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/client/AAudioFlowGraph.cpp b/media/libaaudio/src/client/AAudioFlowGraph.cpp
new file mode 100644
index 0000000..3e43c6b
--- /dev/null
+++ b/media/libaaudio/src/client/AAudioFlowGraph.cpp
@@ -0,0 +1,116 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AAudioFlowGraph"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include "AAudioFlowGraph.h"
+
+#include <flowgraph/ClipToRange.h>
+#include <flowgraph/MonoToMultiConverter.h>
+#include <flowgraph/RampLinear.h>
+#include <flowgraph/SinkFloat.h>
+#include <flowgraph/SinkI16.h>
+#include <flowgraph/SinkI24.h>
+#include <flowgraph/SourceFloat.h>
+#include <flowgraph/SourceI16.h>
+#include <flowgraph/SourceI24.h>
+
+using namespace flowgraph;
+
+aaudio_result_t AAudioFlowGraph::configure(audio_format_t sourceFormat,
+                          int32_t sourceChannelCount,
+                          audio_format_t sinkFormat,
+                          int32_t sinkChannelCount) {
+    AudioFloatOutputPort *lastOutput = nullptr;
+
+    ALOGD("%s() source format = 0x%08x, channels = %d, sink format = 0x%08x, channels = %d",
+          __func__, sourceFormat, sourceChannelCount, sinkFormat, sinkChannelCount);
+
+    switch (sourceFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            mSource = std::make_unique<SourceFloat>(sourceChannelCount);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            mSource = std::make_unique<SourceI16>(sourceChannelCount);
+            break;
+        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+            mSource = std::make_unique<SourceI24>(sourceChannelCount);
+            break;
+        default: // TODO add I32
+            ALOGE("%s() Unsupported source format = %d", __func__, sourceFormat);
+            return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+    lastOutput = &mSource->output;
+
+    // Apply volume as a ramp to avoid pops.
+    mVolumeRamp = std::make_unique<RampLinear>(sourceChannelCount);
+    lastOutput->connect(&mVolumeRamp->input);
+    lastOutput = &mVolumeRamp->output;
+
+    // For a pure float graph, there is chance that the data range may be very large.
+    // So we should clip to a reasonable value that allows a little headroom.
+    if (sourceFormat == AUDIO_FORMAT_PCM_FLOAT && sinkFormat == AUDIO_FORMAT_PCM_FLOAT) {
+        mClipper = std::make_unique<ClipToRange>(sourceChannelCount);
+        lastOutput->connect(&mClipper->input);
+        lastOutput = &mClipper->output;
+    }
+
+    // Expand the number of channels if required.
+    if (sourceChannelCount == 1 && sinkChannelCount > 1) {
+        mChannelConverter = std::make_unique<MonoToMultiConverter>(sinkChannelCount);
+        lastOutput->connect(&mChannelConverter->input);
+        lastOutput = &mChannelConverter->output;
+    } else if (sourceChannelCount != sinkChannelCount) {
+        ALOGE("%s() Channel reduction not supported.", __func__);
+        return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+
+    switch (sinkFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            mSink = std::make_unique<SinkFloat>(sinkChannelCount);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            mSink = std::make_unique<SinkI16>(sinkChannelCount);
+            break;
+        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+            mSink = std::make_unique<SinkI24>(sinkChannelCount);
+            break;
+        default: // TODO add I32
+            ALOGE("%s() Unsupported sink format = %d", __func__, sinkFormat);
+            return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+    lastOutput->connect(&mSink->input);
+
+    return AAUDIO_OK;
+}
+
+void AAudioFlowGraph::process(const void *source, void *destination, int32_t numFrames) {
+    mSource->setData(source, numFrames);
+    mSink->read(destination, numFrames);
+}
+
+/**
+ * @param volume between 0.0 and 1.0
+ */
+void AAudioFlowGraph::setTargetVolume(float volume) {
+    mVolumeRamp->setTarget(volume);
+}
+
+void AAudioFlowGraph::setRampLengthInFrames(int32_t numFrames) {
+    mVolumeRamp->setLengthInFrames(numFrames);
+}
diff --git a/media/libaaudio/src/client/AAudioFlowGraph.h b/media/libaaudio/src/client/AAudioFlowGraph.h
new file mode 100644
index 0000000..a49f64e
--- /dev/null
+++ b/media/libaaudio/src/client/AAudioFlowGraph.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AAUDIO_FLOW_GRAPH_H
+#define ANDROID_AAUDIO_FLOW_GRAPH_H
+
+#include <memory>
+#include <stdint.h>
+#include <sys/types.h>
+#include <system/audio.h>
+
+#include <aaudio/AAudio.h>
+#include <flowgraph/ClipToRange.h>
+#include <flowgraph/MonoToMultiConverter.h>
+#include <flowgraph/RampLinear.h>
+
+class AAudioFlowGraph {
+public:
+    /** Connect several modules together to convert from source to sink.
+     * This should only be called once for each instance.
+     *
+     * @param sourceFormat
+     * @param sourceChannelCount
+     * @param sinkFormat
+     * @param sinkChannelCount
+     * @return
+     */
+    aaudio_result_t configure(audio_format_t sourceFormat,
+                              int32_t sourceChannelCount,
+                              audio_format_t sinkFormat,
+                              int32_t sinkChannelCount);
+
+    void process(const void *source, void *destination, int32_t numFrames);
+
+    /**
+     * @param volume between 0.0 and 1.0
+     */
+    void setTargetVolume(float volume);
+
+    void setRampLengthInFrames(int32_t numFrames);
+
+private:
+    std::unique_ptr<flowgraph::AudioSource>          mSource;
+    std::unique_ptr<flowgraph::RampLinear>           mVolumeRamp;
+    std::unique_ptr<flowgraph::ClipToRange>          mClipper;
+    std::unique_ptr<flowgraph::MonoToMultiConverter> mChannelConverter;
+    std::unique_ptr<flowgraph::AudioSink>            mSink;
+};
+
+
+#endif //ANDROID_AAUDIO_FLOW_GRAPH_H
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 9204824..0a8021a 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -39,7 +39,6 @@
 #include "core/AudioStreamBuilder.h"
 #include "fifo/FifoBuffer.h"
 #include "utility/AudioClock.h"
-#include "utility/LinearRamp.h"
 
 #include "AudioStreamInternal.h"
 
@@ -92,11 +91,11 @@
     }
 
     // We have to do volume scaling. So we prefer FLOAT format.
-    if (getFormat() == AAUDIO_FORMAT_UNSPECIFIED) {
-        setFormat(AAUDIO_FORMAT_PCM_FLOAT);
+    if (getFormat() == AUDIO_FORMAT_DEFAULT) {
+        setFormat(AUDIO_FORMAT_PCM_FLOAT);
     }
     // Request FLOAT for the shared mixer.
-    request.getConfiguration().setFormat(AAUDIO_FORMAT_PCM_FLOAT);
+    request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT);
 
     // Build the request to send to the server.
     request.setUserId(getuid());
@@ -126,7 +125,7 @@
         // if that failed then try switching from mono to stereo if OUTPUT.
         // Only do this in the client. Otherwise we end up with a mono mixer in the service
         // that writes to a stereo MMAP stream.
-        ALOGD("%s - openStream() returned %d, try switching from MONO to STEREO",
+        ALOGD("%s() - openStream() returned %d, try switching from MONO to STEREO",
               __func__, mServiceStreamHandle);
         request.getConfiguration().setSamplesPerFrame(2); // stereo
         mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
@@ -212,9 +211,7 @@
             mCallbackFrames = mFramesPerBurst;
         }
 
-        int32_t bytesPerFrame = getSamplesPerFrame()
-                                * AAudioConvert_formatToSizeInBytes(getFormat());
-        int32_t callbackBufferSize = mCallbackFrames * bytesPerFrame;
+        const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame();
         mCallbackBuffer = new uint8_t[callbackBufferSize];
     }
 
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 0425cd5..3bb9e1e 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -27,7 +27,6 @@
 #include "client/AudioEndpoint.h"
 #include "core/AudioStream.h"
 #include "utility/AudioClock.h"
-#include "utility/LinearRamp.h"
 
 using android::sp;
 using android::IAAudioService;
@@ -193,6 +192,8 @@
 
     int64_t                  mServiceLatencyNanos = 0;
 
+    // Sometimes the hardware is operating with a different channel count from the app.
+    // Then we require conversion in AAudio.
     int32_t                  mDeviceChannelCount = 0;
 };
 
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;
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 795ba2c..2ae37a5 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -43,9 +43,17 @@
 aaudio_result_t AudioStreamInternalPlay::open(const AudioStreamBuilder &builder) {
     aaudio_result_t result = AudioStreamInternal::open(builder);
     if (result == AAUDIO_OK) {
+        result = mFlowGraph.configure(getFormat(),
+                             getSamplesPerFrame(),
+                             getDeviceFormat(),
+                             getDeviceChannelCount());
+
+        if (result != AAUDIO_OK) {
+            close();
+        }
         // Sample rate is constrained to common values by now and should not overflow.
         int32_t numFrames = kRampMSec * getSampleRate() / AAUDIO_MILLIS_PER_SECOND;
-        mVolumeRamp.setLengthInFrames(numFrames);
+        mFlowGraph.setRampLengthInFrames(numFrames);
     }
     return result;
 }
@@ -216,22 +224,10 @@
             }
 
             int32_t numBytes = getBytesPerFrame() * framesToWrite;
-            // Data conversion.
-            float levelFrom;
-            float levelTo;
-            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);
+            mFlowGraph.process((void *)byteBuffer,
+                               wrappingBuffer.data[partIndex],
+                               framesToWrite);
 
             byteBuffer += numBytes;
             framesLeft -= framesToWrite;
@@ -313,6 +309,6 @@
     float combinedVolume = mStreamVolume * getDuckAndMuteVolume();
     ALOGD("%s() mStreamVolume * duckAndMuteVolume = %f * %f = %f",
           __func__, mStreamVolume, getDuckAndMuteVolume(), combinedVolume);
-    mVolumeRamp.setTarget(combinedVolume);
+    mFlowGraph.setTargetVolume(combinedVolume);
     return android::NO_ERROR;
 }
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.h b/media/libaaudio/src/client/AudioStreamInternalPlay.h
index 977a909..cab2942 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.h
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.h
@@ -21,6 +21,7 @@
 #include <aaudio/AAudio.h>
 
 #include "binding/AAudioServiceInterface.h"
+#include "client/AAudioFlowGraph.h"
 #include "client/AudioStreamInternal.h"
 
 using android::sp;
@@ -93,7 +94,7 @@
 
     int64_t                  mLastFramesRead = 0; // used to prevent retrograde motion
 
-    LinearRamp               mVolumeRamp;
+    AAudioFlowGraph          mFlowGraph;
 
 };