi += is; po += os;\
> + *(otype*)po = expr; pi += is; po += os;\
> + }\
> + while(po < end){\
> + *(otype*)po = expr; pi += is; po += os;\
> + }\
> +}
> +
> +CONV_FUNC_CLIP(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, av_clipf(*(const float*)pi, -1.0, 1.0))
> +CONV_FUNC_CLIP(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, av_clipf(*(const float*)pi, -1.0, 1.0))
> +CONV_FUNC_CLIP(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, av_clipf(*(const double*)pi, -1.0, 1.0))
> +CONV_FUNC_CLIP(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, av_clipd(*(const double*)pi, -1.0, 1.0))
> +
> #define FMT_PAIR_FUNC(out, in) [(out) + AV_SAMPLE_FMT_NB*(in)] = CONV_FUNC_NAME(out, in)
>
> static conv_func_type * const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB*AV_SAMPLE_FMT_NB] = {
> @@ -107,6 +128,15 @@ static conv_func_type * const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB*AV_SAM
> FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL),
> };
>
> +#define FMT_PAIR_FUNC_CLIP(out, in) [(out) + AV_SAMPLE_FMT_NB*(in)] = CONV_FUNC_NAME_CLIP(out, in)
> +
> +static conv_func_type * const fmt_pair_to_conv_functions_clip[AV_SAMPLE_FMT_NB*AV_SAMPLE_FMT_NB] = {
> + FMT_PAIR_FUNC_CLIP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT),
> + FMT_PAIR_FUNC_CLIP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT),
> + FMT_PAIR_FUNC_CLIP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL),
> + FMT_PAIR_FUNC_CLIP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL),
> +};
> +
> static void cpy1(uint8_t **dst, const uint8_t **src, int len){
> memcpy(*dst, *src, len);
> }
> @@ -154,9 +184,17 @@ AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt,
> }
> }
>
> - if(HAVE_YASM && HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);
> - if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);
> - if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);
> + if (flags & SWR_FLAG_CLIP) {
> + conv_func_type *f2 = fmt_pair_to_conv_functions_clip[av_get_packed_sample_fmt(out_fmt) + AV_SAMPLE_FMT_NB*av_get_packed_sample_fmt(in_fmt)];
> + if (f2) {
> + f = f2;
> + ctx->simd_f = NULL;
> + }
> + }
> +
> + if(HAVE_YASM && HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels, flags);
> + if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels, flags);
> + if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels, flags);
>
> return ctx;
> }
> diff --git a/libswresample/options.c b/libswresample/options.c
> index 0bcb102..bb68158 100644
> --- a/libswresample/options.c
> +++ b/libswresample/options.c
> @@ -67,6 +67,7 @@ static const AVOption options[]={
> {"flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
> {"swr_flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
> {"res" , "force resampling" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},
> +{"clip" , "clip float/double to -1.0..1.0", 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_CLIP }, INT_MIN, INT_MAX , PARAM, "flags"},
>
> {"dither_scale" , "set dither scale" , OFFSET(dither.scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
>
> diff --git a/libswresample/swresample.c b/libswresample/swresample.c
> index 8e23899..029b85e 100644
> --- a/libswresample/swresample.c
> +++ b/libswresample/swresample.c
> @@ -324,14 +324,14 @@ av_assert0(s->out.ch_count);
>
> if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
> s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
> - s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
> + s-> in_sample_fmt, s-> in.ch_count, NULL, s->flags & SWR_FLAG_CLIP);
> return 0;
> }
>
> s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
> s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
> s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
> - s->int_sample_fmt, s->out.ch_count, NULL, 0);
> + s->int_sample_fmt, s->out.ch_count, NULL, s->flags & SWR_FLAG_CLIP);
>
> if (!s->in_convert || !s->out_convert) {
> ret = AVERROR(ENOMEM);
> @@ -606,6 +606,7 @@ static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_co
> preout= midbuf;
>
> if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
> + && !((s->flags & SWR_FLAG_CLIP) && (s->out_sample_fmt == AV_SAMPLE_FMT_FLT || s->out_sample_fmt == AV_SAMPLE_FMT_DBL))
> && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
> if(preout==in){
> out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
> @@ -685,6 +686,7 @@ static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_co
> //FIXME packed doesn't need more than 1 chan here!
> swri_audio_convert(s->out_convert, out, conv_src, out_count);
> }
> +
> return out_count;
> }
>
> diff --git a/libswresample/swresample.h b/libswresample/swresample.h
> index 10eaebc..3f56758 100644
> --- a/libswresample/swresample.h
> +++ b/libswresample/swresample.h
> @@ -138,6 +138,7 @@
> */
>
> #define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
> +#define SWR_FLAG_CLIP 2 ///< Clip float/double output to -1.0..1.0
> //TODO use int resample ?
> //long term TODO can we enable this dynamically?
>
> diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h
> index bf0cec7..ab1e853 100644
> --- a/libswresample/swresample_internal.h
> +++ b/libswresample/swresample_internal.h
> @@ -206,14 +206,14 @@ int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFo
> void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
> enum AVSampleFormat out_fmt,
> enum AVSampleFormat in_fmt,
> - int channels);
> + int channels, int flags);
> void swri_audio_convert_init_arm(struct AudioConvert *ac,
> enum AVSampleFormat out_fmt,
> enum AVSampleFormat in_fmt,
> - int channels);
> + int channels, int flags);
> void swri_audio_convert_init_x86(struct AudioConvert *ac,
> enum AVSampleFormat out_fmt,
> enum AVSampleFormat in_fmt,
> - int channels);
> + int channels, int flags);
>
> #endif
> diff --git a/libswresample/x86/audio_convert_init.c b/libswresample/x86/audio_convert_init.c
> index 5e5e91d..e831a06 100644
> --- a/libswresample/x86/audio_convert_init.c
> +++ b/libswresample/x86/audio_convert_init.c
> @@ -36,11 +36,17 @@ PROTO4(_unpack_6ch_)
> av_cold void swri_audio_convert_init_x86(struct AudioConvert *ac,
> enum AVSampleFormat out_fmt,
> enum AVSampleFormat in_fmt,
> - int channels){
> + int channels, int flags){
> int mm_flags = av_get_cpu_flags();
>
> ac->simd_f= NULL;
>
> + if ( (flags & SWR_FLAG_CLIP)
> + && av_get_packed_sample_fmt(in_fmt) == AV_SAMPLE_FMT_FLT
> + && av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_FLT) {
> + return;
> + }
> +
> //FIXME add memcpy case
>
> #define MULTI_CAPS_FUNC(flag, cap) \
Wouldn't it be easier to just run float clipping as a postprocessing
step, and not bother with e.g. double->float clipping functions?
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