From aaaf1635c058dd17bf977356f0deb10b009bc059 Mon Sep 17 00:00:00 2001 From: Michael Niedermayer Date: Thu, 17 Jun 2004 15:43:23 +0000 Subject: polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample filters Originally committed as revision 3228 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/resample.c | 153 +++++++++----------------------------------------- 1 file changed, 25 insertions(+), 128 deletions(-) (limited to 'libavcodec/resample.c') diff --git a/libavcodec/resample.c b/libavcodec/resample.c index f6f0bf42b9..b43b4daa5a 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -24,103 +24,17 @@ #include "avcodec.h" -typedef struct { - /* fractional resampling */ - uint32_t incr; /* fractional increment */ - uint32_t frac; - int last_sample; - /* integer down sample */ - int iratio; /* integer divison ratio */ - int icount, isum; - int inv; -} ReSampleChannelContext; +struct AVResampleContext; struct ReSampleContext { - ReSampleChannelContext channel_ctx[2]; + struct AVResampleContext *resample_context; + short *temp[2]; + int temp_len; float ratio; /* channel convert */ int input_channels, output_channels, filter_channels; }; - -#define FRAC_BITS 16 -#define FRAC (1 << FRAC_BITS) - -static void init_mono_resample(ReSampleChannelContext *s, float ratio) -{ - ratio = 1.0 / ratio; - s->iratio = (int)floorf(ratio); - if (s->iratio == 0) - s->iratio = 1; - s->incr = (int)((ratio / s->iratio) * FRAC); - s->frac = FRAC; - s->last_sample = 0; - s->icount = s->iratio; - s->isum = 0; - s->inv = (FRAC / s->iratio); -} - -/* fractional audio resampling */ -static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) -{ - unsigned int frac, incr; - int l0, l1; - short *q, *p, *pend; - - l0 = s->last_sample; - incr = s->incr; - frac = s->frac; - - p = input; - pend = input + nb_samples; - q = output; - - l1 = *p++; - for(;;) { - /* interpolate */ - *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS; - frac = frac + s->incr; - while (frac >= FRAC) { - frac -= FRAC; - if (p >= pend) - goto the_end; - l0 = l1; - l1 = *p++; - } - } - the_end: - s->last_sample = l1; - s->frac = frac; - return q - output; -} - -static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) -{ - short *q, *p, *pend; - int c, sum; - - p = input; - pend = input + nb_samples; - q = output; - - c = s->icount; - sum = s->isum; - - for(;;) { - sum += *p++; - if (--c == 0) { - *q++ = (sum * s->inv) >> FRAC_BITS; - c = s->iratio; - sum = 0; - } - if (p >= pend) - break; - } - s->isum = sum; - s->icount = c; - return q - output; -} - /* n1: number of samples */ static void stereo_to_mono(short *output, short *input, int n1) { @@ -210,31 +124,6 @@ static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) } } -static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) -{ - short *buf1; - short *buftmp; - - buf1= (short*)av_malloc( nb_samples * sizeof(short) ); - - /* first downsample by an integer factor with averaging filter */ - if (s->iratio > 1) { - buftmp = buf1; - nb_samples = integer_downsample(s, buftmp, input, nb_samples); - } else { - buftmp = input; - } - - /* then do a fractional resampling with linear interpolation */ - if (s->incr != FRAC) { - nb_samples = fractional_resample(s, output, buftmp, nb_samples); - } else { - memcpy(output, buftmp, nb_samples * sizeof(short)); - } - av_free(buf1); - return nb_samples; -} - ReSampleContext *audio_resample_init(int output_channels, int input_channels, int output_rate, int input_rate) { @@ -271,16 +160,13 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels, if(s->filter_channels>2) s->filter_channels = 2; - for(i=0;ifilter_channels;i++) { - init_mono_resample(&s->channel_ctx[i], s->ratio); - } + s->resample_context= av_resample_init(output_rate, input_rate); + return s; } /* resample audio. 'nb_samples' is the number of input samples */ /* XXX: optimize it ! */ -/* XXX: do it with polyphase filters, since the quality here is - HORRIBLE. Return the number of samples available in output */ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) { int i, nb_samples1; @@ -296,8 +182,11 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl } /* XXX: move those malloc to resample init code */ - bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) ); - bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) ); + for(i=0; ifilter_channels; i++){ + bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); + memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); + buftmp2[i] = bufin[i] + s->temp_len; + } /* make some zoom to avoid round pb */ lenout= (int)(nb_samples * s->ratio) + 16; @@ -306,27 +195,32 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl if (s->input_channels == 2 && s->output_channels == 1) { - buftmp2[0] = bufin[0]; buftmp3[0] = output; stereo_to_mono(buftmp2[0], input, nb_samples); } else if (s->output_channels >= 2 && s->input_channels == 1) { - buftmp2[0] = input; buftmp3[0] = bufout[0]; + memcpy(buftmp2[0], input, nb_samples*sizeof(short)); } else if (s->output_channels >= 2) { - buftmp2[0] = bufin[0]; - buftmp2[1] = bufin[1]; buftmp3[0] = bufout[0]; buftmp3[1] = bufout[1]; stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); } else { - buftmp2[0] = input; buftmp3[0] = output; + memcpy(buftmp2[0], input, nb_samples*sizeof(short)); } + nb_samples += s->temp_len; + /* resample each channel */ nb_samples1 = 0; /* avoid warning */ for(i=0;ifilter_channels;i++) { - nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); + int consumed; + int is_last= i+1 == s->filter_channels; + + nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last); + s->temp_len= nb_samples - consumed; + s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short)); + memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short)); } if (s->output_channels == 2 && s->input_channels == 1) { @@ -347,5 +241,8 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl void audio_resample_close(ReSampleContext *s) { + av_resample_close(s->resample_context); + av_freep(&s->temp[0]); + av_freep(&s->temp[1]); av_free(s); } -- cgit v1.2.3