From fbcb5ceb166ed6b51be1da366817d64ecc86927a Mon Sep 17 00:00:00 2001
From: "pbos@webrtc.org" <pbos@webrtc.org>
Date: Wed, 11 Feb 2015 14:24:44 +0000
Subject: [PATCH] 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
---
 webrtc/video/video_send_stream_tests.cc | 82 -------------------------
 1 file changed, 82 deletions(-)

diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index a6d77543f5..79904ef184 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -915,88 +915,6 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
   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