diff options
| -rw-r--r-- | Alc/bformatdec.c | 164 |
1 files changed, 99 insertions, 65 deletions
diff --git a/Alc/bformatdec.c b/Alc/bformatdec.c index 356423ce..20e19bb6 100644 --- a/Alc/bformatdec.c +++ b/Alc/bformatdec.c @@ -122,6 +122,12 @@ enum FreqBand { FB_Max }; +static const ALfloat SquarePoints[4][3] = { + { -0.707106781f, 0.0f, -0.707106781f }, + { 0.707106781f, 0.0f, -0.707106781f }, + { -0.707106781f, 0.0f, 0.707106781f }, + { 0.707106781f, 0.0f, 0.707106781f }, +}; static const ALfloat SquareMatrix[4][FB_Max][MAX_AMBI_COEFFS] = { { { 0.353553f, 0.204094f, 0.0f, 0.204094f }, { 0.25f, 0.204094f, 0.0f, 0.204094f } }, { { 0.353553f, -0.204094f, 0.0f, 0.204094f }, { 0.25f, -0.204094f, 0.0f, 0.204094f } }, @@ -180,12 +186,9 @@ static void init_bformatdec(void) for(i = 0;i < COUNTOF(CubePoints);i++) CalcDirectionCoeffs(CubePoints[i], 0.0f, CubeEncoder[i]); - CalcXYZCoeffs(-0.707106781f, 0.0f, -0.707106781f, 0.0f, SquareEncoder[0]); - CalcXYZCoeffs( 0.707106781f, 0.0f, -0.707106781f, 0.0f, SquareEncoder[1]); - CalcXYZCoeffs(-0.707106781f, 0.0f, 0.707106781f, 0.0f, SquareEncoder[2]); - CalcXYZCoeffs( 0.707106781f, 0.0f, 0.707106781f, 0.0f, SquareEncoder[3]); - - for(i = 0;i < 4;i++) + for(i = 0;i < COUNTOF(SquarePoints);i++) + CalcDirectionCoeffs(SquarePoints[i], 0.0f, SquareEncoder[i]); + for(i = 0;i < COUNTOF(SquarePoints);i++) { /* Remove the skipped height-related coefficients for 2D rendering. */ SquareEncoder[i][2] = SquareEncoder[i][3]; @@ -227,9 +230,7 @@ typedef struct BFormatDec { struct { BandSplitter XOver[4]; - const ALfloat (*restrict Matrix)[FB_Max][MAX_AMBI_COEFFS]; - const ALfloat (*restrict Encoder)[MAX_AMBI_COEFFS]; - ALuint NumChannels; + ALfloat Gains[4][MAX_OUTPUT_CHANNELS][FB_Max]; } UpSampler; ALuint NumChannels; @@ -282,7 +283,7 @@ void bformatdec_reset(BFormatDec *dec, const AmbDecConf *conf, ALuint chancount, const ALfloat *coeff_scale = UnitScale; ALfloat distgain[MAX_OUTPUT_CHANNELS]; ALfloat maxdist, ratio; - ALuint i; + ALuint i, j, k; al_free(dec->Samples); dec->Samples = NULL; @@ -307,18 +308,42 @@ void bformatdec_reset(BFormatDec *dec, const AmbDecConf *conf, ALuint chancount, ratio = 400.0f / (ALfloat)srate; for(i = 0;i < 4;i++) bandsplit_init(&dec->UpSampler.XOver[i], ratio); + memset(dec->UpSampler.Gains, 0, sizeof(dec->UpSampler.Gains)); if((conf->ChanMask&AMBI_PERIPHONIC_MASK)) { - dec->UpSampler.Matrix = CubeMatrix; - dec->UpSampler.Encoder = (const ALfloat(*)[MAX_AMBI_COEFFS])CubeEncoder; - dec->UpSampler.NumChannels = 8; + /* Combine the matrices that do the in->virt and virt->out conversions + * so we get a single in->out conversion. + */ + for(i = 0;i < 4;i++) + { + for(j = 0;j < dec->NumChannels;j++) + { + ALfloat *gains = dec->UpSampler.Gains[i][j]; + for(k = 0;k < COUNTOF(CubeMatrix);k++) + { + gains[FB_HighFreq] += CubeMatrix[k][FB_HighFreq][i]*CubeEncoder[k][j]; + gains[FB_LowFreq] += CubeMatrix[k][FB_LowFreq][i]*CubeEncoder[k][j]; + } + } + } + dec->Periphonic = AL_TRUE; } else { - dec->UpSampler.Matrix = SquareMatrix; - dec->UpSampler.Encoder = (const ALfloat(*)[MAX_AMBI_COEFFS])SquareEncoder; - dec->UpSampler.NumChannels = 4; + for(i = 0;i < 4;i++) + { + for(j = 0;j < dec->NumChannels;j++) + { + ALfloat *gains = dec->UpSampler.Gains[i][j]; + for(k = 0;k < COUNTOF(SquareMatrix);k++) + { + gains[FB_HighFreq] += SquareMatrix[k][FB_HighFreq][i]*SquareEncoder[k][j]; + gains[FB_LowFreq] += SquareMatrix[k][FB_LowFreq][i]*SquareEncoder[k][j]; + } + } + } + dec->Periphonic = AL_FALSE; } @@ -559,51 +584,48 @@ void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BU void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat (*restrict InSamples)[BUFFERSIZE], ALuint InChannels, ALuint SamplesToDo) { - ALuint i, j, k; - - /* First, split the first-order components into low and high frequency - * bands. This assumes SamplesHF and SamplesLF have enough space for first- - * order content (to which, this up-sampler is only used with second-order - * or higher decoding, so it will). - */ - for(i = 0;i < InChannels;i++) - bandsplit_process(&dec->UpSampler.XOver[i], dec->SamplesHF[i], dec->SamplesLF[i], - InSamples[i], SamplesToDo); + ALuint in, i, j, k; /* This up-sampler is very simplistic. It essentially decodes the first- * order content to a square channel array (or cube if height is desired), - * then encodes those points onto the higher order soundfield. + * then encodes those points onto the higher order soundfield. The decoder + * and encoder matrices have been combined to directly convert each input + * channel to the output, without the need for storing the virtual channel + * array. */ - for(k = 0;k < dec->UpSampler.NumChannels;k++) + for(in = 0;in < InChannels;in++) { - memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat)); - MixMatrixRow(dec->ChannelMix, dec->UpSampler.Matrix[k][FB_HighFreq], - dec->SamplesHF, InChannels, SamplesToDo); - MixMatrixRow(dec->ChannelMix, dec->UpSampler.Matrix[k][FB_LowFreq], - dec->SamplesLF, InChannels, SamplesToDo); - + /* First, split the first-order components into low and high frequency + * bands. + */ + bandsplit_process(&dec->UpSampler.XOver[in], + dec->Samples[FB_HighFreq], dec->Samples[FB_LowFreq], + InSamples[in], SamplesToDo + ); + + /* Now write each band to the output. */ for(j = 0;j < dec->NumChannels;j++) { - ALfloat gain = dec->UpSampler.Encoder[k][j]; - if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) - continue; - for(i = 0;i < SamplesToDo;i++) - OutBuffer[j][i] += dec->ChannelMix[i] * gain; + for(k = 0;k < FB_Max;k++) + { + ALfloat gain = dec->UpSampler.Gains[in][j][k]; + if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) + continue; + + for(i = 0;i < SamplesToDo;i++) + OutBuffer[j][i] += dec->Samples[k][i] * gain; + } } } } typedef struct AmbiUpsampler { - alignas(16) ALfloat SamplesHF[4][BUFFERSIZE]; - alignas(16) ALfloat SamplesLF[4][BUFFERSIZE]; - - alignas(16) ALfloat ChannelMix[BUFFERSIZE]; + alignas(16) ALfloat Samples[FB_Max][BUFFERSIZE]; BandSplitter XOver[4]; - ALfloat Gains[8][MAX_OUTPUT_CHANNELS]; - ALuint NumChannels; + ALfloat Gains[4][MAX_OUTPUT_CHANNELS][FB_Max]; } AmbiUpsampler; AmbiUpsampler *ambiup_alloc() @@ -619,41 +641,53 @@ void ambiup_free(struct AmbiUpsampler *ambiup) void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device) { + ALfloat gains[8][MAX_OUTPUT_CHANNELS]; ALfloat ratio; - ALuint i; + ALuint i, j, k; ratio = 400.0f / (ALfloat)device->Frequency; for(i = 0;i < 4;i++) bandsplit_init(&ambiup->XOver[i], ratio); - ambiup->NumChannels = COUNTOF(CubePoints); - for(i = 0;i < ambiup->NumChannels;i++) - ComputePanningGains(device->Dry, CubeEncoder[i], 1.0f, ambiup->Gains[i]); + for(i = 0;i < COUNTOF(CubePoints);i++) + ComputePanningGains(device->Dry, CubeEncoder[i], 1.0f, gains[i]); + + memset(ambiup->Gains, 0, sizeof(ambiup->Gains)); + for(i = 0;i < 4;i++) + { + for(j = 0;j < device->Dry.NumChannels;j++) + { + for(k = 0;k < COUNTOF(CubePoints);k++) + { + ambiup->Gains[i][j][FB_HighFreq] += CubeMatrix[k][FB_HighFreq][i]*gains[k][j]; + ambiup->Gains[i][j][FB_LowFreq] += CubeMatrix[k][FB_LowFreq][i]*gains[k][j]; + } + } + } } void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALuint OutChannels, ALfloat (*restrict InSamples)[BUFFERSIZE], ALuint SamplesToDo) { - ALuint i, j, k; + ALuint in, i, j, k; - for(i = 0;i < 4;i++) - bandsplit_process(&ambiup->XOver[i], ambiup->SamplesHF[i], ambiup->SamplesLF[i], - InSamples[i], SamplesToDo); - - for(k = 0;k < ambiup->NumChannels;k++) + for(in = 0;in < 4;in++) { - memset(ambiup->ChannelMix, 0, SamplesToDo*sizeof(ALfloat)); - MixMatrixRow(ambiup->ChannelMix, CubeMatrix[k][FB_HighFreq], - ambiup->SamplesHF, 4, SamplesToDo); - MixMatrixRow(ambiup->ChannelMix, CubeMatrix[k][FB_LowFreq], - ambiup->SamplesLF, 4, SamplesToDo); + bandsplit_process(&ambiup->XOver[in], + ambiup->Samples[FB_HighFreq], ambiup->Samples[FB_LowFreq], + InSamples[in], SamplesToDo + ); for(j = 0;j < OutChannels;j++) { - ALfloat gain = ambiup->Gains[k][j]; - if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) - continue; - for(i = 0;i < SamplesToDo;i++) - OutBuffer[j][i] += ambiup->ChannelMix[i] * gain; + for(k = 0;k < FB_Max;k++) + { + ALfloat gain = ambiup->Gains[in][j][k]; + if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) + continue; + + for(i = 0;i < SamplesToDo;i++) + OutBuffer[j][i] += ambiup->Samples[k][i] * gain; + } } } } |