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

Issue 90743004: Add generic interfaces for the sinks of the media stream audio track (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: fixed the bots Created 7 years ago
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« content/renderer/media/remote_media_stream_impl.cc ('K') | « content/renderer/media/webrtc_local_audio_track.cc ('k') | no next file » | no next file with comments »
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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 aac06d0ed400420119d3faa0e1178fa8a0776fb6..f5b668a2fedfe304c32aae43994da0f21c14aa34 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -6,6 +6,7 @@
#include "base/test/test_timeouts.h"
#include "content/renderer/media/rtc_media_constraints.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
#include "content/renderer/media/webrtc_local_audio_source_provider.h"
#include "content/renderer/media/webrtc_local_audio_track.h"
#include "media/audio/audio_parameters.h"
@@ -118,19 +119,20 @@ class MockCapturerSource : public media::AudioCapturerSource {
media::AudioParameters params_;
};
-class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
+// TODO(xians): Use MediaStreamAudioSink.
+class MockMediaStreamAudioSink : public PeerConnectionAudioSink {
public:
- MockWebRtcAudioCapturerSink() {}
- ~MockWebRtcAudioCapturerSink() {}
- int CaptureData(const std::vector<int>& channels,
- const int16* audio_data,
- 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) OVERRIDE {
+ MockMediaStreamAudioSink() {}
+ ~MockMediaStreamAudioSink() {}
+ int OnData(const int16* audio_data,
+ int sample_rate,
+ int number_of_channels,
+ int number_of_frames,
+ const std::vector<int>& channels,
+ int audio_delay_milliseconds,
+ int current_volume,
+ bool need_audio_processing,
+ bool key_pressed) OVERRIDE {
CaptureData(channels.size(),
sample_rate,
number_of_channels,
@@ -150,7 +152,7 @@ class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
int current_volume,
bool need_audio_processing,
bool key_pressed));
- MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
+ MOCK_METHOD1(OnSetFormat, void(const media::AudioParameters& params));
};
} // namespace
@@ -196,11 +198,10 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
static_cast<webrtc::AudioTrackInterface*>(track.get())->
GetRenderer()->AddChannel(i);
}
- scoped_ptr<MockWebRtcAudioCapturerSink> sink(
- new MockWebRtcAudioCapturerSink());
+ scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
const media::AudioParameters params = capturer_->audio_parameters();
base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
+ EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(Return());
EXPECT_CALL(*sink,
CaptureData(kNumberOfNetworkChannels,
params.sample_rate(),
@@ -240,11 +241,10 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
GetRenderer()->AddChannel(0);
EXPECT_TRUE(track->enabled());
EXPECT_TRUE(track->set_enabled(false));
- scoped_ptr<MockWebRtcAudioCapturerSink> sink(
- new MockWebRtcAudioCapturerSink());
+ scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
const media::AudioParameters params = capturer_->audio_parameters();
base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
+ EXPECT_CALL(*sink, OnSetFormat(_)).Times(1);
EXPECT_CALL(*sink,
CaptureData(1,
params.sample_rate(),
@@ -293,11 +293,10 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
GetRenderer()->AddChannel(0);
EXPECT_TRUE(track_1->enabled());
- scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
- new MockWebRtcAudioCapturerSink());
+ scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
const media::AudioParameters params = capturer_->audio_parameters();
base::WaitableEvent event_1(false, false);
- EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
+ EXPECT_CALL(*sink_1, OnSetFormat(_)).WillOnce(Return());
EXPECT_CALL(*sink_1,
CaptureData(1,
params.sample_rate(),
@@ -325,9 +324,8 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
event_1.Reset();
base::WaitableEvent event_2(false, false);
- scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
- new MockWebRtcAudioCapturerSink());
- EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
+ scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
+ EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(Return());
EXPECT_CALL(*sink_1,
CaptureData(1,
params.sample_rate(),
@@ -404,10 +402,9 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
// Verify the data flow by connecting the sink to |track_1|.
- scoped_ptr<MockWebRtcAudioCapturerSink> sink(
- new MockWebRtcAudioCapturerSink());
+ scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
event.Reset();
- EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(SignalEvent(&event));
+ EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(SignalEvent(&event));
EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
.Times(AnyNumber()).WillRepeatedly(Return());
track_1->AddSink(sink.get());
@@ -431,7 +428,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
// Adding a new track to the capturer.
track_2->AddSink(sink.get());
- EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(0);
+ EXPECT_CALL(*sink, OnSetFormat(_)).Times(0);
// Stop the capturer again will not trigger stopping the source of the
// capturer again..
@@ -490,14 +487,13 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
GetRenderer()->AddChannel(i);
}
// Verify the data flow by connecting the |sink_1| to |track_1|.
- scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
- new MockWebRtcAudioCapturerSink());
+ scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
EXPECT_CALL(
*sink_1.get(),
CaptureData(
kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
.Times(AnyNumber()).WillRepeatedly(Return());
- EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(AnyNumber());
+ EXPECT_CALL(*sink_1.get(), OnSetFormat(_)).Times(AnyNumber());
track_1->AddSink(sink_1.get());
// Create a new capturer with new source with different audio format.
@@ -527,15 +523,14 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
GetRenderer()->AddChannel(i);
}
// Verify the data flow by connecting the |sink_2| to |track_2|.
- scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
- new MockWebRtcAudioCapturerSink());
+ scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
base::WaitableEvent event(false, false);
EXPECT_CALL(
*sink_2,
CaptureData(
kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false))
.Times(AnyNumber()).WillRepeatedly(Return());
- EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(SignalEvent(&event));
+ EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(SignalEvent(&event));
track_2->AddSink(sink_2.get());
EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
@@ -583,10 +578,9 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
track->Start();
// Verify the data flow by connecting the |sink| to |track|.
- scoped_ptr<MockWebRtcAudioCapturerSink> sink(
- new MockWebRtcAudioCapturerSink());
+ scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
base::WaitableEvent event(false, false);
- EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
+ EXPECT_CALL(*sink, OnSetFormat(_)).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;
« content/renderer/media/remote_media_stream_impl.cc ('K') | « content/renderer/media/webrtc_local_audio_track.cc ('k') | no next file » | no next file with comments »

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