diff options
author | tapted@chromium.org <tapted@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-10-14 02:04:43 +0000 |
---|---|---|
committer | tapted@chromium.org <tapted@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-10-14 02:04:43 +0000 |
commit | 166df683c229ed6c028afeb3e7bbe77f2565bda1 (patch) | |
tree | 289cf1c7f34a108e8ab5a51a16f8236040acb148 /media | |
parent | 7ca78e8c5785b5238c8a2ddedb2fcb0740497981 (diff) | |
download | chromium_src-166df683c229ed6c028afeb3e7bbe77f2565bda1.zip chromium_src-166df683c229ed6c028afeb3e7bbe77f2565bda1.tar.gz chromium_src-166df683c229ed6c028afeb3e7bbe77f2565bda1.tar.bz2 |
Revert 228002 "Enable cast_unittests"
Breaks compile on the `All` target on win_x64_rel. Note that this is only
built by default on manually submitted tryjobs. Not CQ or waterfall.
(see bug). For the errors see, e.g.,
http://build.chromium.org/p/tryserver.chromium/builders/win_x64_rel/builds/45000/
BUG=304877
> Enable cast_unittests
> And fixed associated errors
>
> Review URL: https://codereview.chromium.org/26486005
TBR=pwestin@google.com
Review URL: https://codereview.chromium.org/27141002
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@228412 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'media')
-rw-r--r-- | media/cast/audio_receiver/audio_receiver_unittest.cc | 9 | ||||
-rw-r--r-- | media/cast/audio_sender/audio_encoder.cc | 2 | ||||
-rw-r--r-- | media/cast/cast.gyp | 11 | ||||
-rw-r--r-- | media/cast/video_receiver/video_receiver_unittest.cc | 8 | ||||
-rw-r--r-- | media/cast/video_sender/video_sender_unittest.cc | 2 |
5 files changed, 15 insertions, 17 deletions
diff --git a/media/cast/audio_receiver/audio_receiver_unittest.cc b/media/cast/audio_receiver/audio_receiver_unittest.cc index e35d3fa..72190f2 100644 --- a/media/cast/audio_receiver/audio_receiver_unittest.cc +++ b/media/cast/audio_receiver/audio_receiver_unittest.cc @@ -41,12 +41,7 @@ class TestAudioEncoderCallback : int number_times_called() { return num_called_;} - protected: - virtual ~TestAudioEncoderCallback() {} - private: - friend class base::RefCountedThreadSafe<TestAudioEncoderCallback>; - int num_called_; uint8 expected_frame_id_; base::TimeTicks expected_playout_time_; @@ -74,8 +69,8 @@ class AudioReceiverTest : public ::testing::Test { testing_clock_.Advance( base::TimeDelta::FromMilliseconds(kStartMillisecond)); task_runner_ = new test::FakeTaskRunner(&testing_clock_); - cast_environment_ = new CastEnvironment(&testing_clock_, task_runner_, - task_runner_, task_runner_, task_runner_, task_runner_); + cast_environment_ = new CastEnvironment(task_runner_, task_runner_, + task_runner_, task_runner_, task_runner_); test_audio_encoder_callback_ = new TestAudioEncoderCallback(); } diff --git a/media/cast/audio_sender/audio_encoder.cc b/media/cast/audio_sender/audio_encoder.cc index 5d9cbd5..bbfeab4 100644 --- a/media/cast/audio_sender/audio_encoder.cc +++ b/media/cast/audio_sender/audio_encoder.cc @@ -129,7 +129,7 @@ void AudioEncoder::EncodeAudioFrameThread( const FrameEncodedCallback& frame_encoded_callback, const base::Closure release_callback) { DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::AUDIO_ENCODER)); - size_t samples_per_10ms = audio_frame->frequency / 100; + int samples_per_10ms = audio_frame->frequency / 100; size_t number_of_10ms_blocks = audio_frame->samples.size() / (samples_per_10ms * audio_frame->channels); DCHECK(webrtc::AudioFrame::kMaxDataSizeSamples > samples_per_10ms) diff --git a/media/cast/cast.gyp b/media/cast/cast.gyp index d6dd00a8..bd0bd00 100644 --- a/media/cast/cast.gyp +++ b/media/cast/cast.gyp @@ -42,8 +42,13 @@ '<(DEPTH)/third_party/webrtc/', ], 'sources': [ - 'audio_receiver/audio_decoder_unittest.cc', - 'audio_receiver/audio_receiver_unittest.cc', + # TODO(hclam): These files are excluded because it triggers + # compiler warnings on Win x64. Refactor and build this + # file to appease the compiler. + # 'audio_receiver/audio_decoder_unittest.cc', + # 'audio_receiver/audio_receiver_unittest.cc', + # 'video_receiver/video_receiver_unittest.cc', + # 'video_sender/video_sender_unittest.cc', 'audio_sender/audio_encoder_unittest.cc', 'audio_sender/audio_sender_unittest.cc', 'congestion_control/congestion_control_unittest.cc', @@ -73,11 +78,9 @@ 'test/fake_task_runner.cc', 'test/video_utility.cc', 'video_receiver/video_decoder_unittest.cc', - 'video_receiver/video_receiver_unittest.cc', 'video_sender/mock_video_encoder_controller.cc', 'video_sender/mock_video_encoder_controller.h', 'video_sender/video_encoder_unittest.cc', - 'video_sender/video_sender_unittest.cc', ], # source }, ], # targets diff --git a/media/cast/video_receiver/video_receiver_unittest.cc b/media/cast/video_receiver/video_receiver_unittest.cc index 15fa1aa..3e76f20 100644 --- a/media/cast/video_receiver/video_receiver_unittest.cc +++ b/media/cast/video_receiver/video_receiver_unittest.cc @@ -70,10 +70,10 @@ class VideoReceiverTest : public ::testing::Test { config_.codec = kVp8; config_.use_external_decoder = false; task_runner_ = new test::FakeTaskRunner(&testing_clock_); - cast_environment_ = new CastEnvironment(&testing_clock_, task_runner_, - NULL, NULL, task_runner_, task_runner_); - receiver_.reset( - new PeerVideoReceiver(cast_environment_, config_, &mock_transport_)); + cast_environment_ = new CastEnvironment(task_runner_, NULL, NULL, + task_runner_, task_runner_); + receiver_.reset(new + PeerVideoReceiver(cast_environment_, config_, &mock_transport_)); testing_clock_.Advance( base::TimeDelta::FromMilliseconds(kStartMillisecond)); video_receiver_callback_ = new TestVideoReceiverCallback(); diff --git a/media/cast/video_sender/video_sender_unittest.cc b/media/cast/video_sender/video_sender_unittest.cc index 843b42f..b5819e3 100644 --- a/media/cast/video_sender/video_sender_unittest.cc +++ b/media/cast/video_sender/video_sender_unittest.cc @@ -153,7 +153,7 @@ TEST_F(VideoSenderTest, ExternalEncoder) { } TEST_F(VideoSenderTest, RtcpTimer) { - EXPECT_CALL(mock_transport_, SendRtcpPacket(_)).Times(1); + EXPECT_CALL(mock_transport_, SendRtcpPacket(_)).Times(2); InitEncoder(false); // Make sure that we send at least one RTCP packet. |