Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_state_audio_path_unittest.cc

+/*

[ossu, 2016/11/21 17:07:32]

As we spoke about offline, I don't think you should add a new file for these two
unit tests. It's not how other unit tests in WebRTC are laid out, and I think
you could easily merge the functionality in the two files. Also, they're pretty
small, so there's no mental overhead in having them all in the same file.

[aleloi, 2016/11/22 13:23:40]

Done.

+ *  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 {
+
+const int kSampleRate = 8000;
+const int kNumberOfChannels = 1;
+const int kBytesPerSample = 2;
+
+struct ConfigHelper {
+  ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
+    using testing::_;
+
+    EXPECT_CALL(mock_voice_engine, RegisterVoiceEngineObserver(testing::_))
+        .WillOnce(testing::Return(0));
+    EXPECT_CALL(mock_voice_engine, DeRegisterVoiceEngineObserver())
+        .WillOnce(testing::Return(0));
+    EXPECT_CALL(mock_voice_engine, audio_processing());
+    EXPECT_CALL(mock_voice_engine, audio_transport());
+
+    ON_CALL(mock_voice_engine, audio_transport())
+        .WillByDefault(testing::Return(&audio_transport));
+
+    audio_state_config.voice_engine = &mock_voice_engine;
+    audio_state_config.audio_mixer = audio_mixer;
+
+    EXPECT_CALL(voice_engine(), audio_device_module()).Times(2);
+    auto device = static_cast<MockAudioDeviceModule*>(
+        voice_engine().audio_device_module());
+
+    // Populate the audio transport proxy pointer to the most recent
+    // transport connected to the Audio Device.
+    ON_CALL(*device, RegisterAudioCallback(testing::_))
+        .WillByDefault(testing::Invoke([this](AudioTransport* transport) {
+          registered_audio_transport = transport;
+          return 0;
+        }));
+  }
+
+  rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
+
+  MockAudioTransport& original_audio_transport() { return audio_transport; }
+  AudioTransport* audio_transport_proxy() { return registered_audio_transport; }
+
+  AudioState::Config& config() { return audio_state_config; }
+  MockVoiceEngine& voice_engine() { return mock_voice_engine; }

[the sun, 2016/11/22 08:47:45]

nit: this doesn't strictly need to be public...

+
+ private:
+  testing::StrictMock<MockVoiceEngine> mock_voice_engine;
+  MockAudioTransport audio_transport;
+  rtc::scoped_refptr<AudioMixer> audio_mixer;
+  AudioTransport* registered_audio_transport = nullptr;
+  AudioState::Config audio_state_config;
+};
+
+class FakeAudioSource : public AudioMixer::Source {
+ public:
+  // TODO(aleloi): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
+  int Ssrc() const /*override*/ { return 0; }
+
+  int PreferredSampleRate() const /*override*/ { return kSampleRate; }
+
+  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;
+
+  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, kSampleRate / 100, kBytesPerSample,
+                              kNumberOfChannels, kSampleRate, 0, 0, 0, false,
+                              testing::Ref(new_mic_level)));
+
+  helper.audio_transport_proxy()->RecordedDataIsAvailable(
+      nullptr, kSampleRate / 100, kBytesPerSample, kNumberOfChannels,
+      kSampleRate, 0, 0, 0, false, new_mic_level);
+}
+
+TEST(AudioStateAudioPathTest,
+     QueryingProxyForAudioShouldResultInGetAudioCallOnMixerSource) {
+  ConfigHelper helper;
+
+  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_ = kNumberOfChannels;
+            return AudioMixer::Source::AudioFrameInfo::kNormal;
+          }));
+
+  int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
+  size_t n_samples_out;
+  int64_t elapsed_time_ms;
+  int64_t ntp_time_ms;
+  helper.audio_transport_proxy()->NeedMorePlayData(
+      kSampleRate / 100, kBytesPerSample, kNumberOfChannels, kSampleRate,
+      audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
+}
+}  // namespace test
+}  // namespace webrtc