aaudio: use new flowgraph to simplify processing

Construct a flowgraph based on the source and destination
format and channelCount. This is groundwork for supporting 24-bit
PCM formats.

Also cleaned up handling of device related format.

This CL removes more code than it adds.

Bug: 65067568
Test: write_sine_callback.cpp -pl
Test: write_sine_callback.cpp -pl -x
Test: input_monitor -pl
Test: input_monitor -pl -x
Change-Id: Ia155bff0164912011d09b61b54f983ccf4490bd1
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 505cd77..1572f0d 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -88,9 +88,9 @@
 
     int32_t notificationFrames = 0;
 
-    audio_format_t format = (getFormat() == AAUDIO_FORMAT_UNSPECIFIED)
+    const audio_format_t format = (getFormat() == AUDIO_FORMAT_DEFAULT)
             ? AUDIO_FORMAT_PCM_FLOAT
-            : AAudioConvert_aaudioToAndroidDataFormat(getFormat());
+            : getFormat();
 
     // Setup the callback if there is one.
     AudioTrack::callback_t callback = nullptr;
@@ -178,10 +178,8 @@
 
     // Get the actual values from the AudioTrack.
     setSamplesPerFrame(mAudioTrack->channelCount());
-    aaudio_format_t aaudioFormat =
-            AAudioConvert_androidToAAudioDataFormat(mAudioTrack->format());
-    setFormat(aaudioFormat);
-    setDeviceFormat(aaudioFormat);
+    setFormat(mAudioTrack->format());
+    setDeviceFormat(mAudioTrack->format());
 
     int32_t actualSampleRate = mAudioTrack->getSampleRate();
     ALOGW_IF(actualSampleRate != getSampleRate(),