commit 0127c1be07d63e046e136126cd465c3fa8977768
author Phil Burk <philburk@google.com> Thu Mar 29 13:48:06 2018 -0700
committer Phil Burk <philburk@google.com> Fri Sep 07 11:08:24 2018 -0700
tree 4eb611ec5d010becd3583f93c640cc9468acb3b3
parent fdfae2898f0785f19de198ed00488e03bf52bca9

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

media/libaaudio/src/client/AudioStreamInternalCapture.cpp

diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 0719fe1..4a0e6da 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -20,6 +20,7 @@
 #include <utils/Log.h>
 
 #include <algorithm>
+#include <audio_utils/primitives.h>
 #include <aaudio/AAudio.h>
 
 #include "client/AudioStreamInternalCapture.h"
@@ -165,35 +166,36 @@
     // Read data in one or two parts.
     for (int partIndex = 0; framesLeft > 0 && partIndex < WrappingBuffer::SIZE; partIndex++) {
         int32_t framesToProcess = framesLeft;
-        int32_t framesAvailable = wrappingBuffer.numFrames[partIndex];
+        const int32_t framesAvailable = wrappingBuffer.numFrames[partIndex];
         if (framesAvailable <= 0) break;
 
         if (framesToProcess > framesAvailable) {
             framesToProcess = framesAvailable;
         }
 
-        int32_t numBytes = getBytesPerFrame() * framesToProcess;
-        int32_t numSamples = framesToProcess * getSamplesPerFrame();
+        const int32_t numBytes = getBytesPerFrame() * framesToProcess;
+        const int32_t numSamples = framesToProcess * getSamplesPerFrame();
 
+        const audio_format_t sourceFormat = getDeviceFormat();
+        const audio_format_t destinationFormat = getFormat();
         // TODO factor this out into a utility function
-        if (getDeviceFormat() == getFormat()) {
+        if (sourceFormat == destinationFormat) {
             memcpy(destination, wrappingBuffer.data[partIndex], numBytes);
-        } else if (getDeviceFormat() == AAUDIO_FORMAT_PCM_I16
-                   && getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
-            AAudioConvert_pcm16ToFloat(
-                    (const int16_t *) wrappingBuffer.data[partIndex],
+        } else if (sourceFormat == AUDIO_FORMAT_PCM_16_BIT
+                   && destinationFormat == AUDIO_FORMAT_PCM_FLOAT) {
+            memcpy_to_float_from_i16(
                     (float *) destination,
-                    numSamples,
-                    1.0f);
-        } else if (getDeviceFormat() == AAUDIO_FORMAT_PCM_FLOAT
-                   && getFormat() == AAUDIO_FORMAT_PCM_I16) {
-            AAudioConvert_floatToPcm16(
-                    (const float *) wrappingBuffer.data[partIndex],
+                    (const int16_t *) wrappingBuffer.data[partIndex],
+                    numSamples);
+        } else if (sourceFormat == AUDIO_FORMAT_PCM_FLOAT
+                   && destinationFormat == AUDIO_FORMAT_PCM_16_BIT) {
+            memcpy_to_i16_from_float(
                     (int16_t *) destination,
-                    numSamples,
-                    1.0f);
+                    (const float *) wrappingBuffer.data[partIndex],
+                    numSamples);
         } else {
-            ALOGE("Format conversion not supported!");
+            ALOGE("%s() - Format conversion not supported! audio_format_t source = %u, dest = %u",
+                __func__, sourceFormat, destinationFormat);
             return AAUDIO_ERROR_INVALID_FORMAT;
         }
         destination += numBytes;