1 files changed, 134 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/audio/audio_state_unittest.cc b/third_party/libwebrtc/webrtc/audio/audio_state_unittest.cc
new file mode 100644
index 0000000000..28b0a715f6
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/audio/audio_state_unittest.cc
@@ -0,0 +1,134 @@
+/*
+ *  Copyright (c) 2015 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 "audio/audio_state.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "test/gtest.h"
+#include "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()) {
+    EXPECT_CALL(mock_voice_engine, audio_transport())
+        .WillRepeatedly(testing::Return(&audio_transport));
+
+    audio_state_config.voice_engine = &mock_voice_engine;
+    audio_state_config.audio_mixer = audio_mixer;
+    audio_state_config.audio_processing =
+        new rtc::RefCountedObject<MockAudioProcessing>();
+  }
+  AudioState::Config& config() { return audio_state_config; }
+  MockVoiceEngine& voice_engine() { return mock_voice_engine; }
+  rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
+  MockAudioTransport& original_audio_transport() { return audio_transport; }
+
+ private:
+  testing::StrictMock<MockVoiceEngine> mock_voice_engine;
+  AudioState::Config audio_state_config;
+  rtc::scoped_refptr<AudioMixer> audio_mixer;
+  MockAudioTransport audio_transport;
+};
+
+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(AudioStateTest, Create) {
+  ConfigHelper helper;
+  rtc::scoped_refptr<AudioState> audio_state =
+      AudioState::Create(helper.config());
+  EXPECT_TRUE(audio_state.get());
+}
+
+TEST(AudioStateTest, ConstructDestruct) {
+  ConfigHelper helper;
+  std::unique_ptr<internal::AudioState> audio_state(
+      new internal::AudioState(helper.config()));
+}
+
+TEST(AudioStateTest, GetVoiceEngine) {
+  ConfigHelper helper;
+  std::unique_ptr<internal::AudioState> audio_state(
+      new internal::AudioState(helper.config()));
+  EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine());
+}
+
+// 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)));
+
+  audio_state->audio_transport()->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;
+  audio_state->audio_transport()->NeedMorePlayData(
+      kSampleRate / 100, kBytesPerSample, kNumberOfChannels, kSampleRate,
+      audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
+}
+}  // namespace test
+}  // namespace webrtc