Remove VideoSendStreamTest.ProducesStats.
This test is covered by EndToEndTests.GetStats and there's no need for a
duplicate test.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39049004
Cr-Commit-Position: refs/heads/master@{#8332}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8332 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index a6d7754..79904ef 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -915,88 +915,6 @@
RunBaseTest(&test);
}
-TEST_F(VideoSendStreamTest, ProducesStats) {
- class ProducesStats : public test::SendTest {
- public:
- ProducesStats()
- : SendTest(kDefaultTimeoutMs),
- stream_(NULL),
- event_(EventWrapper::Create()) {}
-
- virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
- event_->Set();
-
- return SEND_PACKET;
- }
-
- private:
- bool WaitForFilledStats() {
- Clock* clock = Clock::GetRealTimeClock();
- int64_t now = clock->TimeInMilliseconds();
- int64_t stop_time = now + kDefaultTimeoutMs;
- while (now < stop_time) {
- int64_t time_left = stop_time - now;
- if (time_left > 0 && event_->Wait(time_left) == kEventSignaled &&
- CheckStats()) {
- return true;
- }
- now = clock->TimeInMilliseconds();
- }
- return false;
- }
-
- bool CheckStats() {
- VideoSendStream::Stats stats = stream_->GetStats();
- // Check that all applicable data sources have been used.
- if (stats.input_frame_rate > 0 && stats.encode_frame_rate > 0
- && !stats.substreams.empty()) {
- uint32_t ssrc = stats.substreams.begin()->first;
- EXPECT_NE(
- config_.rtp.ssrcs.end(),
- std::find(
- config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc));
- // Check for data populated by various sources. RTCP excluded as this
- // data is received from remote side. Tested in call tests instead.
- const SsrcStats& entry = stats.substreams[ssrc];
- if (entry.frame_counts.key_frames > 0 &&
- entry.frame_counts.delta_frames > 0 &&
- entry.total_bitrate_bps > 0 &&
- entry.rtp_stats.transmitted.packets > 0u &&
- entry.avg_delay_ms > 0 && entry.max_delay_ms > 0) {
- return true;
- }
- }
- return false;
- }
-
- void SetConfig(const VideoSendStream::Config& config) { config_ = config; }
-
- virtual void ModifyConfigs(
- VideoSendStream::Config* send_config,
- std::vector<VideoReceiveStream::Config>* receive_configs,
- VideoEncoderConfig* encoder_config) OVERRIDE {
- SetConfig(*send_config);
- }
-
- virtual void OnStreamsCreated(
- VideoSendStream* send_stream,
- const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
- stream_ = send_stream;
- }
-
- virtual void PerformTest() OVERRIDE {
- EXPECT_TRUE(WaitForFilledStats())
- << "Timed out waiting for filled statistics.";
- }
-
- VideoSendStream* stream_;
- VideoSendStream::Config config_;
- scoped_ptr<EventWrapper> event_;
- } test;
-
- RunBaseTest(&test);
-}
-
// This test first observes "high" bitrate use at which point it sends a REMB to
// indicate that it should be lowered significantly. The test then observes that
// the bitrate observed is sinking well below the min-transmit-bitrate threshold
