title('y-int');
endfunction
+
+function decode_from_bit_stream(samname)
+ max_amp = 80;
+ bits_per_param = [6 1 8 8 4 1];
+ load vq;
+ k = 10;
+ dec_in_time = 1;
+ train = 0;
+ decimate = 4;
+ synth_phase = 1;
+
+ Am_out_name = sprintf("%s_am.out", samname);
+ Aw_out_name = sprintf("%s_aw.out", samname);
+ bit_stream_name = strcat(samname,".bit");
+ faw = fopen(Aw_out_name,"wb");
+ fam = fopen(Am_out_name,"wb");
+ faw = fopen(Aw_out_name,"wb");
+
+ Wo_out_name = sprintf("%s_Wo.out", samname);
+ fWo = fopen(Wo_out_name,"wb");
+
+ % read in bit stream and convert to ind_log[]
+
+ ind_log = [];
+ fbit = fopen(bit_stream_name, "rb");
+ bits_per_frame = sum(bits_per_param);
+ nind = length(bits_per_param);
+
+ [frame nread] = fread(fbit, sum(bits_per_param), "uchar");
+ while (nread == bits_per_frame)
+ % read a frame, convert to indexes
+
+ nbit = 1;
+ ind = [];
+ for i=1:nind
+ field = frame(nbit:nbit+bits_per_param(i)-1);
+ nbit += bits_per_param(i);
+ ind = [ind bits_to_index(field, bits_per_param(i))];
+ end
+ ind_log = [ind_log; ind];
+ [frame nread] = fread(fbit, sum(bits_per_param), "uchar");
+ endwhile
+ fclose(fbit);
+
+ % convert ind_log to modem params
+
+ frames = 4*length(ind_log);
+ model_ = zeros(frames, max_amp+2);
+ v = zeros(frames,1);
+
+ fdec = 1;
+ for f=1:4:frames
+ ind_Wo = ind_log(fdec,1);
+
+ Wo = decode_log_Wo(ind_Wo, 6);
+ L = floor(pi/Wo);
+ L = min([L max_amp-1]);
+ model_(f,1) = Wo;
+ model_(f,2) = L;
+
+ v1 = ind_log(fdec,2);
+ if (fdec+1) < length(ind_log)
+ v5 = ind_log(fdec+1,2);
+ else
+ v5 = 0;
+ end
+ v(f:f+3) = est_voicing_bits(v1, v5);
+
+ ind_vq = ind_log(fdec,3:5) + 1;
+ [dk_ D1_] = index_to_params(ind_vq, vq);
+ maskdB_ = params_to_mask(L, k, dk_, D1_);
+ Am_ = zeros(1,max_amp);
+ Am_ = 10 .^ (maskdB_(1:L)/20);
+ model_(f,3:(L+2)) = Am_;
+
+ fdec += 1;
+ end
+
+ % decoder loop -----------------------------------------------------
+
+ if train
+ % short circuit decoder
+ frames = 0;
+ end
+
+ % run post filter ahead of time so dec in time has post filtered frames to work with
+
+ for f=1:frames
+ model_(f,:) = post_filter(model_(f,:));
+ end
+
+ for f=1:frames
+ %printf("frame: %d\n", f);
+ L = min([model_(f,2) max_amp-1]);
+ Wo = model_(f,1);
+ Am_ = model_(f,3:(L+2));
+
+ maskdB_ = 20*log10(Am_);
+
+ if dec_in_time
+ % decimate mask samples in time
+
+ [maskdB_ Wo L] = decimate_frame_rate2(model_, decimate, f, frames);
+ model_(f,1) = Wo;
+ model_(f,2) = L;
+ end
+
+ Am_ = zeros(1,max_amp);
+ Am_(2:L) = 10 .^ (maskdB_(1:L-1)/20); % C array doesnt use A[0]
+ fwrite(fam, Am_, "float32");
+ fwrite(fWo, Wo, "float32");
+
+ if synth_phase
+
+ % synthesis phase spectra from magnitiude spectra using minimum phase techniques
+
+ fft_enc = 512;
+ model_(f,3:(L+2)) = 10 .^ (maskdB_(1:L)/20);
+ phase = determine_phase(model_, f);
+ assert(length(phase) == fft_enc);
+ Aw = zeros(1, fft_enc*2);
+ Aw(1:2:fft_enc*2) = cos(phase);
+ Aw(2:2:fft_enc*2) = -sin(phase);
+ fwrite(faw, Aw, "float32");
+ end
+ end
+
+ fclose(fam);
+ fclose(fWo);
+ if synth_phase
+ fclose(faw);
+ end
+
+ % save voicing file
+
+ v_out_name = sprintf("%s_v.txt", samname);
+ fv = fopen(v_out_name,"wt");
+ for f=1:length(v)
+ fprintf(fv,"%d\n", v(f));
+ end
+ fclose(fv);
+
+endfunction
+
+
+
+
+
+% decimate frame rate of mask, use linear interpolation in the log domain
+
+function [maskdB_ Wo L] = decimate_frame_rate2(model, decimate, f, frames)
+ max_amp = 80;
+
+ % determine frames that bracket the one we need to interp
+
+ left_f = decimate*floor((f-1)/decimate)+1;
+ right_f = left_f + decimate;
+ if right_f > frames
+ right_f = left_f;
+ end
+
+ % determine fraction of each frame to use
+
+ left_fraction = 1 - mod((f-1),decimate)/decimate;
+ right_fraction = 1 - left_fraction;
+
+ % printf("f: %d left_f: %d right_f: %d left_fraction: %3.2f right_fraction: %3.2f \n", f, left_f, right_f, left_fraction, right_fraction)
+
+ % fit splines to left and right masks
+
+ left_Wo = model(left_f,1);
+ left_L = min([model(left_f,2) max_amp]);
+ left_AmdB = 20*log10(model(left_f,3:(left_L+2)));
+ left_mask_sample_freqs_kHz = (1:left_L)*left_Wo*4/pi;
+
+ right_Wo = model(right_f,1);
+ right_L = min([model(right_f,2) max_amp]);
+ right_AmdB = 20*log10(model(right_f,3:(right_L+2)));
+ right_mask_sample_freqs_kHz = (1:right_L)*right_Wo*4/pi;
+
+ % printf(" right_Wo: %f left_Wo: %f right_L: %d left_L %d\n",right_Wo,left_Wo,right_L,left_L);
+ printf("%f %f\n", left_AmdB(left_L), right_AmdB(right_L));
+
+ maskdB_left_pp = splinefit(left_mask_sample_freqs_kHz, left_AmdB, left_L);
+ maskdB_right_pp = splinefit(right_mask_sample_freqs_kHz, right_AmdB, right_L);
+
+ % determine mask for left and right frames, sampling at Wo for this frame
+
+ Wo = left_fraction*left_Wo + right_fraction*right_Wo;
+ L = floor(pi/Wo);
+ %Wo = model(f,1); L = model(f,2);
+
+ mask_sample_freqs_kHz = (1:L)*Wo*4/pi;
+ maskdB_left = ppval(maskdB_left_pp, mask_sample_freqs_kHz);
+ maskdB_right = ppval(maskdB_right_pp, mask_sample_freqs_kHz);
+
+ maskdB_ = left_fraction*maskdB_left + right_fraction*maskdB_right;
+endfunction
+
+function amodel = post_filter(amodel)
+ max_amp = 80;
+
+ % post filter
+
+ L = min([amodel(2) max_amp-1]);
+ Wo = amodel(1);
+ Am_ = amodel(3:(L+2));
+ AmdB_ = 20*log10(Am_);
+ AmdB_pf = AmdB_*1.5;
+ AmdB_pf += max(AmdB_) - max(AmdB_pf);
+ amodel(3:(L+2)) = 10 .^ (AmdB_pf(1:L)/20);
+endfunction
+
+
+function index = encode_log_Wo(Wo, bits)
+ Wo_levels = 2.^bits;
+ Wo_min = 2*pi/160;
+ Wo_max = 2*pi/20;
+
+ norm = (log10(Wo) - log10(Wo_min))/(log10(Wo_max) - log10(Wo_min));
+ index = floor(Wo_levels * norm + 0.5);
+ index = max(index, 0);
+ index = min(index, Wo_levels-1);
+endfunction
+
+
+function Wo = decode_log_Wo(index, bits)
+ Wo_levels = 2.^bits;
+ Wo_min = 2*pi/160;
+ Wo_max = 2*pi/20;
+
+ step = (log10(Wo_max) - log10(Wo_min))/Wo_levels;
+ Wo = log10(Wo_min) + step*index;
+
+ Wo = 10 .^ Wo;
+endfunction
+
+% Given a matrix with indexes on each row, convert to a bit stream and
+% write to file. We only write every 4th frame due to DIT
+
+function write_bit_stream_file(fn, ind_log, bits_per_param)
+ fbit = fopen(fn,"wb");
+ decimate = 4;
+
+ % take a row of quantiser indexes, convert to bits, save to file
+
+ [frames nind] = size(ind_log);
+ for f=1:decimate:frames
+ frame_of_bits = [];
+ arow = ind_log(f,:);
+ for i=1:nind
+ %printf("i: %d bits_per_param: %d\n", i, bits_per_param(i));
+ some_bits = index_to_bits(arow(i), bits_per_param(i));
+ frame_of_bits = [frame_of_bits some_bits];
+ end
+ fwrite(fbit, frame_of_bits, "uchar");
+ end
+ fclose(fbit);
+endfunction
+
+
+% convert index to binary bits
+
+function bits = index_to_bits(value, numbits)
+ levels = 2.^numbits;
+ bits = zeros(1, numbits);
+ for b=1:numbits
+ bits(b) = bitand(value,2^(numbits-b)) != 0;
+ end
+end
+
+
+function value = bits_to_index(bits, numbits)
+ value = 2.^(numbits-1:-1:0) * bits;
+endfunction
+
+
+% determine 4 voicing bits based on 2 decimated voicing bits
+
+function [v] = est_voicing_bits(v1, v5)
+ if v1 == v5
+ v(1:4) = v1;
+ else
+ v(1:2) = v1;
+ v(3:4) = v5;
+ end
+endfunction
max_amp = 80;
k = 10;
decimate = 4;
- dec_in_freq = 1;
dec_in_time = 1;
+ dec_in_freq = 1;
+ decimate = 4;
synth_phase = 1;
vq_en = 1;
dk_log = []; D1_log = [];
model_ = zeros(frames, nc);
non_masked_m = zeros(frames,max_amp);
- if nargin == 1
- Am_out_name = sprintf("%s_am.out", samname);
- Aw_out_name = sprintf("%s_aw.out", samname);
- end
- if nargin >= 2
- Am_out_name = optional_Am_out_name;
- end
- if nargin >= 3
- Aw_out_name = optional_Aw_out_name;
- faw = fopen(Aw_out_name,"wb");
- end
-
- fam = fopen(Am_out_name,"wb");
- if synth_phase
- faw = fopen(Aw_out_name,"wb");
- end
-
- Wo_out_name = sprintf("%s_Wo.out", samname);
- fWo = fopen(Wo_out_name,"wb");
-
voicing_name = strcat(samname,"_pitche.txt");
voicing = zeros(1,frames);
if exist(voicing_name, "file") == 2
bit_stream_name = strcat(samname,".bit");
bits_per_param = [6 1 8 8 4 1];
write_bit_stream_file(bit_stream_name, ind_log, bits_per_param);
-
- % read in bit stream and convert to ind_log[]
-
- ind_log = [];
- fbit = fopen(bit_stream_name, "rb");
- bits_per_frame = sum(bits_per_param);
- nind = length(bits_per_param);
-
- [frame nread] = fread(fbit, sum(bits_per_param), "uchar");
- while (nread == bits_per_frame)
- % read a frame, convert to indexes
-
- nbit = 1;
- ind = [];
- for i=1:nind
- field = frame(nbit:nbit+bits_per_param(i)-1);
- nbit += bits_per_param(i);
- ind = [ind bits_to_index(field, bits_per_param(i))];
- end
- ind_log = [ind_log; ind];
- [frame nread] = fread(fbit, sum(bits_per_param), "uchar");
- endwhile
- fclose(fbit);
-
- % convert ind_log to modem params
-
- frames = 4*length(ind_log);
- model_ = zeros(frames, max_amp+2);
- v = zeros(frames,1);
-
- fdec = 1;
- for f=1:4:frames
- ind_Wo = ind_log(fdec,1);
-
- Wo = decode_log_Wo(ind_Wo, 6);
- L = floor(pi/Wo);
- L = min([L max_amp-1]);
- model_(f,1) = Wo;
- model_(f,2) = L;
-
- v1 = ind_log(fdec,2);
- if (fdec+1) < length(ind_log)
- v5 = ind_log(fdec+1,2);
- else
- v5 = 0;
- end
- v(f:f+3) = est_voicing_bits(v1, v5);
-
- ind_vq = ind_log(fdec,3:5) + 1;
- [dk_ D1_] = index_to_params(ind_vq, vq);
- maskdB_ = params_to_mask(L, k, dk_, D1_);
- Am_ = zeros(1,max_amp);
- Am_ = 10 .^ (maskdB_(1:L)/20);
- model_(f,3:(L+2)) = Am_;
-
- fdec += 1;
- end
-
- % decoder loop -----------------------------------------------------
-
- if train
- % short circuit decoder
- frames = 0;
- end
-
- % run post filter ahead of time so dec in time has post filtered frames to work with
-
- for f=1:frames
- model_(f,:) = post_filter(model_(f,:));
- end
-
- for f=1:frames
- %printf("frame: %d\n", f);
- L = min([model_(f,2) max_amp-1]);
- Wo = model_(f,1);
- Am_ = model_(f,3:(L+2));
-
- maskdB_ = 20*log10(Am_);
-
- if dec_in_time
- % decimate mask samples in time
-
- [maskdB_ Wo L] = decimate_frame_rate2(model_, decimate, f, frames);
- model_(f,1) = Wo;
- model_(f,2) = L;
- end
-
- Am_ = zeros(1,max_amp);
- Am_(2:L) = 10 .^ (maskdB_(1:L-1)/20); % C array doesnt use A[0]
- fwrite(fam, Am_, "float32");
- fwrite(fWo, Wo, "float32");
-
- if synth_phase
-
- % synthesis phase spectra from magnitiude spectra using minimum phase techniques
-
- fft_enc = 512;
- model_(f,3:(L+2)) = 10 .^ (maskdB_(1:L)/20);
- phase = determine_phase(model_, f);
- assert(length(phase) == fft_enc);
- Aw = zeros(1, fft_enc*2);
- Aw(1:2:fft_enc*2) = cos(phase);
- Aw(2:2:fft_enc*2) = -sin(phase);
- fwrite(faw, Aw, "float32");
- end
- end
-
- fclose(fam);
- fclose(fWo);
- if synth_phase
- fclose(faw);
- end
-
- if exist(voicing_name, "file") == 2
- % save voicing file
-
- v_out_name = sprintf("%s_v.txt", samname);
- fv = fopen(v_out_name,"wt");
- for f=1:length(v)
- fprintf(fv,"%d\n", v(f));
- end
- fclose(fv);
- end
-
- printf("\nsd_sum: %5.2f\n", sd_sum/frames);
- printf("\n");
-endfunction
-
-
-% decimate frame rate of mask, use linear interpolation in the log domain
-
-function [maskdB_ Wo L] = decimate_frame_rate2(model, decimate, f, frames)
- max_amp = 80;
-
- % determine frames that bracket the one we need to interp
-
- left_f = decimate*floor((f-1)/decimate)+1;
- right_f = left_f + decimate;
- if right_f > frames
- right_f = left_f;
- end
-
- % determine fraction of each frame to use
-
- left_fraction = 1 - mod((f-1),decimate)/decimate;
- right_fraction = 1 - left_fraction;
-
- % printf("f: %d left_f: %d right_f: %d left_fraction: %3.2f right_fraction: %3.2f \n", f, left_f, right_f, left_fraction, right_fraction)
-
- % fit splines to left and right masks
-
- left_Wo = model(left_f,1);
- left_L = min([model(left_f,2) max_amp]);
- left_AmdB = 20*log10(model(left_f,3:(left_L+2)));
- left_mask_sample_freqs_kHz = (1:left_L)*left_Wo*4/pi;
-
- right_Wo = model(right_f,1);
- right_L = min([model(right_f,2) max_amp]);
- right_AmdB = 20*log10(model(right_f,3:(right_L+2)));
- right_mask_sample_freqs_kHz = (1:right_L)*right_Wo*4/pi;
-
- % printf(" right_Wo: %f left_Wo: %f right_L: %d left_L %d\n",right_Wo,left_Wo,right_L,left_L);
- printf("%f %f\n", left_AmdB(left_L), right_AmdB(right_L));
-
- maskdB_left_pp = splinefit(left_mask_sample_freqs_kHz, left_AmdB, left_L);
- maskdB_right_pp = splinefit(right_mask_sample_freqs_kHz, right_AmdB, right_L);
-
- % determine mask for left and right frames, sampling at Wo for this frame
-
- Wo = left_fraction*left_Wo + right_fraction*right_Wo;
- L = floor(pi/Wo);
- %Wo = model(f,1); L = model(f,2);
-
- mask_sample_freqs_kHz = (1:L)*Wo*4/pi;
- maskdB_left = ppval(maskdB_left_pp, mask_sample_freqs_kHz);
- maskdB_right = ppval(maskdB_right_pp, mask_sample_freqs_kHz);
-
- maskdB_ = left_fraction*maskdB_left + right_fraction*maskdB_right;
-endfunction
-
-function amodel = post_filter(amodel)
- max_amp = 80;
-
- % post filter
-
- L = min([amodel(2) max_amp-1]);
- Wo = amodel(1);
- Am_ = amodel(3:(L+2));
- AmdB_ = 20*log10(Am_);
- AmdB_pf = AmdB_*1.5;
- AmdB_pf += max(AmdB_) - max(AmdB_pf);
- amodel(3:(L+2)) = 10 .^ (AmdB_pf(1:L)/20);
-endfunction
-
-
-function index = encode_log_Wo(Wo, bits)
- Wo_levels = 2.^bits;
- Wo_min = 2*pi/160;
- Wo_max = 2*pi/20;
-
- norm = (log10(Wo) - log10(Wo_min))/(log10(Wo_max) - log10(Wo_min));
- index = floor(Wo_levels * norm + 0.5);
- index = max(index, 0);
- index = min(index, Wo_levels-1);
-endfunction
-
-
-function Wo = decode_log_Wo(index, bits)
- Wo_levels = 2.^bits;
- Wo_min = 2*pi/160;
- Wo_max = 2*pi/20;
-
- step = (log10(Wo_max) - log10(Wo_min))/Wo_levels;
- Wo = log10(Wo_min) + step*index;
-
- Wo = 10 .^ Wo;
-endfunction
-
-% Given a matrix with indexes on each row, convert to a bit stream and
-% write to file. We only write every 4th frame due to DIT
-
-function write_bit_stream_file(fn, ind_log, bits_per_param)
- fbit = fopen(fn,"wb");
- decimate = 4;
-
- % take a row of quantiser indexes, convert to bits, save to file
-
- [frames nind] = size(ind_log);
- for f=1:decimate:frames
- frame_of_bits = [];
- arow = ind_log(f,:);
- for i=1:nind
- %printf("i: %d bits_per_param: %d\n", i, bits_per_param(i));
- some_bits = index_to_bits(arow(i), bits_per_param(i));
- frame_of_bits = [frame_of_bits some_bits];
- end
- fwrite(fbit, frame_of_bits, "uchar");
- end
- fclose(fbit);
-endfunction
-
-
-% convert index to binary bits
-
-function bits = index_to_bits(value, numbits)
- levels = 2.^numbits;
- bits = zeros(1, numbits);
- for b=1:numbits
- bits(b) = bitand(value,2^(numbits-b)) != 0;
- end
-end
-
-
-function value = bits_to_index(bits, numbits)
- value = 2.^(numbits-1:-1:0) * bits;
-endfunction
-
-
-% determine 4 voicing bits based on 2 decimated voicing bits
-
-function [v] = est_voicing_bits(v1, v5)
- if v1 == v5
- v(1:4) = v1;
- else
- v(1:2) = v1;
- v(3:4) = v5;
- end
+ decode_from_bit_stream(samname);
endfunction
+
projects / freetel-svn-tracking.git / commitdiff