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

1 files changed, 98 insertions, 0 deletions

diff --git a/services/audioflinger/SourceAudioBufferProvider.cpp b/services/audioflinger/SourceAudioBufferProvider.cpp
new file mode 100644
index 00000000..e9e8c161
--- /dev/null
+++ b/services/audioflinger/SourceAudioBufferProvider.cpp
@@ -0,0 +1,98 @@
+/*
+ * 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 "SourceAudioBufferProvider"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include "SourceAudioBufferProvider.h"
+
+namespace android {
+
+SourceAudioBufferProvider::SourceAudioBufferProvider(const sp<NBAIO_Source>& source) :
+    mSource(source),
+    // mFrameBitShiftFormat below
+    mAllocated(NULL), mSize(0), mOffset(0), mRemaining(0), mGetCount(0)
+{
+    ALOG_ASSERT(source != 0);
+
+    // negotiate with source
+    NBAIO_Format counterOffers[1];
+    size_t numCounterOffers = 1;
+    ssize_t index = source->negotiate(NULL, 0, counterOffers, numCounterOffers);
+    ALOG_ASSERT(index == (ssize_t) NEGOTIATE && numCounterOffers > 0);
+    numCounterOffers = 0;
+    index = source->negotiate(counterOffers, 1, NULL, numCounterOffers);
+    ALOG_ASSERT(index == 0);
+    mFrameBitShift = Format_frameBitShift(source->format());
+}
+
+SourceAudioBufferProvider::~SourceAudioBufferProvider()
+{
+    free(mAllocated);
+}
+
+status_t SourceAudioBufferProvider::getNextBuffer(Buffer *buffer, int64_t pts)
+{
+    ALOG_ASSERT(buffer != NULL && buffer->frameCount > 0 && mGetCount == 0);
+    // any leftover data available?
+    if (mRemaining > 0) {
+        ALOG_ASSERT(mOffset + mRemaining <= mSize);
+        if (mRemaining < buffer->frameCount) {
+            buffer->frameCount = mRemaining;
+        }
+        buffer->raw = (char *) mAllocated + (mOffset << mFrameBitShift);
+        mGetCount = buffer->frameCount;
+        return OK;
+    }
+    // do we need to reallocate?
+    if (buffer->frameCount > mSize) {
+        free(mAllocated);
+        mAllocated = malloc(buffer->frameCount << mFrameBitShift);
+        mSize = buffer->frameCount;
+    }
+    // read from source
+    ssize_t actual = mSource->read(mAllocated, buffer->frameCount);
+    if (actual > 0) {
+        ALOG_ASSERT((size_t) actual <= buffer->frameCount);
+        mOffset = 0;
+        mRemaining = actual;
+        buffer->raw = mAllocated;
+        buffer->frameCount = actual;
+        mGetCount = actual;
+        return OK;
+    }
+    buffer->raw = NULL;
+    buffer->frameCount = 0;
+    mGetCount = 0;
+    return NOT_ENOUGH_DATA;
+}
+
+void SourceAudioBufferProvider::releaseBuffer(Buffer *buffer)
+{
+    ALOG_ASSERT((buffer != NULL) &&
+            (buffer->raw == (char *) mAllocated + (mOffset << mFrameBitShift)) &&
+            (buffer->frameCount <= mGetCount) &&
+            (mGetCount <= mRemaining) &&
+            (mOffset + mRemaining <= mSize));
+    mOffset += buffer->frameCount;
+    mRemaining -= buffer->frameCount;
+    buffer->raw = NULL;
+    buffer->frameCount = 0;
+    mGetCount = 0;
+}
+
+}   // namespace android