Remove the requirement to call set_sample_rate_hz and friends.

Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)

Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.

TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.

R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_processing/audio_processing_impl_unittest.cc

@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/modules/interface/module_common_types.h"
+
+using ::testing::Invoke;
+using ::testing::Return;
+
+namespace webrtc {
+
+class MockInitialize : public AudioProcessingImpl {
+ public:
+  MOCK_METHOD0(InitializeLocked, int());
+
+  int RealInitializeLocked() { return AudioProcessingImpl::InitializeLocked(); }
+};
+
+TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
+  MockInitialize mock;
+  ON_CALL(mock, InitializeLocked())
+      .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
+
+  EXPECT_CALL(mock, InitializeLocked()).Times(1);
+  mock.Initialize();
+
+  AudioFrame frame;
+  // Call with the default parameters; there should be no init.
+  frame.num_channels_ = 1;
+  SetFrameSampleRate(&frame, 16000);
+  EXPECT_CALL(mock, InitializeLocked())
+      .Times(0);
+  EXPECT_EQ(kNoErr, mock.ProcessStream(&frame));
+  EXPECT_EQ(kNoErr, mock.AnalyzeReverseStream(&frame));
+
+  // New sample rate. (Only impacts ProcessStream).
+  SetFrameSampleRate(&frame, 32000);
+  EXPECT_CALL(mock, InitializeLocked())
+      .Times(1);
+  EXPECT_EQ(kNoErr, mock.ProcessStream(&frame));
+
+  // New number of channels.
+  frame.num_channels_ = 2;
+  EXPECT_CALL(mock, InitializeLocked())
+      .Times(2);
+  EXPECT_EQ(kNoErr, mock.ProcessStream(&frame));
+  // ProcessStream sets num_channels_ == num_output_channels.
+  frame.num_channels_ = 2;
+  EXPECT_EQ(kNoErr, mock.AnalyzeReverseStream(&frame));
+
+  // A new sample rate passed to AnalyzeReverseStream should be an error and
+  // not cause an init.
+  SetFrameSampleRate(&frame, 16000);
+  EXPECT_CALL(mock, InitializeLocked())
+      .Times(0);
+  EXPECT_EQ(mock.kBadSampleRateError, mock.AnalyzeReverseStream(&frame));
+}
+
+}  // namespace webrtc