diff --git a/webrtc/base/rate_statistics.cc b/webrtc/base/rate_statistics.cc index 6529aa1f7a..1fd63cc6d2 100644 --- a/webrtc/base/rate_statistics.cc +++ b/webrtc/base/rate_statistics.cc @@ -16,23 +16,26 @@ namespace webrtc { -RateStatistics::RateStatistics(uint32_t window_size_ms, float scale) - : num_buckets_(window_size_ms + 1), // N ms in (N+1) buckets. - buckets_(new size_t[num_buckets_]()), +RateStatistics::RateStatistics(int64_t window_size_ms, float scale) + : buckets_(new Bucket[window_size_ms]()), accumulated_count_(0), - oldest_time_(0), + num_samples_(0), + oldest_time_(-window_size_ms), oldest_index_(0), - scale_(scale) {} + scale_(scale), + max_window_size_ms_(window_size_ms), + current_window_size_ms_(max_window_size_ms_) {} RateStatistics::~RateStatistics() {} void RateStatistics::Reset() { accumulated_count_ = 0; - oldest_time_ = 0; + num_samples_ = 0; + oldest_time_ = -max_window_size_ms_; oldest_index_ = 0; - for (int i = 0; i < num_buckets_; i++) { - buckets_[i] = 0; - } + current_window_size_ms_ = max_window_size_ms_; + for (int64_t i = 0; i < max_window_size_ms_; i++) + buckets_[i] = Bucket(); } void RateStatistics::Update(size_t count, int64_t now_ms) { @@ -43,46 +46,74 @@ void RateStatistics::Update(size_t count, int64_t now_ms) { EraseOld(now_ms); - int now_offset = static_cast(now_ms - oldest_time_); - RTC_DCHECK_LT(now_offset, num_buckets_); - int index = oldest_index_ + now_offset; - if (index >= num_buckets_) { - index -= num_buckets_; - } - buckets_[index] += count; + // First ever sample, reset window to start now. + if (!IsInitialized()) + oldest_time_ = now_ms; + + uint32_t now_offset = static_cast(now_ms - oldest_time_); + RTC_DCHECK_LT(now_offset, max_window_size_ms_); + uint32_t index = oldest_index_ + now_offset; + if (index >= max_window_size_ms_) + index -= max_window_size_ms_; + buckets_[index].sum += count; + ++buckets_[index].samples; accumulated_count_ += count; + ++num_samples_; } -uint32_t RateStatistics::Rate(int64_t now_ms) { +rtc::Optional RateStatistics::Rate(int64_t now_ms) { EraseOld(now_ms); - float scale = scale_ / (now_ms - oldest_time_ + 1); - return static_cast(accumulated_count_ * scale + 0.5f); + + // If window is a single bucket or there is only one sample in a data set that + // has not grown to the full window size, treat this as rate unavailable. + int64_t active_window_size = now_ms - oldest_time_ + 1; + if (num_samples_ == 0 || active_window_size <= 1 || + (num_samples_ <= 1 && active_window_size < current_window_size_ms_)) { + return rtc::Optional(); + } + + float scale = scale_ / active_window_size; + return rtc::Optional( + static_cast(accumulated_count_ * scale + 0.5f)); } void RateStatistics::EraseOld(int64_t now_ms) { - int64_t new_oldest_time = now_ms - num_buckets_ + 1; - if (new_oldest_time <= oldest_time_) { - if (accumulated_count_ == 0) - oldest_time_ = now_ms; + if (!IsInitialized()) return; - } - while (oldest_time_ < new_oldest_time) { - size_t count_in_oldest_bucket = buckets_[oldest_index_]; - RTC_DCHECK_GE(accumulated_count_, count_in_oldest_bucket); - accumulated_count_ -= count_in_oldest_bucket; - buckets_[oldest_index_] = 0; - if (++oldest_index_ >= num_buckets_) { + + // New oldest time that is included in data set. + int64_t new_oldest_time = now_ms - current_window_size_ms_ + 1; + + // New oldest time is older than the current one, no need to cull data. + if (new_oldest_time <= oldest_time_) + return; + + // Loop over buckets and remove too old data points. + while (num_samples_ > 0 && oldest_time_ < new_oldest_time) { + const Bucket& oldest_bucket = buckets_[oldest_index_]; + RTC_DCHECK_GE(accumulated_count_, oldest_bucket.sum); + RTC_DCHECK_GE(num_samples_, oldest_bucket.samples); + accumulated_count_ -= oldest_bucket.sum; + num_samples_ -= oldest_bucket.samples; + buckets_[oldest_index_] = Bucket(); + if (++oldest_index_ >= max_window_size_ms_) oldest_index_ = 0; - } ++oldest_time_; - if (accumulated_count_ == 0) { - // This guarantees we go through all the buckets at most once, even if - // |new_oldest_time| is far greater than |oldest_time_|. - new_oldest_time = now_ms; - break; - } } oldest_time_ = new_oldest_time; } +bool RateStatistics::SetWindowSize(int64_t window_size_ms, int64_t now_ms) { + if (window_size_ms <= 0 || window_size_ms > max_window_size_ms_) + return false; + + current_window_size_ms_ = window_size_ms; + EraseOld(now_ms); + return true; +} + +bool RateStatistics::IsInitialized() { + return oldest_time_ != -max_window_size_ms_; +} + } // namespace webrtc diff --git a/webrtc/base/rate_statistics.h b/webrtc/base/rate_statistics.h index aea8d793fe..3e913cc1bb 100644 --- a/webrtc/base/rate_statistics.h +++ b/webrtc/base/rate_statistics.h @@ -13,41 +13,56 @@ #include +#include "webrtc/base/optional.h" #include "webrtc/typedefs.h" namespace webrtc { class RateStatistics { public: - // window_size = window size in ms for the rate estimation + // max_window_size_ms = Maximum window size in ms for the rate estimation. + // Initial window size is set to this, but may be changed + // to something lower by calling SetWindowSize(). // scale = coefficient to convert counts/ms to desired units, // ex: if counts represents bytes, use 8*1000 to go to bits/s - RateStatistics(uint32_t window_size_ms, float scale); + RateStatistics(int64_t max_window_size_ms, float scale); ~RateStatistics(); void Reset(); void Update(size_t count, int64_t now_ms); - uint32_t Rate(int64_t now_ms); + rtc::Optional Rate(int64_t now_ms); + bool SetWindowSize(int64_t window_size_ms, int64_t now_ms); private: void EraseOld(int64_t now_ms); + bool IsInitialized(); // Counters are kept in buckets (circular buffer), with one bucket // per millisecond. - const int num_buckets_; - std::unique_ptr buckets_; + struct Bucket { + size_t sum; // Sum of all samples in this bucket. + size_t samples; // Number of samples in this bucket. + }; + std::unique_ptr buckets_; // Total count recorded in buckets. size_t accumulated_count_; + // The total number of samples in the buckets. + size_t num_samples_; + // Oldest time recorded in buckets. int64_t oldest_time_; // Bucket index of oldest counter recorded in buckets. - int oldest_index_; + uint32_t oldest_index_; // To convert counts/ms to desired units const float scale_; + + // The window sizes, in ms, over which the rate is calculated. + const int64_t max_window_size_ms_; + int64_t current_window_size_ms_; }; } // namespace webrtc diff --git a/webrtc/base/rate_statistics_unittest.cc b/webrtc/base/rate_statistics_unittest.cc index 9702da0699..f6ce528ce4 100644 --- a/webrtc/base/rate_statistics_unittest.cc +++ b/webrtc/base/rate_statistics_unittest.cc @@ -27,77 +27,97 @@ class RateStatisticsTest : public ::testing::Test { TEST_F(RateStatisticsTest, TestStrictMode) { int64_t now_ms = 0; - // Should be initialized to 0. - EXPECT_EQ(0u, stats_.Rate(now_ms)); - stats_.Update(1500, now_ms); + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + + const uint32_t kPacketSize = 1500u; + const uint32_t kExpectedRateBps = kPacketSize * 1000 * 8; + + // Single data point is not enough for valid estimate. + stats_.Update(kPacketSize, now_ms++); + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + // Expecting 1200 kbps since the window is initially kept small and grows as // we have more data. - EXPECT_EQ(12000000u, stats_.Rate(now_ms)); + stats_.Update(kPacketSize, now_ms); + EXPECT_EQ(kExpectedRateBps, *stats_.Rate(now_ms)); + stats_.Reset(); // Expecting 0 after init. - EXPECT_EQ(0u, stats_.Rate(now_ms)); + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + + const int kInterval = 10; for (int i = 0; i < 100000; ++i) { - if (now_ms % 10 == 0) { - stats_.Update(1500, now_ms); - } + if (i % kInterval == 0) + stats_.Update(kPacketSize, now_ms); + // Approximately 1200 kbps expected. Not exact since when packets // are removed we will jump 10 ms to the next packet. - if (now_ms > 0 && now_ms % kWindowMs == 0) { - EXPECT_NEAR(1200000u, stats_.Rate(now_ms), 22000u); + if (i > kInterval) { + rtc::Optional rate = stats_.Rate(now_ms); + EXPECT_TRUE(static_cast(rate)); + uint32_t samples = i / kInterval + 1; + uint64_t total_bits = samples * kPacketSize * 8; + uint32_t rate_bps = static_cast((1000 * total_bits) / (i + 1)); + EXPECT_NEAR(rate_bps, *rate, 22000u); } now_ms += 1; } now_ms += kWindowMs; // The window is 2 seconds. If nothing has been received for that time // the estimate should be 0. - EXPECT_EQ(0u, stats_.Rate(now_ms)); + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); } TEST_F(RateStatisticsTest, IncreasingThenDecreasingBitrate) { int64_t now_ms = 0; stats_.Reset(); // Expecting 0 after init. - uint32_t bitrate = stats_.Rate(now_ms); - EXPECT_EQ(0u, bitrate); + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + + stats_.Update(1000, ++now_ms); const uint32_t kExpectedBitrate = 8000000; // 1000 bytes per millisecond until plateau is reached. int prev_error = kExpectedBitrate; + rtc::Optional bitrate; while (++now_ms < 10000) { stats_.Update(1000, now_ms); bitrate = stats_.Rate(now_ms); - int error = kExpectedBitrate - bitrate; + EXPECT_TRUE(static_cast(bitrate)); + int error = kExpectedBitrate - *bitrate; error = std::abs(error); // Expect the estimation error to decrease as the window is extended. EXPECT_LE(error, prev_error + 1); prev_error = error; } // Window filled, expect to be close to 8000000. - EXPECT_EQ(kExpectedBitrate, bitrate); + EXPECT_EQ(kExpectedBitrate, *bitrate); // 1000 bytes per millisecond until 10-second mark, 8000 kbps expected. while (++now_ms < 10000) { stats_.Update(1000, now_ms); bitrate = stats_.Rate(now_ms); - EXPECT_EQ(kExpectedBitrate, bitrate); + EXPECT_EQ(kExpectedBitrate, *bitrate); } + // Zero bytes per millisecond until 0 is reached. while (++now_ms < 20000) { stats_.Update(0, now_ms); - uint32_t new_bitrate = stats_.Rate(now_ms); - if (new_bitrate != bitrate) { + rtc::Optional new_bitrate = stats_.Rate(now_ms); + if (static_cast(new_bitrate) && *new_bitrate != *bitrate) { // New bitrate must be lower than previous one. - EXPECT_LT(new_bitrate, bitrate); + EXPECT_LT(*new_bitrate, *bitrate); } else { // 0 kbps expected. - EXPECT_EQ(0u, bitrate); + EXPECT_EQ(0u, *new_bitrate); break; } bitrate = new_bitrate; } + // Zero bytes per millisecond until 20-second mark, 0 kbps expected. while (++now_ms < 20000) { stats_.Update(0, now_ms); - EXPECT_EQ(0u, stats_.Rate(now_ms)); + EXPECT_EQ(0u, *stats_.Rate(now_ms)); } } @@ -105,28 +125,156 @@ TEST_F(RateStatisticsTest, ResetAfterSilence) { int64_t now_ms = 0; stats_.Reset(); // Expecting 0 after init. - uint32_t bitrate = stats_.Rate(now_ms); - EXPECT_EQ(0u, bitrate); + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + const uint32_t kExpectedBitrate = 8000000; // 1000 bytes per millisecond until the window has been filled. int prev_error = kExpectedBitrate; + rtc::Optional bitrate; while (++now_ms < 10000) { stats_.Update(1000, now_ms); bitrate = stats_.Rate(now_ms); - int error = kExpectedBitrate - bitrate; - error = std::abs(error); - // Expect the estimation error to decrease as the window is extended. - EXPECT_LE(error, prev_error + 1); - prev_error = error; + if (bitrate) { + int error = kExpectedBitrate - *bitrate; + error = std::abs(error); + // Expect the estimation error to decrease as the window is extended. + EXPECT_LE(error, prev_error + 1); + prev_error = error; + } } // Window filled, expect to be close to 8000000. - EXPECT_EQ(kExpectedBitrate, bitrate); + EXPECT_EQ(kExpectedBitrate, *bitrate); now_ms += kWindowMs + 1; - EXPECT_EQ(0u, stats_.Rate(now_ms)); + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); stats_.Update(1000, now_ms); - // We expect one sample of 1000 bytes, and that the bitrate is measured over - // 1 ms, i.e., 8 * 1000 / 0.001 = 8000000. - EXPECT_EQ(kExpectedBitrate, stats_.Rate(now_ms)); + ++now_ms; + stats_.Update(1000, now_ms); + // We expect two samples of 1000 bytes, and that the bitrate is measured over + // 500 ms, i.e. 2 * 8 * 1000 / 0.500 = 32000. + EXPECT_EQ(32000u, *stats_.Rate(now_ms)); + + // Reset, add the same samples again. + stats_.Reset(); + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + stats_.Update(1000, now_ms); + ++now_ms; + stats_.Update(1000, now_ms); + // We expect two samples of 1000 bytes, and that the bitrate is measured over + // 2 ms (window size has been reset) i.e. 2 * 8 * 1000 / 0.002 = 8000000. + EXPECT_EQ(kExpectedBitrate, *stats_.Rate(now_ms)); +} + +TEST_F(RateStatisticsTest, HandlesChangingWindowSize) { + int64_t now_ms = 0; + stats_.Reset(); + + // Sanity test window size. + EXPECT_TRUE(stats_.SetWindowSize(kWindowMs, now_ms)); + EXPECT_FALSE(stats_.SetWindowSize(kWindowMs + 1, now_ms)); + EXPECT_FALSE(stats_.SetWindowSize(0, now_ms)); + EXPECT_TRUE(stats_.SetWindowSize(1, now_ms)); + EXPECT_TRUE(stats_.SetWindowSize(kWindowMs, now_ms)); + + // Fill the buffer at a rate of 1 byte / millisecond (8 kbps). + const int kBatchSize = 10; + for (int i = 0; i <= kWindowMs; i += kBatchSize) + stats_.Update(kBatchSize, now_ms += kBatchSize); + EXPECT_EQ(static_cast(8000), *stats_.Rate(now_ms)); + + // Halve the window size, rate should stay the same. + EXPECT_TRUE(stats_.SetWindowSize(kWindowMs / 2, now_ms)); + EXPECT_EQ(static_cast(8000), *stats_.Rate(now_ms)); + + // Double the window size again, rate should stay the same. (As the window + // won't actually expand until new bit and bobs fall into it. + EXPECT_TRUE(stats_.SetWindowSize(kWindowMs, now_ms)); + EXPECT_EQ(static_cast(8000), *stats_.Rate(now_ms)); + + // Fill the now empty half with bits it twice the rate. + for (int i = 0; i < kWindowMs / 2; i += kBatchSize) + stats_.Update(kBatchSize * 2, now_ms += kBatchSize); + + // Rate should have increase be 50%. + EXPECT_EQ(static_cast((8000 * 3) / 2), *stats_.Rate(now_ms)); +} + +TEST_F(RateStatisticsTest, RespectsWindowSizeEdges) { + int64_t now_ms = 0; + stats_.Reset(); + // Expecting 0 after init. + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + + // One byte per ms, using one big sample. + stats_.Update(kWindowMs, now_ms); + now_ms += kWindowMs - 2; + // Shouldn't work! (Only one sample, not full window size.) + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + + // Window size should be full, and the single data point should be accepted. + ++now_ms; + rtc::Optional bitrate = stats_.Rate(now_ms); + EXPECT_TRUE(static_cast(bitrate)); + EXPECT_EQ(1000 * 8u, *bitrate); + + // Add another, now we have twice the bitrate. + stats_.Update(kWindowMs, now_ms); + bitrate = stats_.Rate(now_ms); + EXPECT_TRUE(static_cast(bitrate)); + EXPECT_EQ(2 * 1000 * 8u, *bitrate); + + // Now that first sample should drop out... + now_ms += 1; + bitrate = stats_.Rate(now_ms); + EXPECT_TRUE(static_cast(bitrate)); + EXPECT_EQ(1000 * 8u, *bitrate); +} + +TEST_F(RateStatisticsTest, HandlesZeroCounts) { + int64_t now_ms = 0; + stats_.Reset(); + // Expecting 0 after init. + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + + stats_.Update(kWindowMs, now_ms); + now_ms += kWindowMs - 1; + stats_.Update(0, now_ms); + rtc::Optional bitrate = stats_.Rate(now_ms); + EXPECT_TRUE(static_cast(bitrate)); + EXPECT_EQ(1000 * 8u, *bitrate); + + // Move window along so first data point falls out. + ++now_ms; + bitrate = stats_.Rate(now_ms); + EXPECT_TRUE(static_cast(bitrate)); + EXPECT_EQ(0u, *bitrate); + + // Move window so last data point falls out. + now_ms += kWindowMs; + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); +} + +TEST_F(RateStatisticsTest, HandlesQuietPeriods) { + int64_t now_ms = 0; + stats_.Reset(); + // Expecting 0 after init. + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + + stats_.Update(0, now_ms); + now_ms += kWindowMs - 1; + rtc::Optional bitrate = stats_.Rate(now_ms); + EXPECT_TRUE(static_cast(bitrate)); + EXPECT_EQ(0u, *bitrate); + + // Move window along so first data point falls out. + ++now_ms; + EXPECT_FALSE(static_cast(stats_.Rate(now_ms))); + + // Move window a long way out. + now_ms += 2 * kWindowMs; + stats_.Update(0, now_ms); + bitrate = stats_.Rate(now_ms); + EXPECT_TRUE(static_cast(bitrate)); + EXPECT_EQ(0u, *bitrate); } } // namespace diff --git a/webrtc/common_video/bitrate_adjuster.cc b/webrtc/common_video/bitrate_adjuster.cc index ada6c5db4b..9c5c077a83 100644 --- a/webrtc/common_video/bitrate_adjuster.cc +++ b/webrtc/common_video/bitrate_adjuster.cc @@ -70,7 +70,7 @@ uint32_t BitrateAdjuster::GetAdjustedBitrateBps() const { return adjusted_bitrate_bps_; } -uint32_t BitrateAdjuster::GetEstimatedBitrateBps() { +rtc::Optional BitrateAdjuster::GetEstimatedBitrateBps() { rtc::CritScope cs(&crit_); return bitrate_tracker_.Rate(clock_->TimeInMilliseconds()); } @@ -121,8 +121,9 @@ void BitrateAdjuster::UpdateBitrate(uint32_t current_time_ms) { frames_since_last_update_ < kBitrateUpdateFrameInterval) { return; } - float estimated_bitrate_bps = bitrate_tracker_.Rate(current_time_ms); float target_bitrate_bps = target_bitrate_bps_; + float estimated_bitrate_bps = + bitrate_tracker_.Rate(current_time_ms).value_or(target_bitrate_bps); float error = target_bitrate_bps - estimated_bitrate_bps; // Adjust if we've overshot by any amount or if we've undershot too much. diff --git a/webrtc/common_video/bitrate_adjuster_unittest.cc b/webrtc/common_video/bitrate_adjuster_unittest.cc index 23b278731e..d0517e4fcb 100644 --- a/webrtc/common_video/bitrate_adjuster_unittest.cc +++ b/webrtc/common_video/bitrate_adjuster_unittest.cc @@ -48,7 +48,8 @@ class BitrateAdjusterTest : public ::testing::Test { // target bitrate within clamp. uint32_t target_bitrate_bps = adjuster_.GetTargetBitrateBps(); uint32_t adjusted_bitrate_bps = adjuster_.GetAdjustedBitrateBps(); - uint32_t estimated_bitrate_bps = adjuster_.GetEstimatedBitrateBps(); + uint32_t estimated_bitrate_bps = + adjuster_.GetEstimatedBitrateBps().value_or(target_bitrate_bps); uint32_t adjusted_lower_bound_bps = GetTargetBitrateBpsPct(kMinAdjustedBitratePct); uint32_t adjusted_upper_bound_bps = diff --git a/webrtc/common_video/include/bitrate_adjuster.h b/webrtc/common_video/include/bitrate_adjuster.h index 1f2474f88b..5fd1e3851b 100644 --- a/webrtc/common_video/include/bitrate_adjuster.h +++ b/webrtc/common_video/include/bitrate_adjuster.h @@ -47,7 +47,7 @@ class BitrateAdjuster { uint32_t GetAdjustedBitrateBps() const; // Returns what we think the current bitrate is. - uint32_t GetEstimatedBitrateBps(); + rtc::Optional GetEstimatedBitrateBps(); // This should be called after each frame is encoded. The timestamp at which // it is called is used to estimate the output bitrate of the encoder. diff --git a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc index c6d0c88280..295a2f4ddf 100644 --- a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc +++ b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc @@ -37,7 +37,7 @@ AimdRateControl::AimdRateControl() rate_control_state_(kRcHold), rate_control_region_(kRcMaxUnknown), time_last_bitrate_change_(-1), - current_input_(kBwNormal, 0, 1.0), + current_input_(kBwNormal, rtc::Optional(), 1.0), updated_(false), time_first_incoming_estimate_(-1), bitrate_is_initialized_(false), @@ -87,8 +87,9 @@ uint32_t AimdRateControl::LatestEstimate() const { } uint32_t AimdRateControl::UpdateBandwidthEstimate(int64_t now_ms) { - current_bitrate_bps_ = ChangeBitrate(current_bitrate_bps_, - current_input_.incoming_bitrate, now_ms); + current_bitrate_bps_ = ChangeBitrate( + current_bitrate_bps_, + current_input_.incoming_bitrate.value_or(current_bitrate_bps_), now_ms); if (now_ms - time_of_last_log_ > kLogIntervalMs) { time_of_last_log_ = now_ms; } @@ -100,7 +101,7 @@ void AimdRateControl::SetRtt(int64_t rtt) { } void AimdRateControl::Update(const RateControlInput* input, int64_t now_ms) { - assert(input); + RTC_CHECK(input); // Set the initial bit rate value to what we're receiving the first half // second. @@ -108,12 +109,11 @@ void AimdRateControl::Update(const RateControlInput* input, int64_t now_ms) { const int64_t kInitializationTimeMs = 5000; RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTimeMs); if (time_first_incoming_estimate_ < 0) { - if (input->incoming_bitrate > 0) { + if (input->incoming_bitrate) time_first_incoming_estimate_ = now_ms; - } } else if (now_ms - time_first_incoming_estimate_ > kInitializationTimeMs && - input->incoming_bitrate > 0) { - current_bitrate_bps_ = input->incoming_bitrate; + input->incoming_bitrate) { + current_bitrate_bps_ = *input->incoming_bitrate; bitrate_is_initialized_ = true; } } diff --git a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h index 93ae2190d6..3ac80752b4 100644 --- a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h +++ b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h @@ -67,7 +67,6 @@ class AimdRateControl { uint32_t min_configured_bitrate_bps_; uint32_t max_configured_bitrate_bps_; uint32_t current_bitrate_bps_; - uint32_t max_hold_rate_bps_; float avg_max_bitrate_kbps_; float var_max_bitrate_kbps_; RateControlState rate_control_state_; diff --git a/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h b/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h index 3fb7e29e5b..9aa82cff97 100644 --- a/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h +++ b/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h @@ -11,6 +11,7 @@ #ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_ #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_ +#include "webrtc/base/optional.h" #include "webrtc/typedefs.h" #define BWE_MAX(a, b) ((a) > (b) ? (a) : (b)) @@ -32,14 +33,14 @@ enum RateControlRegion { kRcNearMax, kRcAboveMax, kRcMaxUnknown }; struct RateControlInput { RateControlInput(BandwidthUsage bw_state, - uint32_t incoming_bitrate, + const rtc::Optional& incoming_bitrate, double noise_var) : bw_state(bw_state), incoming_bitrate(incoming_bitrate), noise_var(noise_var) {} BandwidthUsage bw_state; - uint32_t incoming_bitrate; + rtc::Optional incoming_bitrate; double noise_var; }; } // namespace webrtc diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc index 8ad60aea65..5975c5f3b8 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc @@ -85,6 +85,7 @@ bool RemoteBitrateEstimatorAbsSendTime::IsWithinClusterBounds( estimator_(), detector_(OverUseDetectorOptions()), incoming_bitrate_(kBitrateWindowMs, 8000), + incoming_bitrate_initialized_(false), total_probes_received_(0), first_packet_time_ms_(-1), last_update_ms_(-1), @@ -243,6 +244,18 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo( int64_t now_ms = arrival_time_ms; // TODO(holmer): SSRCs are only needed for REMB, should be broken out from // here. + + // Check if incoming bitrate estimate is valid, and if it needs to be reset. + rtc::Optional incoming_bitrate = incoming_bitrate_.Rate(now_ms); + if (incoming_bitrate) { + incoming_bitrate_initialized_ = true; + } else if (incoming_bitrate_initialized_) { + // Incoming bitrate had a previous valid value, but now not enough data + // point are left within the current window. Reset incoming bitrate + // estimator so that the window size will only contain new data points. + incoming_bitrate_.Reset(); + incoming_bitrate_initialized_ = false; + } incoming_bitrate_.Update(payload_size, now_ms); if (first_packet_time_ms_ == -1) @@ -303,10 +316,12 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo( if (last_update_ms_ == -1 || now_ms - last_update_ms_ > remote_rate_.GetFeedbackInterval()) { update_estimate = true; - } else if (detector_.State() == kBwOverusing && - remote_rate_.TimeToReduceFurther( - now_ms, incoming_bitrate_.Rate(now_ms))) { - update_estimate = true; + } else if (detector_.State() == kBwOverusing) { + rtc::Optional incoming_rate = incoming_bitrate_.Rate(now_ms); + if (incoming_rate && + remote_rate_.TimeToReduceFurther(now_ms, *incoming_rate)) { + update_estimate = true; + } } } diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h index a6119091ce..e84c749d1b 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h @@ -124,6 +124,7 @@ class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator { std::unique_ptr estimator_; OveruseDetector detector_; RateStatistics incoming_bitrate_; + bool incoming_bitrate_initialized_; std::vector recent_propagation_delta_ms_; std::vector recent_update_time_ms_; std::list probes_; diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc index a57fcb5114..6f8696a440 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc @@ -35,7 +35,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, RateIncreaseReordering) { } TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, RateIncreaseRtpTimestamps) { - RateIncreaseRtpTimestampsTestHelper(1229); + RateIncreaseRtpTimestampsTestHelper(1237); } TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropOneStream) { diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc index f38ef78306..d391f03262 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc @@ -44,16 +44,17 @@ struct RemoteBitrateEstimatorSingleStream::Detector { OveruseDetector detector; }; - RemoteBitrateEstimatorSingleStream::RemoteBitrateEstimatorSingleStream( - RemoteBitrateObserver* observer, - Clock* clock) - : clock_(clock), - incoming_bitrate_(kBitrateWindowMs, 8000), - remote_rate_(new AimdRateControl()), - observer_(observer), - crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - last_process_time_(-1), - process_interval_ms_(kProcessIntervalMs) { +RemoteBitrateEstimatorSingleStream::RemoteBitrateEstimatorSingleStream( + RemoteBitrateObserver* observer, + Clock* clock) + : clock_(clock), + incoming_bitrate_(kBitrateWindowMs, 8000), + last_valid_incoming_bitrate_(0), + remote_rate_(new AimdRateControl()), + observer_(observer), + crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + last_process_time_(-1), + process_interval_ms_(kProcessIntervalMs) { assert(observer_); LOG(LS_INFO) << "RemoteBitrateEstimatorSingleStream: Instantiating."; } @@ -90,7 +91,20 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket(int64_t arrival_time_ms, } Detector* estimator = it->second; estimator->last_packet_time_ms = now_ms; + + // Check if incoming bitrate estimate is valid, and if it needs to be reset. + rtc::Optional incoming_bitrate = incoming_bitrate_.Rate(now_ms); + if (incoming_bitrate) { + last_valid_incoming_bitrate_ = *incoming_bitrate; + } else if (last_valid_incoming_bitrate_ > 0) { + // Incoming bitrate had a previous valid value, but now not enough data + // point are left within the current window. Reset incoming bitrate + // estimator so that the window size will only contain new data points. + incoming_bitrate_.Reset(); + last_valid_incoming_bitrate_ = 0; + } incoming_bitrate_.Update(payload_size, now_ms); + const BandwidthUsage prior_state = estimator->detector.State(); uint32_t timestamp_delta = 0; int64_t time_delta = 0; @@ -106,9 +120,11 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket(int64_t arrival_time_ms, estimator->estimator.num_of_deltas(), now_ms); } if (estimator->detector.State() == kBwOverusing) { - uint32_t incoming_bitrate_bps = incoming_bitrate_.Rate(now_ms); - if (prior_state != kBwOverusing || - remote_rate_->TimeToReduceFurther(now_ms, incoming_bitrate_bps)) { + rtc::Optional incoming_bitrate_bps = + incoming_bitrate_.Rate(now_ms); + if (incoming_bitrate_bps && + (prior_state != kBwOverusing || + remote_rate_->TimeToReduceFurther(now_ms, *incoming_bitrate_bps))) { // The first overuse should immediately trigger a new estimate. // We also have to update the estimate immediately if we are overusing // and the target bitrate is too high compared to what we are receiving. @@ -167,6 +183,7 @@ void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) { remote_rate_.reset(new AimdRateControl()); return; } + double mean_noise_var = sum_var_noise / static_cast(overuse_detectors_.size()); const RateControlInput input(bw_state, diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h index 41d570b707..244dd42af7 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h @@ -56,6 +56,7 @@ class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator { Clock* clock_; SsrcOveruseEstimatorMap overuse_detectors_ GUARDED_BY(crit_sect_.get()); RateStatistics incoming_bitrate_ GUARDED_BY(crit_sect_.get()); + uint32_t last_valid_incoming_bitrate_ GUARDED_BY(crit_sect_.get()); std::unique_ptr remote_rate_ GUARDED_BY(crit_sect_.get()); RemoteBitrateObserver* observer_ GUARDED_BY(crit_sect_.get()); std::unique_ptr crit_sect_; diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc index 97e3abaa32..98f495eac8 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc @@ -35,7 +35,7 @@ TEST_F(RemoteBitrateEstimatorSingleTest, RateIncreaseReordering) { } TEST_F(RemoteBitrateEstimatorSingleTest, RateIncreaseRtpTimestamps) { - RateIncreaseRtpTimestampsTestHelper(1240); + RateIncreaseRtpTimestampsTestHelper(1267); } TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropOneStream) { @@ -47,15 +47,15 @@ TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropOneStreamWrap) { } TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropTwoStreamsWrap) { - CapacityDropTestHelper(2, true, 600); + CapacityDropTestHelper(2, true, 767); } TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThreeStreamsWrap) { - CapacityDropTestHelper(3, true, 767); + CapacityDropTestHelper(3, true, 567); } TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThirteenStreamsWrap) { - CapacityDropTestHelper(13, true, 733); + CapacityDropTestHelper(13, true, 567); } TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropNineteenStreamsWrap) { diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc index f04d9e68c0..3cff49882d 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc @@ -18,6 +18,9 @@ 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 testing { void TestBitrateObserver::OnReceiveBitrateChanged( @@ -317,16 +320,16 @@ void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper( EXPECT_FALSE(bitrate_observer_->updated()); bitrate_observer_->Reset(); clock_.AdvanceTimeMilliseconds(1000); - // Inserting a packet. Still no valid estimate. We need to wait 5 seconds. - IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, - absolute_send_time, true); - bitrate_estimator_->Process(); - EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); - EXPECT_EQ(0u, ssrcs.size()); - EXPECT_FALSE(bitrate_observer_->updated()); - bitrate_observer_->Reset(); // Inserting packets for 5 seconds to get a valid estimate. - for (int i = 0; i < 5 * kFramerate + 1; ++i) { + 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(); + } + IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, absolute_send_time, true); clock_.AdvanceTimeMilliseconds(1000 / kFramerate); @@ -355,12 +358,16 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper( const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate); uint32_t timestamp = 0; uint32_t absolute_send_time = 0; - IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp, - absolute_send_time, true); - bitrate_estimator_->Process(); - EXPECT_FALSE(bitrate_observer_->updated()); // No valid estimate. - // Inserting packets for one second to get a valid estimate. - for (int i = 0; i < 5 * kFramerate + 1; ++i) { + // 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. + } + IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, absolute_send_time, true); clock_.AdvanceTimeMilliseconds(kFrameIntervalMs); diff --git a/webrtc/video/receive_statistics_proxy.cc b/webrtc/video/receive_statistics_proxy.cc index 7fa39612e9..57d1f569cf 100644 --- a/webrtc/video/receive_statistics_proxy.cc +++ b/webrtc/video/receive_statistics_proxy.cc @@ -247,7 +247,7 @@ void ReceiveStatisticsProxy::OnDecodedFrame() { rtc::CritScope lock(&crit_); decode_fps_estimator_.Update(1, now); - stats_.decode_frame_rate = decode_fps_estimator_.Rate(now); + stats_.decode_frame_rate = decode_fps_estimator_.Rate(now).value_or(0); } void ReceiveStatisticsProxy::OnRenderedFrame(int width, int height) { @@ -257,7 +257,7 @@ void ReceiveStatisticsProxy::OnRenderedFrame(int width, int height) { rtc::CritScope lock(&crit_); renders_fps_estimator_.Update(1, now); - stats_.render_frame_rate = renders_fps_estimator_.Rate(now); + stats_.render_frame_rate = renders_fps_estimator_.Rate(now).value_or(0); render_width_counter_.Add(width); render_height_counter_.Add(height); render_fps_tracker_.AddSamples(1);