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#if !defined(__RESAMPLE_H_) #define __RESAMPLE_H_ class CResample { protected: typedef struct { /* fractional resampling */ UINT incr; /* fractional increment */ UINT frac; int last_sample; /* integer down sample */ int iratio; /* integer divison ratio */ int icount, isum; int inv; } ReSampleChannelContext; typedef struct ReSampleContext { ReSampleChannelContext channel_ctx[2]; float ratio; /* channel convert */ int input_channels, output_channels, filter_channels; } ReSampleContext; ReSampleContext m_context; protected: static void init_mono_resample(ReSampleChannelContext *s, float ratio); static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples); static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples); static void stereo_to_mono(short *output, short *input, int n1); static void mono_to_stereo(short *output, short *input, int n1); static void stereo_split(short *output1, short *output2, short *input, int n); static void stereo_mux(short *output, short *input1, short *input2, int n); 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); public: CResample(); virtual ~CResample(); BOOL audio_resample_init(int output_channels, int input_channels, int output_rate, int input_rate); int audio_resample(short *output, short *input, int nb_samples); static int mono_8bit_to_16bit(short* lp16bits, unsigned char* lp8bits, int len); static int mono_16bit_to_8bit(unsigned char* lp8bits, short* lp16bits, int len); }; #endif // !defined(__RESAMPLE_H_) ========================================================================= ========================================================================= #include "..\\stdafx.h" #include "Resample.h" #include <math.h> #include <stdlib.h> #include <stdio.h> #define FRAC_BITS 16 #define FRAC (1 << FRAC_BITS) #define av_free(p) {if(p) free(p);} #define av_malloc(size) malloc(size) ////////////////////////////////////////////////////////////////////// // Construction/Destruction ////////////////////////////////////////////////////////////////////// CResample::CResample() { } CResample::~CResample() { } //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// int CResample::mono_8bit_to_16bit(short* lp16bits, unsigned char* lp8bits, int len) { int i=0; for(i=0; i<len; i++) { *lp16bits++ = ((*lp8bits++) -128) << 8; } return i<<1; } //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// void CResample::init_mono_resample(ReSampleChannelContext *s, float ratio) { ratio = (float)(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 */ int CResample::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; } int CResample::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 */ void CResample::stereo_to_mono(short *output, short *input, int n1) { short *p, *q; int n = n1; p = input; q = output; while (n >= 4) { q[0] = (p[0] + p[1]) >> 1; q[1] = (p[2] + p[3]) >> 1; q[2] = (p[4] + p[5]) >> 1; q[3] = (p[6] + p[7]) >> 1; q += 4; p += 8; n -= 4; } while (n > 0) { q[0] = (p[0] + p[1]) >> 1; q++; p += 2; n--; } } /* n1: number of samples */ void CResample::mono_to_stereo(short *output, short *input, int n1) { short *p, *q; int n = n1; int v; p = input; q = output; while (n >= 4) { v = p[0]; q[0] = v; q[1] = v; v = p[1]; q[2] = v; q[3] = v; v = p[2]; q[4] = v; q[5] = v; v = p[3]; q[6] = v; q[7] = v; q += 8; p += 4; n -= 4; } while (n > 0) { v = p[0]; q[0] = v; q[1] = v; q += 2; p += 1; n--; } } /* XXX: should use more abstract 'N' channels system */ void CResample::stereo_split(short *output1, short *output2, short *input, int n) { int i; for(i=0;i<n;i++) { *output1++ = *input++; *output2++ = *input++; } } void CResample::stereo_mux(short *output, short *input1, short *input2, int n) { int i; for(i=0;i<n;i++) { *output++ = *input1++; *output++ = *input2++; } } void CResample::ac3_5p1_mux(short *output, short *input1, short *input2, int n) { int i; short l,r; for(i=0;i<n;i++) { l=*input1++; r=*input2++; *output++ = l; /* left */ *output++ = (l/2)+(r/2); /* center */ *output++ = r; /* right */ *output++ = 0; /* left surround */ *output++ = 0; /* right surroud */ *output++ = 0; /* low freq */ } } int CResample::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; } BOOL CResample::audio_resample_init(int output_channels, int input_channels, int output_rate, int input_rate) { int i; if ( input_channels > 2) { printf("Resampling with input channels greater than 2 unsupported."); return FALSE; } memset(&m_context, 0, sizeof(ReSampleContext)); m_context.ratio = (float)output_rate / (float)input_rate; m_context.input_channels = input_channels; m_context.output_channels = output_channels; m_context.filter_channels = m_context.input_channels; if (m_context.output_channels < m_context.filter_channels) m_context.filter_channels = m_context.output_channels; /* * ac3 output is the only case where filter_channels could be greater than 2. * input channels can't be greater than 2, so resample the 2 channels and then * expand to 6 channels after the resampling. */ if(m_context.filter_channels>2) m_context.filter_channels = 2; for(i=0;i<m_context.filter_channels;i++) { init_mono_resample(&m_context.channel_ctx[i], m_context.ratio); } return TRUE; } /* 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 CResample::audio_resample(short *output, short *input, int nb_samples) { int i, nb_samples1; short *bufin[2]; short *bufout[2]; short *buftmp2[2], *buftmp3[2]; int lenout; if (m_context.input_channels == m_context.output_channels && m_context.ratio == 1.0) { /* nothing to do */ memcpy(output, input, nb_samples * m_context.input_channels * sizeof(short)); return nb_samples; } /* 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) ); /* make some zoom to avoid round pb */ lenout= (int)(nb_samples * m_context.ratio) + 16; bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); if (m_context.input_channels == 2 && m_context.output_channels == 1) { buftmp2[0] = bufin[0]; buftmp3[0] = output; stereo_to_mono(buftmp2[0], input, nb_samples); } else if (m_context.output_channels >= 2 && m_context.input_channels == 1) { buftmp2[0] = input; buftmp3[0] = bufout[0]; } else if (m_context.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; } /* resample each channel */ nb_samples1 = 0; /* avoid warning */ for(i=0;i<m_context.filter_channels;i++) { nb_samples1 = mono_resample(&m_context.channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); } if (m_context.output_channels == 2 && m_context.input_channels == 1) { mono_to_stereo(output, buftmp3[0], nb_samples1); } else if (m_context.output_channels == 2) { stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); } else if (m_context.output_channels == 6) { ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); } av_free(bufin[0]); av_free(bufin[1]); av_free(bufout[0]); av_free(bufout[1]); return nb_samples1; } int CResample::mono_16bit_to_8bit(unsigned char* lp8bits, short* lp16bits, int len) { int i=0; for(i=0; i<len; i++) { *lp8bits++ = ((*lp16bits++) >> 8) + 128; } return i>>1; } |
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