aaudio: convert mono output to stereo

MMAP hardware streams are often stereo.
But apps often want to output a mono stream.
Since converting from mono to stereo is easy,
we can go ahead and open a stereo hardware stream
and then just convert the apps mono data to stereo for the HW.

Add getDeviceChannelCount().

Test:  adb shell write_sine_callback -pl -s10 -c1 -m3
Test:  adb shell write_sine_callback -pl -s10 -c1 -m3 -x
Change-Id: I444a38c6f5cd32d1d6113f16aacec68285a1bc82
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 5660c1b..7695dfa 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -194,7 +194,7 @@
     // ALOGD("AudioStreamInternal::writeNowWithConversion(%p, %d)",
     //              buffer, numFrames);
     WrappingBuffer wrappingBuffer;
-    uint8_t *source = (uint8_t *) buffer;
+    uint8_t *byteBuffer = (uint8_t *) buffer;
     int32_t framesLeft = numFrames;
 
     mAudioEndpoint.getEmptyFramesAvailable(&wrappingBuffer);
@@ -208,69 +208,26 @@
             if (framesToWrite > framesAvailable) {
                 framesToWrite = framesAvailable;
             }
+
             int32_t numBytes = getBytesPerFrame() * framesToWrite;
-            int32_t numSamples = framesToWrite * getSamplesPerFrame();
             // Data conversion.
             float levelFrom;
             float levelTo;
-            bool ramping = mVolumeRamp.nextSegment(framesToWrite, &levelFrom, &levelTo);
-            // The formats are validated when the stream is opened so we do not have to
-            // check for illegal combinations here.
-            // TODO factor this out into a utility function
-            if (getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
-                if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
-                    AAudio_linearRamp(
-                            (const float *) source,
-                            (float *) wrappingBuffer.data[partIndex],
-                            framesToWrite,
-                            getSamplesPerFrame(),
-                            levelFrom,
-                            levelTo);
-                } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
-                    if (ramping) {
-                        AAudioConvert_floatToPcm16(
-                                (const float *) source,
-                                (int16_t *) wrappingBuffer.data[partIndex],
-                                framesToWrite,
-                                getSamplesPerFrame(),
-                                levelFrom,
-                                levelTo);
-                    } else {
-                        AAudioConvert_floatToPcm16(
-                                (const float *) source,
-                                (int16_t *) wrappingBuffer.data[partIndex],
-                                numSamples,
-                                levelTo);
-                    }
-                }
-            } else if (getFormat() == AAUDIO_FORMAT_PCM_I16) {
-                if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
-                    if (ramping) {
-                        AAudioConvert_pcm16ToFloat(
-                                (const int16_t *) source,
-                                (float *) wrappingBuffer.data[partIndex],
-                                framesToWrite,
-                                getSamplesPerFrame(),
-                                levelFrom,
-                                levelTo);
-                    } else {
-                        AAudioConvert_pcm16ToFloat(
-                                (const int16_t *) source,
-                                (float *) wrappingBuffer.data[partIndex],
-                                numSamples,
-                                levelTo);
-                    }
-                } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
-                    AAudio_linearRamp(
-                            (const int16_t *) source,
-                            (int16_t *) wrappingBuffer.data[partIndex],
-                            framesToWrite,
-                            getSamplesPerFrame(),
-                            levelFrom,
-                            levelTo);
-                }
-            }
-            source += numBytes;
+            mVolumeRamp.nextSegment(framesToWrite, &levelFrom, &levelTo);
+
+            AAudioDataConverter::FormattedData source(
+                    (void *)byteBuffer,
+                    getFormat(),
+                    getSamplesPerFrame());
+            AAudioDataConverter::FormattedData destination(
+                    wrappingBuffer.data[partIndex],
+                    getDeviceFormat(),
+                    getDeviceChannelCount());
+
+            AAudioDataConverter::convert(source, destination, framesToWrite,
+                                         levelFrom, levelTo);
+
+            byteBuffer += numBytes;
             framesLeft -= framesToWrite;
         } else {
             break;