audio_core: hle: mf: fix a regression...

... where the sample copying logic is incorrect due to the wrong usage
of `std::array`

src/audio_core/hle/adts.h

@@ -16,8 +16,8 @@ struct ADTSData {
     u32 samplerate;
 };
 
-ADTSData ParseADTS(char* buffer);
+ADTSData ParseADTS(const char* buffer);
 
 // last two bytes of MF AAC decoder user data
 // see https://docs.microsoft.com/en-us/windows/desktop/medfound/aac-decoder#example-media-types
-u16 MFGetAACTag(ADTSData input);
+u16 MFGetAACTag(const ADTSData input);

src/audio_core/hle/adts_reader.cpp

@@ -8,7 +8,7 @@ constexpr std::array<u32, 16> freq_table = {96000, 88200, 64000, 48000, 44100, 3
                                             16000, 12000, 11025, 8000,  7350,  0,     0,     0};
 constexpr std::array<u8, 8> channel_table = {0, 1, 2, 3, 4, 5, 6, 8};
 
-ADTSData ParseADTS(char* buffer) {
+ADTSData ParseADTS(const char* buffer) {
     u32 tmp = 0;
     ADTSData out;
 
@@ -50,7 +50,7 @@ ADTSData ParseADTS(char* buffer) {
 // Frame length flag (1 bit)
 // Depends on core coder (1 bit)
 // Extension flag (1 bit)
-u16 MFGetAACTag(ADTSData input) {
+u16 MFGetAACTag(const ADTSData input) {
     u16 tag = 0;
 
     tag |= input.profile << 11;

src/audio_core/hle/wmf_decoder.cpp

@@ -116,12 +116,14 @@ MFOutputState WMFDecoder::Impl::DecodingLoop(ADTSData adts_header,
 
             // the following was taken from ffmpeg version of the decoder
             f32 val_f32;
-            for (size_t i = 0; i < output_buffer->size(); i++) {
+            for (size_t i = 0; i < output_buffer->size(); ) {
                 for (std::size_t channel = 0; channel < adts_header.channels; channel++) {
                     val_f32 = output_buffer->at(i);
                     s16 val = static_cast<s16>(0x7FFF * val_f32);
                     out_streams[channel].push_back(val & 0xFF);
                     out_streams[channel].push_back(val >> 8);
+                    // i is incremented on per channel basis
+                    i++;
                 }
             }
         }