Merge pull request #225 from DrChat/audio_fix

Fix audio decoder - mono sounds are working now!

src/xenia/apu/audio_decoder.cc

@@ -87,7 +87,7 @@ int AudioDecoder::Initialize(int bits) {
 
   // Current frame stuff whatever
   // samples per frame * 2 max channels * output bytes
-  current_frame_ = new uint8_t[XMAContextData::kSamplesPerFrame * 2 * (bits / 8)];
+  current_frame_ = new uint8_t[XMAContextData::kSamplesPerFrame * 2 * (bits/8)];
   current_frame_pos_ = 0;
   frame_samples_size_ = 0;
 
@@ -179,6 +179,7 @@ int AudioDecoder::DecodePacket(uint8_t* output, size_t output_offset, size_t out
 
     // Successfully decoded a frame
     if (got_frame) {
+      // Validity checks.
       if (decoded_frame_->nb_samples > XMAContextData::kSamplesPerFrame) {
         return -2;
       } else if (context_->sample_fmt != AV_SAMPLE_FMT_FLTP) {
@@ -193,24 +194,26 @@ int AudioDecoder::DecodePacket(uint8_t* output, size_t output_offset, size_t out
         return -4;
       }
 
-      float* curSample = (float *)decoded_frame_->data[0];
+      // Output sample array
+      float* sample_array = (float *)decoded_frame_->data[0];
 
       // Loop through every sample, convert and drop it into the output array
       for (int i = 0; i < decoded_frame_->nb_samples; i++) {
-        float fSample = curSample[i] * (1 << (bits_ - 1));
+        // Raw sample should be within [-1, 1]
-        // Weird problem: Sometimes the samples are outside [-1,1]
+        float fRawSample = sample_array[i];
-        if (fSample >= 0) {
+        float fScaledSample = fRawSample * (1 << (bits_ - 1));
-          fSample += 0.5f;
+
-          if (fSample > (1 << (bits_ - 1)) - 1) {
+        // Clamp the sample in range
-            fSample = (float)(1 << (bits_ - 1)) - 1;
+        int64_t range = (1 << (bits_ - 1));
-          }
+        if (fScaledSample > (range - 1)) {
-        } else {
+          fScaledSample = (float)range;
-          fSample -= 0.5f;
+        } else if (fScaledSample < (-range + 1)) {
+          fScaledSample = (float)-range;
         }
 
-        // Convert the sample
+        // Convert the sample and output it in big endian
-        int sample = (int)fSample;
+        int sample = (int)fScaledSample;
-        for (uint32_t j = 0; j < sample_size; j++) {
+        for (int32_t j = sample_size-1; j >= 0; j--) {
           current_frame_[i * sample_size + j] = sample & 0xFF;
           sample >>= 8;
         }

src/xenia/apu/audio_system.cc

@@ -246,8 +246,8 @@ void AudioSystem::DecoderThreadMain() {
                               data.input_buffer_1_block_count) * 2048;
           size_t input_offset = (data.input_buffer_read_offset / 8 - 4);
           size_t input_remaining = input_size - input_offset;
-          if (input_remaining == 0) {
+          if (input_offset > input_size) {
-            // We're finished!
+            // We're finished. Break.
             break;
           }
 
@@ -267,27 +267,24 @@ void AudioSystem::DecoderThreadMain() {
           int read = context.decoder->DecodePacket(out, output_offset,
                                                    output_remaining);
           if (read < 0) {
+            // Sometimes the decoder will fail on a packet. I think it's
+            // looking for cross-packet frames and failing. If you run it again
+            // on the same packet it'll work though.
             XELOGAPU("APU failed to decode packet (returned %.8X)", -read);
-            context.decoder->DiscardPacket();
 
-            // TODO: Set error state
+            continue;
-
-            break;
           }
 
           if (read == 0) {
             // Select sample rate.
             int sample_rate = 0;
             if (data.sample_rate == 0) {
-              // TODO: Test this
               sample_rate = 24000;
             } else if (data.sample_rate == 1) {
               sample_rate = 32000;
             } else if (data.sample_rate == 2) {
-              // TODO: Test this
               sample_rate = 44100;
             } else if (data.sample_rate == 3) {
-              // TODO: Test this
               sample_rate = 48000;
             }
 
@@ -300,13 +297,14 @@ void AudioSystem::DecoderThreadMain() {
             // New packet time.
             // TODO: Select input buffer 1 if necessary.
             auto packet = in0 + input_offset;
-            context.decoder->PreparePacket(packet, 2048, sample_rate, channels);
+            context.decoder->PreparePacket(packet, 2048, sample_rate,
+                                           channels);
             input_offset += 2048;
           }
 
           output_offset += read;
 
-          // blah copy these back to the context
+          // Copy the variables we changed back to the context.
           data.input_buffer_read_offset = (input_offset + 4) * 8;
           data.output_buffer_write_offset = output_offset / 256;
         }