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Unified Diff: webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc

Issue 2126793002: Reset InterArrival if arrival time clock makes a jump. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix a few test issues. Created 4 years, 5 months ago
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Index: webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
diff --git a/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc b/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
new file mode 100644
index 0000000000000000000000000000000000000000..c6907f7a0b2822d75d0b2cbb84f0bffc8d5e3e21
--- /dev/null
+++ b/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
@@ -0,0 +1,500 @@
+/*
+ * 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 "webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h"
+
+#include <algorithm>
+#include <limits>
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/congestion_controller/delay_based_bwe.h"
+
+namespace webrtc {
+
+const size_t kMtu = 1200;
+const uint32_t kAcceptedBitrateErrorBps = 50000;
+
+// Number of packets needed before we have a valid estimate.
+const int kNumInitialPackets = 2;
+
+namespace test {
+
+void TestBitrateObserver::OnReceiveBitrateChanged(
+ const std::vector<uint32_t>& ssrcs,
+ uint32_t bitrate) {
+ latest_bitrate_ = bitrate;
+ updated_ = true;
+}
+
+RtpStream::RtpStream(int fps, int bitrate_bps)
+ : fps_(fps),
+ bitrate_bps_(bitrate_bps),
+ next_rtp_time_(0),
+ sequence_number_(0) {
+ RTC_CHECK_GT(fps_, 0);
+}
+
+// Generates a new frame for this stream. If called too soon after the
+// previous frame, no frame will be generated. The frame is split into
+// packets.
+int64_t RtpStream::GenerateFrame(int64_t time_now_us,
+ std::vector<PacketInfo>* packets) {
+ if (time_now_us < next_rtp_time_) {
+ return next_rtp_time_;
+ }
+ RTC_CHECK(packets != NULL);
+ size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
+ size_t n_packets =
+ std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
+ size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
+ for (size_t i = 0; i < n_packets; ++i) {
+ PacketInfo packet(-1, sequence_number_++);
+ packet.send_time_ms = (time_now_us + kSendSideOffsetUs) / 1000;
+ packet.payload_size = payload_size;
+ packet.probe_cluster_id = PacketInfo::kNotAProbe;
+ packets->push_back(packet);
+ }
+ next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
+ return next_rtp_time_;
+}
+
+// The send-side time when the next frame can be generated.
+int64_t RtpStream::next_rtp_time() const {
+ return next_rtp_time_;
+}
+
+void RtpStream::set_bitrate_bps(int bitrate_bps) {
+ ASSERT_GE(bitrate_bps, 0);
+ bitrate_bps_ = bitrate_bps;
+}
+
+int RtpStream::bitrate_bps() const {
+ return bitrate_bps_;
+}
+
+bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
+ const std::unique_ptr<RtpStream>& rhs) {
+ return lhs->next_rtp_time_ < rhs->next_rtp_time_;
+}
+
+StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
+ : capacity_(capacity), prev_arrival_time_us_(time_now) {}
+
+// Add a new stream.
+void StreamGenerator::AddStream(RtpStream* stream) {
+ streams_.push_back(std::unique_ptr<RtpStream>(stream));
+}
+
+// Set the link capacity.
+void StreamGenerator::set_capacity_bps(int capacity_bps) {
+ ASSERT_GT(capacity_bps, 0);
+ capacity_ = capacity_bps;
+}
+
+// Divides |bitrate_bps| among all streams. The allocated bitrate per stream
+// is decided by the current allocation ratios.
+void StreamGenerator::SetBitrateBps(int bitrate_bps) {
+ ASSERT_GE(streams_.size(), 0u);
+ int total_bitrate_before = 0;
+ for (const auto& stream : streams_) {
+ total_bitrate_before += stream->bitrate_bps();
+ }
+ int64_t bitrate_before = 0;
+ int total_bitrate_after = 0;
+ for (const auto& stream : streams_) {
+ bitrate_before += stream->bitrate_bps();
+ int64_t bitrate_after =
+ (bitrate_before * bitrate_bps + total_bitrate_before / 2) /
+ total_bitrate_before;
+ stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
+ total_bitrate_after += stream->bitrate_bps();
+ }
+ ASSERT_EQ(bitrate_before, total_bitrate_before);
+ EXPECT_EQ(total_bitrate_after, bitrate_bps);
+}
+
+// TODO(holmer): Break out the channel simulation part from this class to make
+// it possible to simulate different types of channels.
+int64_t StreamGenerator::GenerateFrame(std::vector<PacketInfo>* packets,
+ int64_t time_now_us) {
+ RTC_CHECK(packets != NULL);
+ RTC_CHECK(packets->empty());
+ RTC_CHECK_GT(capacity_, 0);
+ auto it =
+ std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+ (*it)->GenerateFrame(time_now_us, packets);
+ int i = 0;
+ for (PacketInfo& packet : *packets) {
+ int capacity_bpus = capacity_ / 1000;
+ int64_t required_network_time_us =
+ (8 * 1000 * packet.payload_size + capacity_bpus / 2) / capacity_bpus;
+ prev_arrival_time_us_ =
+ std::max(time_now_us + required_network_time_us,
+ prev_arrival_time_us_ + required_network_time_us);
+ packet.arrival_time_ms = prev_arrival_time_us_ / 1000;
+ ++i;
+ }
+ it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+ return std::max((*it)->next_rtp_time(), time_now_us);
+}
+} // namespace test
+
+DelayBasedBweTest::DelayBasedBweTest()
+ : clock_(100000000),
+ bitrate_observer_(new test::TestBitrateObserver),
+ bitrate_estimator_(new DelayBasedBwe(bitrate_observer_.get(), &clock_)),
+ stream_generator_(
+ new test::StreamGenerator(1e6, // Capacity.
+ clock_.TimeInMicroseconds())),
+ arrival_time_offset_ms_(0) {}
+
+DelayBasedBweTest::~DelayBasedBweTest() {}
+
+void DelayBasedBweTest::AddDefaultStream() {
+ stream_generator_->AddStream(new test::RtpStream(30, 3e5));
+}
+
+const uint32_t DelayBasedBweTest::kDefaultSsrc = 0;
+
+void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
+ int64_t send_time_ms,
+ uint16_t sequence_number,
+ size_t payload_size) {
+ IncomingFeedback(arrival_time_ms, send_time_ms, sequence_number, payload_size,
+ 0);
+}
+
+void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
+ int64_t send_time_ms,
+ uint16_t sequence_number,
+ size_t payload_size,
+ int probe_cluster_id) {
+ RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
+ PacketInfo packet(arrival_time_ms + arrival_time_offset_ms_, send_time_ms,
+ sequence_number, payload_size, probe_cluster_id);
+ std::vector<PacketInfo> packets;
+ packets.push_back(packet);
+ bitrate_estimator_->IncomingPacketFeedbackVector(packets);
+}
+
+// Generates a frame of packets belonging to a stream at a given bitrate and
+// with a given ssrc. The stream is pushed through a very simple simulated
+// network, and is then given to the receive-side bandwidth estimator.
+// Returns true if an over-use was seen, false otherwise.
+// The StreamGenerator::updated() should be used to check for any changes in
+// target bitrate after the call to this function.
+bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
+ uint32_t bitrate_bps) {
+ stream_generator_->SetBitrateBps(bitrate_bps);
+ std::vector<PacketInfo> packets;
+ int64_t next_time_us =
+ stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
+ if (packets.empty())
+ return false;
+
+ bool overuse = false;
+ bitrate_observer_->Reset();
+ clock_.AdvanceTimeMicroseconds(1000 * packets.back().arrival_time_ms -
+ clock_.TimeInMicroseconds());
+ for (auto& packet : packets) {
+ RTC_CHECK_GE(packet.arrival_time_ms + arrival_time_offset_ms_, 0);
+ packet.arrival_time_ms += arrival_time_offset_ms_;
+ }
+ bitrate_estimator_->IncomingPacketFeedbackVector(packets);
+
+ if (bitrate_observer_->updated()) {
+ if (bitrate_observer_->latest_bitrate() < bitrate_bps)
+ overuse = true;
+ }
+
+ clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
+ return overuse;
+}
+
+// Run the bandwidth estimator with a stream of |number_of_frames| frames, or
+// until it reaches |target_bitrate|.
+// Can for instance be used to run the estimator for some time to get it
+// into a steady state.
+uint32_t DelayBasedBweTest::SteadyStateRun(uint32_t ssrc,
+ int max_number_of_frames,
+ uint32_t start_bitrate,
+ uint32_t min_bitrate,
+ uint32_t max_bitrate,
+ uint32_t target_bitrate) {
+ uint32_t bitrate_bps = start_bitrate;
+ bool bitrate_update_seen = false;
+ // Produce |number_of_frames| frames and give them to the estimator.
+ for (int i = 0; i < max_number_of_frames; ++i) {
+ bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
+ if (overuse) {
+ EXPECT_LT(bitrate_observer_->latest_bitrate(), max_bitrate);
+ EXPECT_GT(bitrate_observer_->latest_bitrate(), min_bitrate);
+ bitrate_bps = bitrate_observer_->latest_bitrate();
+ bitrate_update_seen = true;
+ } else if (bitrate_observer_->updated()) {
+ bitrate_bps = bitrate_observer_->latest_bitrate();
+ bitrate_observer_->Reset();
+ }
+ if (bitrate_update_seen && bitrate_bps > target_bitrate) {
+ break;
+ }
+ }
+ EXPECT_TRUE(bitrate_update_seen);
+ return bitrate_bps;
+}
+
+void DelayBasedBweTest::InitialBehaviorTestHelper(
+ uint32_t expected_converge_bitrate) {
+ const int kFramerate = 50; // 50 fps to avoid rounding errors.
+ const int kFrameIntervalMs = 1000 / kFramerate;
+ uint32_t bitrate_bps = 0;
+ int64_t send_time_ms = 0;
+ uint16_t sequence_number = 0;
+ std::vector<uint32_t> ssrcs;
+ EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
+ EXPECT_EQ(0u, ssrcs.size());
+ clock_.AdvanceTimeMilliseconds(1000);
+ bitrate_estimator_->Process();
+ EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
+ EXPECT_FALSE(bitrate_observer_->updated());
+ bitrate_observer_->Reset();
+ clock_.AdvanceTimeMilliseconds(1000);
+ // Inserting packets for 5 seconds to get a valid estimate.
+ for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+ if (i == kNumInitialPackets) {
+ bitrate_estimator_->Process();
+ EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
+ EXPECT_EQ(0u, ssrcs.size());
+ EXPECT_FALSE(bitrate_observer_->updated());
+ bitrate_observer_->Reset();
+ }
+
+ IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
+ sequence_number++, kMtu);
+ clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
+ send_time_ms += kFrameIntervalMs;
+ }
+ bitrate_estimator_->Process();
+ EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
+ ASSERT_EQ(1u, ssrcs.size());
+ EXPECT_EQ(kDefaultSsrc, ssrcs.front());
+ EXPECT_NEAR(expected_converge_bitrate, bitrate_bps, kAcceptedBitrateErrorBps);
+ EXPECT_TRUE(bitrate_observer_->updated());
+ bitrate_observer_->Reset();
+ EXPECT_EQ(bitrate_observer_->latest_bitrate(), bitrate_bps);
+ bitrate_estimator_->RemoveStream(kDefaultSsrc);
+ EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
+ ASSERT_EQ(0u, ssrcs.size());
+ EXPECT_EQ(0u, bitrate_bps);
+}
+
+void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
+ uint32_t expected_bitrate_bps) {
+ const int kFramerate = 50; // 50 fps to avoid rounding errors.
+ const int kFrameIntervalMs = 1000 / kFramerate;
+ int64_t send_time_ms = 0;
+ uint16_t sequence_number = 0;
+ // Inserting packets for five seconds to get a valid estimate.
+ for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+ // TODO(sprang): Remove this hack once the single stream estimator is gone,
+ // as it doesn't do anything in Process().
+ if (i == kNumInitialPackets) {
+ // Process after we have enough frames to get a valid input rate estimate.
+ bitrate_estimator_->Process();
+ EXPECT_FALSE(bitrate_observer_->updated()); // No valid estimate.
+ }
+
+ IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
+ sequence_number++, kMtu);
+ clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+ send_time_ms += kFrameIntervalMs;
+ }
+ bitrate_estimator_->Process();
+ EXPECT_TRUE(bitrate_observer_->updated());
+ EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_->latest_bitrate(),
+ kAcceptedBitrateErrorBps);
+ for (int i = 0; i < 10; ++i) {
+ clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+ send_time_ms += 2 * kFrameIntervalMs;
+ IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
+ sequence_number + 2, 1000);
+ IncomingFeedback(clock_.TimeInMilliseconds(),
+ send_time_ms - kFrameIntervalMs, sequence_number + 1,
+ 1000);
+ sequence_number += 2;
+ }
+ bitrate_estimator_->Process();
+ EXPECT_TRUE(bitrate_observer_->updated());
+ EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_->latest_bitrate(),
+ kAcceptedBitrateErrorBps);
+}
+
+// Make sure we initially increase the bitrate as expected.
+void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
+ int expected_iterations) {
+ // This threshold corresponds approximately to increasing linearly with
+ // bitrate(i) = 1.04 * bitrate(i-1) + 1000
+ // until bitrate(i) > 500000, with bitrate(1) ~= 30000.
+ uint32_t bitrate_bps = 30000;
+ int iterations = 0;
+ AddDefaultStream();
+ // Feed the estimator with a stream of packets and verify that it reaches
+ // 500 kbps at the expected time.
+ while (bitrate_bps < 5e5) {
+ bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+ if (overuse) {
+ EXPECT_GT(bitrate_observer_->latest_bitrate(), bitrate_bps);
+ bitrate_bps = bitrate_observer_->latest_bitrate();
+ bitrate_observer_->Reset();
+ } else if (bitrate_observer_->updated()) {
+ bitrate_bps = bitrate_observer_->latest_bitrate();
+ bitrate_observer_->Reset();
+ }
+ ++iterations;
+ ASSERT_LE(iterations, expected_iterations);
+ }
+ ASSERT_EQ(expected_iterations, iterations);
+}
+
+void DelayBasedBweTest::CapacityDropTestHelper(
+ int number_of_streams,
+ bool wrap_time_stamp,
+ uint32_t expected_bitrate_drop_delta,
+ int64_t receiver_clock_offset_change_ms) {
+ const int kFramerate = 30;
+ const int kStartBitrate = 900e3;
+ const int kMinExpectedBitrate = 800e3;
+ const int kMaxExpectedBitrate = 1100e3;
+ const uint32_t kInitialCapacityBps = 1000e3;
+ const uint32_t kReducedCapacityBps = 500e3;
+
+ int steady_state_time = 0;
+ if (number_of_streams <= 1) {
+ steady_state_time = 10;
+ AddDefaultStream();
+ } else {
+ steady_state_time = 10 * number_of_streams;
+ int bitrate_sum = 0;
+ int kBitrateDenom = number_of_streams * (number_of_streams - 1);
+ for (int i = 0; i < number_of_streams; i++) {
+ // First stream gets half available bitrate, while the rest share the
+ // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
+ int bitrate = kStartBitrate / 2;
+ if (i > 0) {
+ bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
+ }
+ stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate));
+ bitrate_sum += bitrate;
+ }
+ ASSERT_EQ(bitrate_sum, kStartBitrate);
+ }
+
+ // Run in steady state to make the estimator converge.
+ stream_generator_->set_capacity_bps(kInitialCapacityBps);
+ uint32_t bitrate_bps = SteadyStateRun(
+ kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
+ kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
+ EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 130000u);
+ bitrate_observer_->Reset();
+
+ // Add an offset to make sure the BWE can handle it.
+ arrival_time_offset_ms_ += receiver_clock_offset_change_ms;
+
+ // Reduce the capacity and verify the decrease time.
+ stream_generator_->set_capacity_bps(kReducedCapacityBps);
+ int64_t overuse_start_time = clock_.TimeInMilliseconds();
+ int64_t bitrate_drop_time = -1;
+ for (int i = 0; i < 100 * number_of_streams; ++i) {
+ GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+ if (bitrate_drop_time == -1 &&
+ bitrate_observer_->latest_bitrate() <= kReducedCapacityBps) {
+ bitrate_drop_time = clock_.TimeInMilliseconds();
+ }
+ if (bitrate_observer_->updated())
+ bitrate_bps = bitrate_observer_->latest_bitrate();
+ }
+
+ EXPECT_NEAR(expected_bitrate_drop_delta,
+ bitrate_drop_time - overuse_start_time, 33);
+}
+
+void DelayBasedBweTest::TestTimestampGroupingTestHelper() {
+ const int kFramerate = 50; // 50 fps to avoid rounding errors.
+ const int kFrameIntervalMs = 1000 / kFramerate;
+ int64_t send_time_ms = 0;
+ uint16_t sequence_number = 0;
+ // Initial set of frames to increase the bitrate. 6 seconds to have enough
+ // time for the first estimate to be generated and for Process() to be called.
+ for (int i = 0; i <= 6 * kFramerate; ++i) {
+ IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
+ sequence_number++, 1000);
+
+ bitrate_estimator_->Process();
+ clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+ send_time_ms += kFrameIntervalMs;
+ }
+ EXPECT_TRUE(bitrate_observer_->updated());
+ EXPECT_GE(bitrate_observer_->latest_bitrate(), 400000u);
+
+ // Insert batches of frames which were sent very close in time. Also simulate
+ // capacity over-use to see that we back off correctly.
+ const int kTimestampGroupLength = 15;
+ for (int i = 0; i < 100; ++i) {
+ for (int j = 0; j < kTimestampGroupLength; ++j) {
+ // Insert |kTimestampGroupLength| frames with just 1 timestamp ticks in
+ // between. Should be treated as part of the same group by the estimator.
+ IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
+ sequence_number++, 100);
+ clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
+ send_time_ms += 1;
+ }
+ // Increase time until next batch to simulate over-use.
+ clock_.AdvanceTimeMilliseconds(10);
+ send_time_ms += kFrameIntervalMs - kTimestampGroupLength;
+ bitrate_estimator_->Process();
+ }
+ EXPECT_TRUE(bitrate_observer_->updated());
+ // Should have reduced the estimate.
+ EXPECT_LT(bitrate_observer_->latest_bitrate(), 400000u);
+}
+
+void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
+ const int kFramerate = 100;
+ const int kFrameIntervalMs = 1000 / kFramerate;
+ int64_t send_time_ms = 0;
+ uint16_t sequence_number = 0;
+
+ for (size_t i = 0; i < 3000; ++i) {
+ IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
+ sequence_number++, 1000);
+ clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+ send_time_ms += kFrameIntervalMs;
+ bitrate_estimator_->Process();
+ }
+ uint32_t bitrate_before = 0;
+ std::vector<uint32_t> ssrcs;
+ bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);
+
+ clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
+ send_time_ms += silence_time_s * 1000;
+ bitrate_estimator_->Process();
+
+ for (size_t i = 0; i < 21; ++i) {
+ IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
+ sequence_number++, 1000);
+ clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+ send_time_ms += kFrameIntervalMs;
+ bitrate_estimator_->Process();
+ }
+ uint32_t bitrate_after = 0;
+ bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
+ EXPECT_LT(bitrate_after, bitrate_before);
+}
+} // namespace webrtc
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