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Unified Diff: content/renderer/media/webrtc_audio_capturer_unittest.cc

Issue 1966043006: Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 7 months ago
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Index: content/renderer/media/webrtc_audio_capturer_unittest.cc
diff --git a/content/renderer/media/webrtc_audio_capturer_unittest.cc b/content/renderer/media/webrtc_audio_capturer_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..373b95ba50e17a28e962d3c769253082cb899ab2
--- /dev/null
+++ b/content/renderer/media/webrtc_audio_capturer_unittest.cc
@@ -0,0 +1,152 @@
+// Copyright 2013 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "base/logging.h"
+#include "build/build_config.h"
+#include "content/public/renderer/media_stream_audio_sink.h"
+#include "content/renderer/media/mock_constraint_factory.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
+#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc_local_audio_track.h"
+#include "media/base/audio_bus.h"
+#include "media/base/audio_parameters.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+
+namespace content {
+
+namespace {
+
+class MockCapturerSource : public media::AudioCapturerSource {
+ public:
+ MockCapturerSource() {}
+ MOCK_METHOD3(Initialize, void(const media::AudioParameters& params,
+ CaptureCallback* callback,
+ int session_id));
+ MOCK_METHOD0(Start, void());
+ MOCK_METHOD0(Stop, void());
+ MOCK_METHOD1(SetVolume, void(double volume));
+ MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
+
+ protected:
+ ~MockCapturerSource() override {}
+};
+
+class MockMediaStreamAudioSink : public MediaStreamAudioSink {
+ public:
+ MockMediaStreamAudioSink() {}
+ ~MockMediaStreamAudioSink() override {}
+ void OnData(const media::AudioBus& audio_bus,
+ base::TimeTicks estimated_capture_time) override {
+ EXPECT_EQ(audio_bus.channels(), params_.channels());
+ EXPECT_EQ(audio_bus.frames(), params_.frames_per_buffer());
+ EXPECT_FALSE(estimated_capture_time.is_null());
+ OnDataCallback();
+ }
+ MOCK_METHOD0(OnDataCallback, void());
+ void OnSetFormat(const media::AudioParameters& params) override {
+ params_ = params;
+ FormatIsSet();
+ }
+ MOCK_METHOD0(FormatIsSet, void());
+
+ private:
+ media::AudioParameters params_;
+};
+
+} // namespace
+
+class WebRtcAudioCapturerTest : public testing::Test {
+ protected:
+ WebRtcAudioCapturerTest()
+#if defined(OS_ANDROID)
+ : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+ media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) {
+ // Android works with a buffer size bigger than 20ms.
+#else
+ : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+ media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
+#endif
+ }
+
+ void VerifyAudioParams(const blink::WebMediaConstraints& constraints,
+ bool need_audio_processing) {
+ const std::unique_ptr<WebRtcAudioCapturer> capturer =
+ WebRtcAudioCapturer::CreateCapturer(
+ -1, StreamDeviceInfo(
+ MEDIA_DEVICE_AUDIO_CAPTURE, "", "", params_.sample_rate(),
+ params_.channel_layout(), params_.frames_per_buffer()),
+ constraints, nullptr, nullptr);
+ const scoped_refptr<MockCapturerSource> capturer_source(
+ new MockCapturerSource());
+ EXPECT_CALL(*capturer_source.get(), Initialize(_, capturer.get(), -1));
+ EXPECT_CALL(*capturer_source.get(), SetAutomaticGainControl(true));
+ EXPECT_CALL(*capturer_source.get(), Start());
+ capturer->SetCapturerSource(capturer_source, params_);
+
+ scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
+ WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
+ const std::unique_ptr<WebRtcLocalAudioTrack> track(
+ new WebRtcLocalAudioTrack(adapter.get()));
+ capturer->AddTrack(track.get());
+
+ // Connect a mock sink to the track.
+ std::unique_ptr<MockMediaStreamAudioSink> sink(
+ new MockMediaStreamAudioSink());
+ track->AddSink(sink.get());
+
+ int delay_ms = 65;
+ bool key_pressed = true;
+ double volume = 0.9;
+
+ std::unique_ptr<media::AudioBus> audio_bus =
+ media::AudioBus::Create(params_);
+ audio_bus->Zero();
+
+ media::AudioCapturerSource::CaptureCallback* callback =
+ static_cast<media::AudioCapturerSource::CaptureCallback*>(
+ capturer.get());
+
+ // Verify the sink is getting the correct values.
+ EXPECT_CALL(*sink, FormatIsSet());
+ EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1));
+ callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed);
+
+ track->RemoveSink(sink.get());
+ EXPECT_CALL(*capturer_source.get(), Stop());
+ capturer->Stop();
+ }
+
+ media::AudioParameters params_;
+};
+
+TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) {
+ // Turn off the default constraints to verify that the sink will get packets
+ // with a buffer size smaller than 10ms.
+ MockConstraintFactory constraint_factory;
+ constraint_factory.DisableDefaultAudioConstraints();
+ VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false);
+}
+
+TEST_F(WebRtcAudioCapturerTest, FailToCreateCapturerWithWrongConstraints) {
+ MockConstraintFactory constraint_factory;
+ const std::string dummy_constraint = "dummy";
+ // Set a non-audio constraint.
+ constraint_factory.basic().width.setExact(240);
+
+ std::unique_ptr<WebRtcAudioCapturer> capturer(
+ WebRtcAudioCapturer::CreateCapturer(
+ 0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "",
+ params_.sample_rate(), params_.channel_layout(),
+ params_.frames_per_buffer()),
+ constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
+ EXPECT_TRUE(capturer.get() == NULL);
+}
+
+
+} // namespace content
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