Resolve dependency between rtc_event_log_api and remote_bitrate_estimator
BUG=webrtc:7257
Review-Url: https://codereview.webrtc.org/2800633004
Cr-Commit-Position: refs/heads/master@{#17638}
diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn
index 11c816e..6fcab59 100644
--- a/webrtc/logging/BUILD.gn
+++ b/webrtc/logging/BUILD.gn
@@ -23,12 +23,6 @@
}
rtc_source_set("rtc_event_log_api") {
- # TODO(kjellander): Remove (bugs.webrtc.org/7257)
- # Enabling GN check triggers cyclic dependency error:
- # //webrtc/logging:rtc_event_log_api ->
- # //webrtc/modules/audio_coding:audio_network_adaptor ->
- # //webrtc/logging:rtc_event_log_api
- check_includes = false
sources = [
"rtc_event_log/rtc_event_log.h",
]
@@ -55,6 +49,7 @@
"../base:rtc_base_approved",
"../call:call_interfaces",
"../modules/audio_coding:audio_network_adaptor",
+ "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../system_wrappers",
]
@@ -89,6 +84,7 @@
"..:webrtc_common",
"../call:call_interfaces",
"../modules/audio_coding:audio_network_adaptor",
+ "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp",
"../system_wrappers",
]
@@ -118,6 +114,7 @@
"../base:rtc_base_tests_utils",
"../call",
"../modules/audio_coding:audio_network_adaptor",
+ "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../system_wrappers:metrics_default",
"../test:test_support",
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc
index 68d213e..ffeb9d2 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
@@ -24,6 +24,7 @@
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h
index f842252..a98c71b 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.h
@@ -17,7 +17,6 @@
#include "webrtc/base/platform_file.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/audio_send_stream.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@@ -34,6 +33,7 @@
struct AudioEncoderRuntimeConfig;
enum class MediaType;
+enum class BandwidthUsage;
enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
enum ProbeFailureReason {
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
index feffff0..a6ff8af 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -24,6 +24,7 @@
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
@@ -97,14 +98,14 @@
rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
switch (detector_state) {
case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
- return kBwNormal;
+ return BandwidthUsage::kBwNormal;
case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
- return kBwUnderusing;
+ return BandwidthUsage::kBwUnderusing;
case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
- return kBwOverusing;
+ return BandwidthUsage::kBwOverusing;
}
RTC_NOTREACHED();
- return kBwNormal;
+ return BandwidthUsage::kBwNormal;
}
std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
index d41a883..8be739e 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -23,6 +23,7 @@
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
@@ -560,11 +561,11 @@
std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
log_dumper->StartLogging(temp_filename, 10000000);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
- log_dumper->LogDelayBasedBweUpdate(bitrate1, kBwNormal);
+ log_dumper->LogDelayBasedBweUpdate(bitrate1, BandwidthUsage::kBwNormal);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
- log_dumper->LogDelayBasedBweUpdate(bitrate2, kBwOverusing);
+ log_dumper->LogDelayBasedBweUpdate(bitrate2, BandwidthUsage::kBwOverusing);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
- log_dumper->LogDelayBasedBweUpdate(bitrate3, kBwUnderusing);
+ log_dumper->LogDelayBasedBweUpdate(bitrate3, BandwidthUsage::kBwUnderusing);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->StopLogging();
@@ -577,11 +578,11 @@
EXPECT_EQ(5u, parsed_log.GetNumberOfEvents());
RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
RtcEventLogTestHelper::VerifyBweDelayEvent(parsed_log, 1, bitrate1,
- kBwNormal);
+ BandwidthUsage::kBwNormal);
RtcEventLogTestHelper::VerifyBweDelayEvent(parsed_log, 2, bitrate2,
- kBwOverusing);
+ BandwidthUsage::kBwOverusing);
RtcEventLogTestHelper::VerifyBweDelayEvent(parsed_log, 3, bitrate3,
- kBwUnderusing);
+ BandwidthUsage::kBwUnderusing);
RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 4);
// Clean up temporary file - can be pretty slow.
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index 7519ee5..4c0668c 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -16,6 +16,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
@@ -48,14 +49,14 @@
rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
switch (detector_state) {
case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
- return kBwNormal;
+ return BandwidthUsage::kBwNormal;
case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
- return kBwUnderusing;
+ return BandwidthUsage::kBwUnderusing;
case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
- return kBwOverusing;
+ return BandwidthUsage::kBwOverusing;
}
RTC_NOTREACHED();
- return kBwNormal;
+ return BandwidthUsage::kBwNormal;
}
rtclog::BweProbeResult::ResultType GetProbeResultType(
diff --git a/webrtc/modules/congestion_controller/delay_based_bwe.cc b/webrtc/modules/congestion_controller/delay_based_bwe.cc
index 11ea0e9..453b03e 100644
--- a/webrtc/modules/congestion_controller/delay_based_bwe.cc
+++ b/webrtc/modules/congestion_controller/delay_based_bwe.cc
@@ -163,7 +163,7 @@
probing_interval_estimator_(&rate_control_),
consecutive_delayed_feedbacks_(0),
last_logged_bitrate_(0),
- last_logged_state_(kBwNormal) {
+ last_logged_state_(BandwidthUsage::kBwNormal) {
LOG(LS_INFO) << "Using Trendline filter for delay change estimation.";
network_thread_.DetachFromThread();
@@ -277,7 +277,7 @@
rtc::Optional<uint32_t> acked_bitrate_bps =
receiver_incoming_bitrate_.bitrate_bps();
// Currently overusing the bandwidth.
- if (detector_.State() == kBwOverusing) {
+ if (detector_.State() == BandwidthUsage::kBwOverusing) {
if (acked_bitrate_bps &&
rate_control_.TimeToReduceFurther(now_ms, *acked_bitrate_bps)) {
result.updated =
diff --git a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
index 5b0ba5a..2428ddf 100644
--- a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
+++ b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -147,7 +147,8 @@
// An over-use should always trigger us to reduce the bitrate, even though
// we have not yet established our first estimate. By acting on the over-use,
// we will end up with a valid estimate.
- if (!bitrate_is_initialized_ && input.bw_state != kBwOverusing)
+ if (!bitrate_is_initialized_ &&
+ input.bw_state != BandwidthUsage::kBwOverusing)
return current_bitrate_bps_;
ChangeState(input, now_ms);
@@ -279,18 +280,18 @@
void AimdRateControl::ChangeState(const RateControlInput& input,
int64_t now_ms) {
switch (input.bw_state) {
- case kBwNormal:
+ case BandwidthUsage::kBwNormal:
if (rate_control_state_ == kRcHold) {
time_last_bitrate_change_ = now_ms;
rate_control_state_ = kRcIncrease;
}
break;
- case kBwOverusing:
+ case BandwidthUsage::kBwOverusing:
if (rate_control_state_ != kRcDecrease) {
rate_control_state_ = kRcDecrease;
}
break;
- case kBwUnderusing:
+ case BandwidthUsage::kBwUnderusing:
rate_control_state_ = kRcHold;
break;
default:
diff --git a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
index 68e6f51..d15418f 100644
--- a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
@@ -77,7 +77,7 @@
constexpr int kBitrate = 300000;
states.aimd_rate_control->SetEstimate(
kBitrate, states.simulated_clock->TimeInMilliseconds());
- UpdateRateControl(states, kBwOverusing, kBitrate - 2000,
+ UpdateRateControl(states, BandwidthUsage::kBwOverusing, kBitrate - 2000,
states.simulated_clock->TimeInMilliseconds());
EXPECT_EQ(rtc::Optional<int>(46700),
states.aimd_rate_control->GetLastBitrateDecreaseBps());
@@ -90,7 +90,7 @@
kAckedBitrate, states.simulated_clock->TimeInMilliseconds());
while (states.simulated_clock->TimeInMilliseconds() - kClockInitialTime <
20000) {
- UpdateRateControl(states, kBwNormal, kAckedBitrate,
+ UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate,
states.simulated_clock->TimeInMilliseconds());
states.simulated_clock->AdvanceTimeMilliseconds(100);
}
@@ -106,7 +106,7 @@
kAckedBitrate, states.simulated_clock->TimeInMilliseconds());
while (states.simulated_clock->TimeInMilliseconds() - kClockInitialTime <
20000) {
- UpdateRateControl(states, kBwNormal, kAckedBitrate,
+ UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate,
states.simulated_clock->TimeInMilliseconds());
states.simulated_clock->AdvanceTimeMilliseconds(100);
}
@@ -114,7 +114,7 @@
// If the acked bitrate decreases the BWE shouldn't be reduced to 1.5x
// what's being acked, but also shouldn't get to increase more.
uint32_t prev_estimate = states.aimd_rate_control->LatestEstimate();
- UpdateRateControl(states, kBwNormal, kAckedBitrate / 2,
+ UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate / 2,
states.simulated_clock->TimeInMilliseconds());
uint32_t new_estimate = states.aimd_rate_control->LatestEstimate();
EXPECT_NEAR(new_estimate, static_cast<uint32_t>(1.5 * kAckedBitrate + 10000),
diff --git a/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h b/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
index 26bfb28..ab05e7f 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
+++ b/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
@@ -35,7 +35,7 @@
kBweNamesMax = 4
};
-enum BandwidthUsage {
+enum class BandwidthUsage {
kBwNormal = 0,
kBwUnderusing = 1,
kBwOverusing = 2,
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc b/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
index 2e3b4b3..399daf0 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
@@ -70,7 +70,7 @@
prev_offset_(0.0),
time_over_using_(-1),
overuse_counter_(0),
- hypothesis_(kBwNormal) {
+ hypothesis_(BandwidthUsage::kBwNormal) {
if (!AdaptiveThresholdExperimentIsDisabled())
InitializeExperiment();
}
@@ -86,7 +86,7 @@
int num_of_deltas,
int64_t now_ms) {
if (num_of_deltas < 2) {
- return kBwNormal;
+ return BandwidthUsage::kBwNormal;
}
const double T = std::min(num_of_deltas, kMinNumDeltas) * offset;
BWE_TEST_LOGGING_PLOT(1, "offset_ms#1", now_ms, offset);
@@ -106,17 +106,17 @@
if (offset >= prev_offset_) {
time_over_using_ = 0;
overuse_counter_ = 0;
- hypothesis_ = kBwOverusing;
+ hypothesis_ = BandwidthUsage::kBwOverusing;
}
}
} else if (T < -threshold_) {
time_over_using_ = -1;
overuse_counter_ = 0;
- hypothesis_ = kBwUnderusing;
+ hypothesis_ = BandwidthUsage::kBwUnderusing;
} else {
time_over_using_ = -1;
overuse_counter_ = 0;
- hypothesis_ = kBwNormal;
+ hypothesis_ = BandwidthUsage::kBwNormal;
}
prev_offset_ = offset;
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc b/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
index fb5d4e7..0178f33 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
@@ -59,7 +59,7 @@
receive_time_ms_,
now_ms_ + static_cast<int64_t>(
random_.Gaussian(0, standard_deviation_ms) + 0.5));
- if (kBwOverusing == overuse_detector_->State()) {
+ if (BandwidthUsage::kBwOverusing == overuse_detector_->State()) {
if (last_overuse + 1 != i) {
unique_overuse++;
}
@@ -82,7 +82,7 @@
receive_time_ms_,
now_ms_ + static_cast<int64_t>(
random_.Gaussian(0, standard_deviation_ms) + 0.5));
- if (kBwOverusing == overuse_detector_->State()) {
+ if (BandwidthUsage::kBwOverusing == overuse_detector_->State()) {
return i + 1;
}
}
@@ -139,7 +139,7 @@
UpdateDetector(rtp_timestamp, now_ms_, packet_size);
now_ms_ += frame_duration_ms;
rtp_timestamp += frame_duration_ms * 90;
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
}
@@ -157,7 +157,7 @@
} else {
now_ms_ += frame_duration_ms + 5;
}
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
}
@@ -175,7 +175,7 @@
} else {
rtp_timestamp += (frame_duration_ms + 5) * 90;
}
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
}
@@ -226,7 +226,7 @@
} else {
now_ms_ += frame_duration_ms + offset;
}
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
// Simulate a higher send pace, that is too high.
// Above noise generate a standard deviation of approximately 28 ms.
@@ -235,10 +235,10 @@
UpdateDetector(rtp_timestamp, now_ms_, packet_size);
now_ms_ += frame_duration_ms + drift_per_frame_ms;
rtp_timestamp += frame_duration_ms * 90;
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
UpdateDetector(rtp_timestamp, now_ms_, packet_size);
- EXPECT_EQ(kBwOverusing, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwOverusing, overuse_detector_->State());
}
TEST_F(OveruseDetectorTest, DISABLED_OveruseWithLowVariance100Kbit10fps) {
@@ -258,7 +258,7 @@
} else {
now_ms_ += frame_duration_ms + offset;
}
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
// Simulate a higher send pace, that is too high.
// Total build up of 6 ms.
@@ -266,10 +266,10 @@
UpdateDetector(rtp_timestamp, now_ms_, packet_size);
now_ms_ += frame_duration_ms + drift_per_frame_ms;
rtp_timestamp += frame_duration_ms * 90;
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
UpdateDetector(rtp_timestamp, now_ms_, packet_size);
- EXPECT_EQ(kBwOverusing, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwOverusing, overuse_detector_->State());
}
TEST_F(OveruseDetectorTest, OveruseWithLowVariance2000Kbit30fps) {
@@ -294,7 +294,7 @@
} else {
now_ms_ += frame_duration_ms + offset;
}
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
// Simulate a higher send pace, that is too high.
// Total build up of 30 ms.
@@ -307,10 +307,10 @@
UpdateDetector(rtp_timestamp, now_ms_, packet_size);
now_ms_ += frame_duration_ms + drift_per_frame_ms * 6;
rtp_timestamp += frame_duration_ms * 90;
- EXPECT_EQ(kBwNormal, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
}
UpdateDetector(rtp_timestamp, now_ms_, packet_size);
- EXPECT_EQ(kBwOverusing, overuse_detector_->State());
+ EXPECT_EQ(BandwidthUsage::kBwOverusing, overuse_detector_->State());
}
#if defined(WEBRTC_ANDROID)
@@ -667,7 +667,7 @@
for (int i = 0; i < kBatchLength; ++i) {
BandwidthUsage overuse_state =
overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
- if (overuse_state == kBwOverusing) {
+ if (overuse_state == BandwidthUsage::kBwOverusing) {
overuse_detected = true;
}
++num_deltas;
@@ -680,7 +680,7 @@
for (int i = 0; i < kBatchLength; ++i) {
BandwidthUsage overuse_state =
overuse_detector_->Detect(1.1 * kOffset, kTsDelta, num_deltas, now_ms);
- if (overuse_state == kBwOverusing) {
+ if (overuse_state == BandwidthUsage::kBwOverusing) {
overuse_detected = true;
}
++num_deltas;
@@ -693,7 +693,7 @@
for (int i = 0; i < kBatchLength; ++i) {
BandwidthUsage overuse_state =
overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
- if (overuse_state == kBwOverusing) {
+ if (overuse_state == BandwidthUsage::kBwOverusing) {
overuse_detected = true;
}
++num_deltas;
@@ -705,7 +705,7 @@
for (int i = 0; i < 15 * kBatchLength; ++i) {
BandwidthUsage overuse_state =
overuse_detector_->Detect(0.7 * kOffset, kTsDelta, num_deltas, now_ms);
- if (overuse_state == kBwOverusing) {
+ if (overuse_state == BandwidthUsage::kBwOverusing) {
overuse_detected = true;
}
++num_deltas;
@@ -717,7 +717,7 @@
for (int i = 0; i < kBatchLength; ++i) {
BandwidthUsage overuse_state =
overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
- if (overuse_state == kBwOverusing) {
+ if (overuse_state == BandwidthUsage::kBwOverusing) {
overuse_detected = true;
}
++num_deltas;
@@ -740,7 +740,7 @@
for (int i = 0; i < kBatchLength; ++i) {
BandwidthUsage overuse_state =
overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
- if (overuse_state == kBwOverusing) {
+ if (overuse_state == BandwidthUsage::kBwOverusing) {
overuse_detected = true;
}
++num_deltas;
@@ -754,7 +754,7 @@
for (int i = 0; i < kShortBatchLength; ++i) {
BandwidthUsage overuse_state =
overuse_detector_->Detect(kLargeOffset, kTsDelta, num_deltas, now_ms);
- if (overuse_state == kBwOverusing) {
+ if (overuse_state == BandwidthUsage::kBwOverusing) {
overuse_detected = true;
}
++num_deltas;
@@ -767,7 +767,7 @@
for (int i = 0; i < kBatchLength; ++i) {
BandwidthUsage overuse_state =
overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
- if (overuse_state == kBwOverusing) {
+ if (overuse_state == BandwidthUsage::kBwOverusing) {
overuse_detected = true;
}
++num_deltas;
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc b/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
index 85efecd..0beab30 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
@@ -65,8 +65,10 @@
E_[0][0] += process_noise_[0];
E_[1][1] += process_noise_[1];
- if ((current_hypothesis == kBwOverusing && offset_ < prev_offset_) ||
- (current_hypothesis == kBwUnderusing && offset_ > prev_offset_)) {
+ if ((current_hypothesis == BandwidthUsage::kBwOverusing &&
+ offset_ < prev_offset_) ||
+ (current_hypothesis == BandwidthUsage::kBwUnderusing &&
+ offset_ > prev_offset_)) {
E_[1][1] += 10 * process_noise_[1];
}
@@ -78,7 +80,8 @@
const double residual = t_ts_delta - slope_*h[0] - offset_;
- const bool in_stable_state = (current_hypothesis == kBwNormal);
+ const bool in_stable_state =
+ (current_hypothesis == BandwidthUsage::kBwNormal);
const double max_residual = 3.0 * sqrt(var_noise_);
// We try to filter out very late frames. For instance periodic key
// frames doesn't fit the Gaussian model well.
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 5e9ba6f..23fbd2a 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
@@ -315,7 +315,7 @@
if (last_update_ms_ == -1 ||
now_ms - last_update_ms_ > remote_rate_.GetFeedbackInterval()) {
update_estimate = true;
- } else if (detector_.State() == kBwOverusing) {
+ } else if (detector_.State() == BandwidthUsage::kBwOverusing) {
rtc::Optional<uint32_t> incoming_rate =
incoming_bitrate_.Rate(arrival_time_ms);
if (incoming_rate &&
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 86ed41f..5ac8746 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
@@ -125,11 +125,11 @@
timestamp_delta_ms,
estimator->estimator.num_of_deltas(), now_ms);
}
- if (estimator->detector.State() == kBwOverusing) {
+ if (estimator->detector.State() == BandwidthUsage::kBwOverusing) {
rtc::Optional<uint32_t> incoming_bitrate_bps =
incoming_bitrate_.Rate(now_ms);
if (incoming_bitrate_bps &&
- (prior_state != kBwOverusing ||
+ (prior_state != BandwidthUsage::kBwOverusing ||
GetRemoteRate()->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
@@ -158,7 +158,7 @@
}
void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) {
- BandwidthUsage bw_state = kBwNormal;
+ BandwidthUsage bw_state = BandwidthUsage::kBwNormal;
double sum_var_noise = 0.0;
SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.begin();
while (it != overuse_detectors_.end()) {
