Periodically update channel parameters and send TargetBitrate message.

webrtc/video/vie_encoder_unittest.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <limits>
 #include <utility>
 
+#include "webrtc/base/fakeclock.h"

[mflodman, 2017/01/13 13:29:38]

AFAICT fakeclock.h depends on messagequeue.h, which is only in rtc_base and not
in rtc_base_approved.

Do we need to use rtc::ScopedFakeClock?

 #include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/modules/video_coding/utility/default_video_bitrate_allocator.h"
 #include "webrtc/system_wrappers/include/metrics_default.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator.h"
+#include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video/vie_encoder.h"
 
 namespace {
 #if defined(WEBRTC_ANDROID)
 // TODO(kthelgason): Lower this limit when better testing
 // on MediaCodec and fallback implementations are in place.
 const int kMinPixelsPerFrame = 320 * 180;
 #else
 const int kMinPixelsPerFrame = 120 * 90;
 #endif
 }
 
 namespace webrtc {
 
 using DegredationPreference = VideoSendStream::DegradationPreference;
 using ScaleReason = ScalingObserverInterface::ScaleReason;
+using ::testing::_;
+using ::testing::Return;
 
 namespace {
 const size_t kMaxPayloadLength = 1440;
 const int kTargetBitrateBps = 100000;
 
 class TestBuffer : public webrtc::I420Buffer {
  public:
   TestBuffer(rtc::Event* event, int width, int height)
       : I420Buffer(width, height), event_(event) {}
 
@@ skipping 108 matching lines @@
     video_send_config_.encoder_settings.payload_name = payload_name;
 
     VideoEncoderConfig video_encoder_config;
     video_encoder_config.number_of_streams = num_streams;
     video_encoder_config.max_bitrate_bps = 1000000;
     video_encoder_config.video_stream_factory =
         new rtc::RefCountedObject<VideoStreamFactory>(num_temporal_layers);
     ConfigureEncoder(std::move(video_encoder_config), nack_enabled);
   }
 
-  VideoFrame CreateFrame(int64_t ntp_ts, rtc::Event* destruction_event) const {
+  VideoFrame CreateFrame(int64_t ntp_time_ms,
+                         rtc::Event* destruction_event) const {
     VideoFrame frame(new rtc::RefCountedObject<TestBuffer>(
                          destruction_event, codec_width_, codec_height_),
                      99, 99, kVideoRotation_0);
-    frame.set_ntp_time_ms(ntp_ts);
+    frame.set_ntp_time_ms(ntp_time_ms);
     return frame;
   }
 
-  VideoFrame CreateFrame(int64_t ntp_ts, int width, int height) const {
+  VideoFrame CreateFrame(int64_t ntp_time_ms, int width, int height) const {
     VideoFrame frame(
         new rtc::RefCountedObject<TestBuffer>(nullptr, width, height), 99, 99,
         kVideoRotation_0);
-    frame.set_ntp_time_ms(ntp_ts);
+    frame.set_ntp_time_ms(ntp_time_ms);
     return frame;
   }
 
   class TestEncoder : public test::FakeEncoder {
    public:
     TestEncoder()
         : FakeEncoder(Clock::GetRealTimeClock()),
           continue_encode_event_(false, false) {}
 
     VideoCodec codec_config() {
@@ skipping 783 matching lines @@
 
   vie_encoder_->Stop();
 
   stats_proxy_.reset();
   EXPECT_EQ(1,
             metrics::NumSamples("WebRTC.Video.CpuLimitedResolutionInPercent"));
   EXPECT_EQ(
       1, metrics::NumEvents("WebRTC.Video.CpuLimitedResolutionInPercent", 50));
 }
 
+TEST_F(ViEEncoderTest, CallsBitrateObserver) {
+  class MockBitrateObserver : public VideoBitrateAllocationObserver {
+   public:
+    MOCK_METHOD1(OnBitrateAllocationUpdated, void(const BitrateAllocation&));
+  } bitrate_observer;
+  vie_encoder_->SetBitrateObserver(&bitrate_observer);
+
+  const int kDefaultFps = 30;
+  const BitrateAllocation expected_bitrate =
+      DefaultVideoBitrateAllocator(fake_encoder_.codec_config())
+          .GetAllocation(kTargetBitrateBps, kDefaultFps);
+
+  rtc::ScopedFakeClock fake_clock;
+
+  // First called on bitrate updated, then again on first frame.
+  EXPECT_CALL(bitrate_observer, OnBitrateAllocationUpdated(expected_bitrate))
+      .Times(2);
+  vie_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
+
+  const int64_t kStartTimeMs = 1;
+  video_source_.IncomingCapturedFrame(
+      CreateFrame(kStartTimeMs, codec_width_, codec_height_));
+  sink_.WaitForEncodedFrame(kStartTimeMs);
+
+  // Not called on second frame.
+  fake_clock.AdvanceTimeMicros(rtc::kNumMicrosecsPerMillisec);
+  EXPECT_CALL(bitrate_observer, OnBitrateAllocationUpdated(expected_bitrate))
+      .Times(0);
+  video_source_.IncomingCapturedFrame(
+      CreateFrame(kStartTimeMs + 1, codec_width_, codec_height_));
+  sink_.WaitForEncodedFrame(kStartTimeMs + 1);
+
+  // Called after a process interval.
+  const int64_t kProcessIntervalMs =
+      vcm::VCMProcessTimer::kDefaultProcessIntervalMs;
+  fake_clock.AdvanceTimeMicros(rtc::kNumMicrosecsPerMillisec *
+                               kProcessIntervalMs);
+  EXPECT_CALL(bitrate_observer, OnBitrateAllocationUpdated(expected_bitrate))
+      .Times(1);
+  video_source_.IncomingCapturedFrame(CreateFrame(
+      kStartTimeMs + kProcessIntervalMs, codec_width_, codec_height_));
+  sink_.WaitForEncodedFrame(kStartTimeMs + kProcessIntervalMs);
+
+  vie_encoder_->Stop();
+}
+
 }  // namespace webrtc