X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libswresample%2Faudioconvert.c;h=c53135677bc948380a71ccca0a35f50f3242bd08;hb=bd1d975cd098062a15da85f1384883e8fb6c526f;hp=96e5083c8792c3811c5be2dadab791d38c06da9d;hpb=be2e211dce2964f922e5dee54a7569ebd6efb0da;p=ffmpeg diff --git a/libswresample/audioconvert.c b/libswresample/audioconvert.c index 96e5083c879..c53135677bc 100644 --- a/libswresample/audioconvert.c +++ b/libswresample/audioconvert.c @@ -32,41 +32,39 @@ #include "audioconvert.h" -typedef void (conv_func_type)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end); - -struct AudioConvert { - int channels; - conv_func_type *conv_f; - const int *ch_map; - uint8_t silence[8]; ///< silence input sample -}; - #define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt //FIXME rounding ? #define CONV_FUNC(ofmt, otype, ifmt, expr)\ static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)\ {\ - do{\ + 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);\ + }\ + while(po < end){\ + *(otype*)po = expr; pi += is; po += os;\ + }\ } //FIXME put things below under ifdefs so we do not waste space for cases no codec will need CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24) -CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) +CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0f/ (1<<7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16) -CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) +CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0f/ (1<<15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi) -CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31))) +CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0f/ (1U<<31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31))) CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15)))) @@ -109,24 +107,55 @@ 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), }; +static void cpy1(uint8_t **dst, const uint8_t **src, int len){ + memcpy(*dst, *src, len); +} +static void cpy2(uint8_t **dst, const uint8_t **src, int len){ + memcpy(*dst, *src, 2*len); +} +static void cpy4(uint8_t **dst, const uint8_t **src, int len){ + memcpy(*dst, *src, 4*len); +} +static void cpy8(uint8_t **dst, const uint8_t **src, int len){ + memcpy(*dst, *src, 8*len); +} + AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx; - conv_func_type *f = fmt_pair_to_conv_functions[out_fmt + AV_SAMPLE_FMT_NB*in_fmt]; + conv_func_type *f = fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt) + AV_SAMPLE_FMT_NB*av_get_packed_sample_fmt(in_fmt)]; if (!f) return NULL; ctx = av_mallocz(sizeof(*ctx)); if (!ctx) return NULL; + + if(channels == 1){ + in_fmt = av_get_planar_sample_fmt( in_fmt); + out_fmt = av_get_planar_sample_fmt(out_fmt); + } + ctx->channels = channels; ctx->conv_f = f; ctx->ch_map = ch_map; - if (in_fmt == AV_SAMPLE_FMT_U8) + if (in_fmt == AV_SAMPLE_FMT_U8 || in_fmt == AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence)); + + if(out_fmt == in_fmt && !ch_map) { + switch(av_get_bytes_per_sample(in_fmt)){ + case 1:ctx->simd_f = cpy1; break; + case 2:ctx->simd_f = cpy2; break; + case 4:ctx->simd_f = cpy4; break; + case 8:ctx->simd_f = cpy8; break; + } + } + + if(HAVE_YASM && HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels); + return ctx; } @@ -138,21 +167,40 @@ void swri_audio_convert_free(AudioConvert **ctx) int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch; + int off=0; + const int os= (out->planar ? 1 :out->ch_count) *out->bps; av_assert0(ctx->channels == out->ch_count); //FIXME optimize common cases + if(ctx->simd_f && !ctx->ch_map){ + off = len/16 * 16; + av_assert1(off>=0); + av_assert1(off<=len); + if(off>0){ + if(out->planar == in->planar){ + int planes = out->planar ? out->ch_count : 1; + for(ch=0; chsimd_f(out->ch+ch, in->ch+ch, off * (out->planar ? 1 :out->ch_count)); + } + }else{ + ctx->simd_f(out->ch, in->ch, off); + } + } + if(off == len) + return 0; + } + for(ch=0; chchannels; ch++){ const int ich= ctx->ch_map ? ctx->ch_map[ch] : ch; const int is= ich < 0 ? 0 : (in->planar ? 1 : in->ch_count) * in->bps; - const int os= (out->planar ? 1 :out->ch_count) *out->bps; const uint8_t *pi= ich < 0 ? ctx->silence : in->ch[ich]; uint8_t *po= out->ch[ch]; uint8_t *end= po + os*len; if(!po) continue; - ctx->conv_f(po, pi, is, os, end); + ctx->conv_f(po+off*os, pi+off*is, is, os, end); } return 0; }