Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_state_audio_path_unittest.cc

+/*
+ *  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/base/refcountedobject.h"

[the sun, 2016/11/14 20:03:46]

why?

[aleloi, 2016/11/15 16:56:54]

Not needed; removed!

+#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 {
+
+class AudioStateAudioPathTest : public testing::Test {

[the sun, 2016/11/14 20:03:46]

Please avoid fixtures. Like in the stream tests, make a little helper class that
you put on the stack in the test cases instead. This usually makes the code
easier to follow (everything you need to know is right in front of you - no
magic initialization happening before the test function runs, etc).

+ public:
+  AudioStateAudioPathTest() : 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_device_module());
+    EXPECT_CALL(voice_engine_, audio_processing());
+    EXPECT_CALL(voice_engine_, audio_transport());
+
+    auto device = static_cast<MockAudioDeviceModule*>(
+        voice_engine_.audio_device_module());
+
+    ON_CALL(*device, RegisterAudioCallback(_))
+        .WillByDefault(testing::Invoke([this](AudioTransport* transport) {
+          audio_transport_proxy_ = transport;
+          return 0;
+        }));
+
+    ON_CALL(voice_engine_, audio_transport())
+        .WillByDefault(testing::Return(&original_audio_transport_));
+
+    EXPECT_CALL(voice_engine_, audio_device_module());
+
+    AudioState::Config config;
+    config.voice_engine = &voice_engine_;
+    config.audio_mixer = audio_mixer_;
+
+    audio_state_.reset(new internal::AudioState(config));
+  }
+
+  rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer_; }
+
+  AudioTransport* audio_transport_proxy() { return audio_transport_proxy_; }
+
+  MockAudioTransport& audio_transport() { return original_audio_transport_; }
+
+ private:
+  testing::StrictMock<MockVoiceEngine> voice_engine_;
+  MockAudioTransport original_audio_transport_;
+  std::unique_ptr<internal::AudioState> audio_state_;
+  AudioTransport* audio_transport_proxy_ = nullptr;
+  rtc::scoped_refptr<AudioMixer> audio_mixer_;
+};
+
+namespace {
+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_F(AudioStateAudioPathTest, RecordedAudioArrivesAtOriginalTransport) {
+  // Setup completed. Ensure call of old transport is forwarded to new.
+  uint32_t new_mic_level;
+  EXPECT_CALL(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_F(AudioStateAudioPathTest,
+       QueryingProxyForAudioShouldResultInGetAudioCallOnMixerSource) {
+  FakeAudioSource fake_source;
+
+  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