Non-blocking audio I/O interface, WIP

Yet another abstraction similar to AudioTrack::Buffer and AudioBufferProvider,
but with support for streaming, non-blocking, and eventually PTS.

This is intended to be used as follows:
 - primary HAL output stream implements a Sink
 - primary HAL input stream implements a Source
 - Pipe implements a Sink
 - PipeReader implements a Source or TimedSource (not shown yet),
   which supports "read at PTS"
 - fast AudioTrack on server side will implement a Source using cblk
 - normal AudioTrack on server side will not be changed initially
 - fast AudioRecord on server side will implement a Sink using cblk
 - normal AudioRecord on server side will not be changed initially
 - fast mixer thread will read from Sources and write to a Sink,
   or (unlikely) implement a Source and multiple Sinks
 - Visualization and PCM logger will read from Source or TimedSource
 - A2DP normal mixer will be connected directly to its output stream
   and there will be a kind of OutputTrack for duplication that will
   read from a Sink with non-blocking write fed by the fast mixer.

Patch set 3 changes:
 - Add more implementations of NBAIO interfaces:
   added SourceAudioBufferProvider, MonoPipe, MonoPipeReader.
 - Added Format_sampleRate and Format_channelCount.
 - Extract out the roundUp() method.
 - Respond to most comments from previous code review.
 - The new classes are untested.

Patch set 4 changes:
 - Fix bugs in MonoPipe::write() and MonoPipeReader::read()
 - Fix bug initializing mFrameBitShift too early
 - renamed roundUp() to roundup()
 - Fix Android.mk
 - Add LOG_TAG an LOG_NDEBUG, use ALOG_ASSERT and utils/Log.h instead of assert
 - Fix build warnings
 - Move constructor and destructor bodies from .h to .cpp
 - Line length 100
 - Following naming conventions for #include double-include protector macros
 - Include what you use
 - More NBAIO logging
 - MonoPipe write can be blocking

Patch set 5 changes:
 - Address code review comments
 - Use a static library so unused implementations don't take memory
 - Comment out libsndfile dependency
 - Remove debugging LOGV and LOG_NDEBUG

Patch set 6 changes (would be 6 at old location, actually 2 at new location):
 - Address code review comments on patchset 5
 - For MonoPipe, allow the full pipe to be used, no need to omit one slot
 - Don't do atomic releasing stores unless needed

Still to do:
 - I'm not happy with the Pipe class names
 - Update build/ for new static library?

Change-Id: Ie6c61f05ce06b676b033be448a8ef9025a2ffcfd
diff --git a/services/audioflinger/AudioBufferProviderSource.cpp b/services/audioflinger/AudioBufferProviderSource.cpp
new file mode 100644
index 0000000..4342171
--- /dev/null
+++ b/services/audioflinger/AudioBufferProviderSource.cpp
@@ -0,0 +1,141 @@
+/*
+ * Copyright (C) 2012 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 "AudioBufferProviderSource"
+//#define LOG_NDEBUG 0
+
+#include <cutils/compiler.h>
+#include <utils/Log.h>
+#include "AudioBufferProviderSource.h"
+
+namespace android {
+
+AudioBufferProviderSource::AudioBufferProviderSource(AudioBufferProvider *provider,
+                                                     NBAIO_Format format) :
+    NBAIO_Source(format), mProvider(provider), mConsumed(0)
+{
+    ALOG_ASSERT(provider != NULL);
+    ALOG_ASSERT(format != Format_Invalid);
+}
+
+AudioBufferProviderSource::~AudioBufferProviderSource()
+{
+    if (mBuffer.raw != NULL) {
+        mProvider->releaseBuffer(&mBuffer);
+    }
+}
+
+ssize_t AudioBufferProviderSource::availableToRead()
+{
+    if (CC_UNLIKELY(!mNegotiated)) {
+        return NEGOTIATE;
+    }
+    return mBuffer.raw != NULL ? mBuffer.frameCount - mConsumed : 0;
+}
+
+ssize_t AudioBufferProviderSource::read(void *buffer, size_t count)
+{
+    if (CC_UNLIKELY(!mNegotiated)) {
+        return NEGOTIATE;
+    }
+    if (CC_UNLIKELY(mBuffer.raw == NULL)) {
+        mBuffer.frameCount = count;
+        status_t status = mProvider->getNextBuffer(&mBuffer, AudioBufferProvider::kInvalidPTS);
+        if (status != OK) {
+            return status == NOT_ENOUGH_DATA ? (ssize_t) WOULD_BLOCK : (ssize_t) status;
+        }
+        ALOG_ASSERT(mBuffer.raw != NULL);
+        // mConsumed is 0 either from constructor or after releaseBuffer()
+    }
+    size_t available = mBuffer.frameCount - mConsumed;
+    if (CC_UNLIKELY(count > available)) {
+        count = available;
+    }
+    // count could be zero, either because count was zero on entry or
+    // available is zero, but both are unlikely so don't check for that
+    memcpy(buffer, (char *) mBuffer.raw + (mConsumed << mBitShift), count << mBitShift);
+    if (CC_UNLIKELY((mConsumed += count) >= mBuffer.frameCount)) {
+        mProvider->releaseBuffer(&mBuffer);
+        mBuffer.raw = NULL;
+        mConsumed = 0;
+    }
+    mFramesRead += count;
+    // For better responsiveness with large values of count,
+    // return a short count rather than continuing with next buffer.
+    // This gives the caller a chance to interpolate other actions.
+    return count;
+}
+
+ssize_t AudioBufferProviderSource::readVia(readVia_t via, size_t total, void *user, size_t block)
+{
+    if (CC_UNLIKELY(!mNegotiated)) {
+        return NEGOTIATE;
+    }
+    if (CC_UNLIKELY(block == 0)) {
+        block = ~0;
+    }
+    for (size_t accumulator = 0; ; ) {
+        ALOG_ASSERT(accumulator <= total);
+        size_t count = total - accumulator;
+        if (CC_UNLIKELY(count == 0)) {
+            return accumulator;
+        }
+        if (CC_LIKELY(count > block)) {
+            count = block;
+        }
+        // 1 <= count <= block
+        if (CC_UNLIKELY(mBuffer.raw == NULL)) {
+            mBuffer.frameCount = count;
+            status_t status = mProvider->getNextBuffer(&mBuffer, AudioBufferProvider::kInvalidPTS);
+            if (CC_LIKELY(status == OK)) {
+                ALOG_ASSERT(mBuffer.raw != NULL && mBuffer.frameCount <= count);
+                // mConsumed is 0 either from constructor or after releaseBuffer()
+                continue;
+            }
+            // FIXME simplify logic - does the initial count and block checks again for no reason;
+            //       don't you just want to fall through to the size_t available line?
+            if (CC_LIKELY(status == NOT_ENOUGH_DATA)) {
+                status = WOULD_BLOCK;
+            }
+            return accumulator > 0 ? accumulator : (ssize_t) status;
+        }
+        size_t available = mBuffer.frameCount - mConsumed;
+        if (CC_UNLIKELY(count > available)) {
+            count = available;
+        }
+        if (CC_LIKELY(count > 0)) {
+            ssize_t ret = via(user, (char *) mBuffer.raw + (mConsumed << mBitShift), count);
+            if (CC_UNLIKELY(ret <= 0)) {
+                if (CC_LIKELY(accumulator > 0)) {
+                    return accumulator;
+                }
+                return ret;
+            }
+            ALOG_ASSERT((size_t) ret <= count);
+            mFramesRead += ret;
+            accumulator += ret;
+            if (CC_LIKELY((mConsumed += ret) < mBuffer.frameCount)) {
+                continue;
+            }
+        }
+        mProvider->releaseBuffer(&mBuffer);
+        mBuffer.raw = NULL;
+        mConsumed = 0;
+        // don't get next buffer until we really need it
+    }
+}
+
+}   // namespace android