Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_state_audio_path_unittest.cc

Index: webrtc/audio/audio_state_audio_path_unittest.cc
diff --git a/webrtc/audio/audio_state_audio_path_unittest.cc b/webrtc/audio/audio_state_audio_path_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..2cb643b8bce8cacab06deb55a994fa789e3f0403
--- /dev/null
+++ b/webrtc/audio/audio_state_audio_path_unittest.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2016 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 <memory>
+
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc/test/gtest.h"
+#include "webrtc/test/mock_voice_engine.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+struct ConfigHelper {
+  ConfigHelper() : audio_mixer_(AudioMixerImpl::Create()) {
+    using testing::_;
+
+    EXPECT_CALL(voice_engine_, RegisterVoiceEngineObserver(testing::_))
+        .WillOnce(testing::Return(0));
+    EXPECT_CALL(voice_engine_, DeRegisterVoiceEngineObserver())
+        .WillOnce(testing::Return(0));
+    EXPECT_CALL(voice_engine_, audio_processing());
+    EXPECT_CALL(voice_engine_, audio_transport());
+
+    ON_CALL(voice_engine_, audio_transport())
+        .WillByDefault(testing::Return(&audio_transport_));
+
+    config_.voice_engine = &voice_engine_;
+    config_.audio_mixer = audio_mixer_;
+  }
+
+  rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer_; }
+
+  MockAudioTransport& original_audio_transport() { return audio_transport_; }
+
+  AudioState::Config& config() { return config_; }
+  MockVoiceEngine& voice_engine() { return voice_engine_; }
+
+  testing::StrictMock<MockVoiceEngine> voice_engine_;
+  MockAudioTransport audio_transport_;
+  rtc::scoped_refptr<AudioMixer> audio_mixer_;
+  AudioState::Config config_;
+};
+
+class FakeAudioSource : public AudioMixer::Source {
+ public:
+  int Ssrc() const /*override*/ { return 0; }
+
+  int PreferredSampleRate() const /*override*/ { return 8000; }
+
+  MOCK_METHOD2(GetAudioFrameWithInfo,
+               AudioFrameInfo(int sample_rate_hz, AudioFrame* audio_frame));
+};
+}  // namespace
+
+// Test that RecordedDataIsAvailable calls get to the original transport.
+TEST(AudioStateAudioPathTest, RecordedAudioArrivesAtOriginalTransport) {
+  ConfigHelper helper;
+
+  EXPECT_CALL(helper.voice_engine(), audio_device_module()).Times(2);
+  auto device = static_cast<MockAudioDeviceModule*>(
+      helper.voice_engine().audio_device_module());
+
+  // Get a local pointer to the most recent transport connected to the
+  // Audio Device.
+  AudioTransport* audio_transport_proxy = nullptr;
+  ON_CALL(*device, RegisterAudioCallback(testing::_))
+      .WillByDefault(
+          testing::Invoke([&audio_transport_proxy](AudioTransport* transport) {
+            audio_transport_proxy = transport;
+            return 0;
+          }));
+
+  rtc::scoped_refptr<AudioState> audio_state =
+      AudioState::Create(helper.config());
+
+  // Setup completed. Ensure call of original transport is forwarded to new.
+  uint32_t new_mic_level;
+  EXPECT_CALL(helper.original_audio_transport(),
+              RecordedDataIsAvailable(nullptr, 80, 2, 1, 8000, 0, 0, 0, false,
+                                      testing::Ref(new_mic_level)));
+
+  audio_transport_proxy->RecordedDataIsAvailable(nullptr, 80, 2, 1, 8000, 0, 0,
+                                                 0, false, new_mic_level);
+}
+
+TEST(AudioStateAudioPathTest,
+     QueryingProxyForAudioShouldResultInGetAudioCallOnMixerSource) {
+  ConfigHelper helper;
+
+  EXPECT_CALL(helper.voice_engine(), audio_device_module()).Times(2);
+  auto device = static_cast<MockAudioDeviceModule*>(
+      helper.voice_engine().audio_device_module());
+
+  AudioTransport* audio_transport_proxy = nullptr;
+  ON_CALL(*device, RegisterAudioCallback(testing::_))
+      .WillByDefault(
+          testing::Invoke([&audio_transport_proxy](AudioTransport* transport) {
+            audio_transport_proxy = transport;
+            return 0;
+          }));
+
+  rtc::scoped_refptr<AudioState> audio_state =
+      AudioState::Create(helper.config());
+
+  FakeAudioSource fake_source;
+
+  helper.mixer()->AddSource(&fake_source);
+
+  EXPECT_CALL(fake_source, GetAudioFrameWithInfo(testing::_, testing::_))
+      .WillOnce(
+          testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
+            audio_frame->sample_rate_hz_ = sample_rate_hz;
+            audio_frame->samples_per_channel_ = sample_rate_hz / 100;
+            audio_frame->num_channels_ = 1;
+            return AudioMixer::Source::AudioFrameInfo::kNormal;
+          }));
+
+  int16_t audio_buffer[80];
+  size_t n_samples_out;
+  int64_t elapsed_time_ms;
+  int64_t ntp_time_ms;
+  audio_transport_proxy->NeedMorePlayData(80, 2, 1, 8000, audio_buffer,
+                                          n_samples_out, &elapsed_time_ms,
+                                          &ntp_time_ms);
+}
+}  // namespace test
+}  // namespace webrtc