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function x = iAACoder3(AACSeq3, fNameOut)
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%Implementation of AAC decoder
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% Usage x = iAACoder3(AACSeq3, fNameOut), where:
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% Inputs
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% - fNameOut is the filename and path of the file that will be
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% written after decoding
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% - AACSeq3 is an array of structs containing K structs, where K is
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% the number of computed frames. Every struct of the array consists
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% of:
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% * a frameType,
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% * a winType,
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% * chl.TNScoeffs which are the quantized TNS coefficients of
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% this frame's left channel,
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% * chr.TNScoeffs which are the quantized TNS coefficients of
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% this frame's right channel,
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% * chl.T which are the psychoacoustic thresholds of this frame's
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% left channel,
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% * chr.T which are the psychoacoustic thresholds of this frame's
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% right channel,
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% * chl.G which are the quantized global gains of this frame's
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% left channel,
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% * chr.G which are the quantized global gains of this frame's
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% right channel,
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% * chl.sfc which is the Huffman encoded sfc sequence of this
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% frame's left channel,
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% * chr.sfc which is the Huffman encoded sfc sequence of this
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% frame's right channel,
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% * chl.stream which is the Huffman encoded quantized MDCT
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% sequence of this frame's left channel,
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% * chr.stream which is the Huffman encoded quantized MDCT
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% sequence of this frame's right channel,
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% * chl.codebook which is the Huffman codebook used for this
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% frame's left channel
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% * chr.codebook which is the Huffman codebook used for this
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% frame's right channel
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%
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% Output
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% - x is an array containing the decoded audio samples
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% Initializes an array to hold the decoded samples
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decodedAudio(1024 * (length(AACSeq3) + 1), 2) = 0;
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% Initializes an array to hold both audio channels
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frameF(1024, 2) = 0;
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% Decodes audio file
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huffLUT = loadLUT();
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for i = 0:length(AACSeq3) - 1
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currFrameStart = i * 1024 + 1;
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currFrameStop = currFrameStart + 2047;
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if i < 2
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frameF(:, 1) = iTNS(AACSeq3(i + 1).chl.stream, ...
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AACSeq3(i+1).frameType, ...
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AACSeq3(i + 1).chl.TNScoeffs);
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frameF(:, 2) = iTNS(AACSeq3(i + 1).chr.stream, ...
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AACSeq3(i+1).frameType, ...
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AACSeq3(i + 1).chr.TNScoeffs);
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frameT = iFilterbank(frameF, AACSeq3(i+1).frameType, AACSeq3(i+1).winType);
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else
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% SL = decodeHuff(AACSeq3(i + 1).chl.stream, ...
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% AACSeq3(i + 1).chl.codebook, huffLUT);
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% SR = decodeHuff(AACSeq3(i + 1).chr.stream, ...
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% AACSeq3(i + 1).chr.codebook, huffLUT);
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%
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% sfcL = decodeHuff(AACSeq3(i + 1).chl.sfc, 12, huffLUT);
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% sfcR = decodeHuff(AACSeq3(i + 1).chr.sfc, 12, huffLUT);
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frameF(:, 1) = iAACquantizer(AACSeq3(i + 1).chl.stream, ...
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AACSeq3(i + 1).chl.sfc, ...
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AACSeq3(i + 1).chl.G, ...
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AACSeq3(i+1).frameType);
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frameF(:, 2) = iAACquantizer(AACSeq3(i + 1).chr.stream, ...
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AACSeq3(i + 1).chr.sfc, ...
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AACSeq3(i + 1).chr.G, ...
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AACSeq3(i+1).frameType);
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frameF(:, 1) = iTNS(frameF(:, 1), ...
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AACSeq3(i+1).frameType, ...
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AACSeq3(i + 1).chl.TNScoeffs);
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frameF(:, 2) = iTNS(frameF(:, 2), ...
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AACSeq3(i+1).frameType, ...
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AACSeq3(i + 1).chr.TNScoeffs);
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frameT = iFilterbank(frameF, AACSeq3(i+1).frameType, AACSeq3(i+1).winType);
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end
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decodedAudio(currFrameStart:currFrameStop, :) = decodedAudio(currFrameStart:currFrameStop, :) + frameT;
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end
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audiowrite(fNameOut, decodedAudio, 48000);
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x = decodedAudio;
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end
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