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

Issue 190643003: Fix/Add content_unittests to work for the code behind --enable-audio-track-processing (Closed) Base URL: http://git.chromium.org/chromium/src.git@master
Patch Set: removed an empty line. Created 6 years, 9 months ago
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Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
index 4a02d35d2c1180c09ec719d2ef9b1bbff60c91e3..8c387bbe3e6c35d9d98515bf8b8b6ec8d2db6467 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -4,6 +4,7 @@
#include "base/synchronization/waitable_event.h"
#include "base/test/test_timeouts.h"
+#include "content/renderer/media/mock_media_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_audio_device_impl.h"
@@ -134,26 +135,32 @@ class MockMediaStreamAudioSink : public PeerConnectionAudioSink {
int current_volume,
bool need_audio_processing,
bool key_pressed) OVERRIDE {
+ EXPECT_EQ(params_.sample_rate(), sample_rate);
+ EXPECT_EQ(params_.channels(), number_of_channels);
+ EXPECT_EQ(params_.frames_per_buffer(), number_of_frames);
CaptureData(channels.size(),
- sample_rate,
- number_of_channels,
- number_of_frames,
audio_delay_milliseconds,
current_volume,
need_audio_processing,
key_pressed);
return 0;
}
- MOCK_METHOD8(CaptureData,
+ MOCK_METHOD5(CaptureData,
void(int number_of_network_channels,
- int sample_rate,
- int number_of_channels,
- int number_of_frames,
int audio_delay_milliseconds,
int current_volume,
bool need_audio_processing,
bool key_pressed));
- MOCK_METHOD1(OnSetFormat, void(const media::AudioParameters& params));
+ void OnSetFormat(const media::AudioParameters& params) {
+ params_ = params;
+ FormatIsSet();
+ }
+ MOCK_METHOD0(FormatIsSet, void());
+
+ const media::AudioParameters& audio_params() const { return params_; }
+
+ private:
+ media::AudioParameters params_;
};
} // namespace
@@ -202,21 +209,16 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
adapter.get())->GetRenderer()->AddChannel(i);
}
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
- const media::AudioParameters params = capturer_->source_audio_parameters();
base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(Return());
+ EXPECT_CALL(*sink, FormatIsSet());
EXPECT_CALL(*sink,
CaptureData(kNumberOfNetworkChannels,
- params.sample_rate(),
- params.channels(),
- params.sample_rate() / 100,
0,
0,
- true,
+ _,
false)).Times(AtLeast(1))
.WillRepeatedly(SignalEvent(&event));
track->AddSink(sink.get());
-
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
track->RemoveSink(sink.get());
@@ -247,29 +249,17 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
const media::AudioParameters params = capturer_->source_audio_parameters();
base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink, OnSetFormat(_)).Times(1);
+ EXPECT_CALL(*sink, FormatIsSet()).Times(1);
EXPECT_CALL(*sink,
- CaptureData(1,
- params.sample_rate(),
- params.channels(),
- params.sample_rate() / 100,
- 0,
- 0,
- true,
- false)).Times(0);
+ CaptureData(1, 0, 0, _, false)).Times(0);
+ EXPECT_EQ(sink->audio_params().frames_per_buffer(),
+ params.sample_rate() / 100);
track->AddSink(sink.get());
EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
event.Reset();
EXPECT_CALL(*sink,
- CaptureData(1,
- params.sample_rate(),
- params.channels(),
- params.sample_rate() / 100,
- 0,
- 0,
- true,
- false)).Times(AtLeast(1))
+ CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
.WillRepeatedly(SignalEvent(&event));
EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
@@ -299,17 +289,12 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
const media::AudioParameters params = capturer_->source_audio_parameters();
base::WaitableEvent event_1(false, false);
- EXPECT_CALL(*sink_1, OnSetFormat(_)).WillOnce(Return());
+ EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
EXPECT_CALL(*sink_1,
- CaptureData(1,
- params.sample_rate(),
- params.channels(),
- params.sample_rate() / 100,
- 0,
- 0,
- true,
- false)).Times(AtLeast(1))
+ CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
.WillRepeatedly(SignalEvent(&event_1));
+ EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
+ params.sample_rate() / 100);
track_1->AddSink(sink_1.get());
EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
@@ -329,27 +314,15 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
base::WaitableEvent event_2(false, false);
scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
- EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(Return());
- EXPECT_CALL(*sink_1,
- CaptureData(1,
- params.sample_rate(),
- params.channels(),
- params.sample_rate() / 100,
- 0,
- 0,
- true,
- false)).Times(AtLeast(1))
+ EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
+ EXPECT_CALL(*sink_1, CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
.WillRepeatedly(SignalEvent(&event_1));
- EXPECT_CALL(*sink_2,
- CaptureData(1,
- params.sample_rate(),
- params.channels(),
- params.sample_rate() / 100,
- 0,
- 0,
- true,
- false)).Times(AtLeast(1))
+ EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
+ params.sample_rate() / 100);
+ EXPECT_CALL(*sink_2, CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
.WillRepeatedly(SignalEvent(&event_2));
+ EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
+ params.sample_rate() / 100);
track_2->AddSink(sink_2.get());
EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
@@ -408,8 +381,8 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
// Verify the data flow by connecting the sink to |track_1|.
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
event.Reset();
- EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(SignalEvent(&event));
- EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, true, false))
+ EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
+ EXPECT_CALL(*sink, CaptureData(_, 0, 0, _, false))
.Times(AnyNumber()).WillRepeatedly(Return());
track_1->AddSink(sink.get());
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
@@ -433,7 +406,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
// Adding a new track to the capturer.
track_2->AddSink(sink.get());
- EXPECT_CALL(*sink, OnSetFormat(_)).Times(0);
+ EXPECT_CALL(*sink, FormatIsSet()).Times(0);
// Stop the capturer again will not trigger stopping the source of the
// capturer again..
@@ -463,12 +436,11 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
}
// Verify the data flow by connecting the |sink_1| to |track_1|.
scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
- EXPECT_CALL(
- *sink_1.get(),
- CaptureData(
- kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, true, false))
+ EXPECT_CALL(*sink_1.get(),
+ CaptureData(kNumberOfNetworkChannelsForTrack1,
+ 0, 0, _, false))
.Times(AnyNumber()).WillRepeatedly(Return());
- EXPECT_CALL(*sink_1.get(), OnSetFormat(_)).Times(AnyNumber());
+ EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
track_1->AddSink(sink_1.get());
// Create a new capturer with new source with different audio format.
@@ -505,13 +477,10 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
// Verify the data flow by connecting the |sink_2| to |track_2|.
scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
base::WaitableEvent event(false, false);
- EXPECT_CALL(
- *sink_2,
- CaptureData(
- kNumberOfNetworkChannelsForTrack2, new_param.sample_rate(),
- new_param.channels(), _, 0, 0, true, false))
+ EXPECT_CALL(*sink_2,
+ CaptureData(kNumberOfNetworkChannelsForTrack2, 0, 0, _, false))
.Times(AnyNumber()).WillRepeatedly(Return());
- EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(SignalEvent(&event));
+ EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
track_2->AddSink(sink_2.get());
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
@@ -536,7 +505,8 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
// Create a capturer with new source which works with the format above.
- blink::WebMediaConstraints constraints;
+ MockMediaConstraintFactory factory;
+ factory.DisableDefaultAudioConstraints();
scoped_refptr<WebRtcAudioCapturer> capturer(
WebRtcAudioCapturer::CreateCapturer(
-1,
@@ -544,7 +514,7 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
"", "", params.sample_rate(),
params.channel_layout(),
params.frames_per_buffer()),
- constraints,
+ factory.CreateWebMediaConstraints(),
NULL));
scoped_refptr<MockCapturerSource> source(
new MockCapturerSource(capturer.get()));
@@ -565,7 +535,7 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
// Verify the data flow by connecting the |sink| to |track|.
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink, OnSetFormat(_)).Times(1);
+ EXPECT_CALL(*sink, FormatIsSet()).Times(1);
// Verify the sinks are getting the packets with an expecting buffer size.
#if defined(OS_ANDROID)
const int expected_buffer_size = params.sample_rate() / 100;
@@ -573,11 +543,11 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
const int expected_buffer_size = params.frames_per_buffer();
#endif
EXPECT_CALL(*sink, CaptureData(
- 0, params.sample_rate(), params.channels(), expected_buffer_size,
- 0, 0, true, false))
+ 0, 0, 0, _, false))
.Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
track->AddSink(sink.get());
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
+ EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
// Stopping the new source will stop the second track.
EXPECT_CALL(*source, OnStop()).Times(1);
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