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