Use webrtc/base/logging.h in stefan@'s ownership.
Replaces system_wrappers' logging in call/, bitrate_controller/, pacing/
and remote_bitrate_estimator/.
BUG=webrtc:5118
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1484503002 .
Cr-Commit-Position: refs/heads/master@{#10896}
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index 3ce96f6..5b07c54 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -15,6 +15,7 @@
#include "webrtc/audio_state.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
@@ -32,26 +33,12 @@
namespace webrtc {
namespace {
// Note: If you consider to re-use this class, think twice and instead consider
-// writing tests that don't depend on the trace system.
-class TraceObserver {
+// writing tests that don't depend on the logging system.
+class LogObserver {
public:
- TraceObserver() {
- Trace::set_level_filter(kTraceTerseInfo);
+ LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }
- Trace::CreateTrace();
- Trace::SetTraceCallback(&callback_);
-
- // Call webrtc trace to initialize the tracer that would otherwise trigger a
- // data-race if left to be initialized by multiple threads (i.e. threads
- // spawned by test::DirectTransport members in BitrateEstimatorTest).
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Instantiate without data races.");
- }
-
- ~TraceObserver() {
- Trace::SetTraceCallback(nullptr);
- Trace::ReturnTrace();
- }
+ ~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
void PushExpectedLogLine(const std::string& expected_log_line) {
callback_.PushExpectedLogLine(expected_log_line);
@@ -60,16 +47,20 @@
EventTypeWrapper Wait() { return callback_.Wait(); }
private:
- class Callback : public TraceCallback {
+ class Callback : public rtc::LogSink {
public:
Callback() : done_(EventWrapper::Create()) {}
- void Print(TraceLevel level, const char* message, int length) override {
+ void OnLogMessage(const std::string& message) override {
rtc::CritScope lock(&crit_sect_);
- std::string msg(message);
- if (msg.find("BitrateEstimator") != std::string::npos) {
- received_log_lines_.push_back(msg);
+ // Ignore log lines that are due to missing AST extensions, these are
+ // logged when we switch back from AST to TOF until the wrapping bitrate
+ // estimator gives up on using AST.
+ if (message.find("BitrateEstimator") != std::string::npos &&
+ message.find("packet is missing") == std::string::npos) {
+ received_log_lines_.push_back(message);
}
+
int num_popped = 0;
while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
std::string a = received_log_lines_.front();
@@ -77,7 +68,7 @@
received_log_lines_.pop_front();
expected_log_lines_.pop_front();
num_popped++;
- EXPECT_TRUE(a.find(b) != std::string::npos);
+ EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
}
if (expected_log_lines_.size() <= 0) {
if (num_popped > 0) {
@@ -260,7 +251,7 @@
};
testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
- TraceObserver receiver_trace_;
+ LogObserver receiver_log_;
rtc::scoped_ptr<test::DirectTransport> send_transport_;
rtc::scoped_ptr<test::DirectTransport> receive_transport_;
rtc::scoped_ptr<Call> sender_call_;
@@ -277,87 +268,87 @@
TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) {
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) {
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) {
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
- receiver_trace_.PushExpectedLogLine(
+ receiver_log_.PushExpectedLogLine(
"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
streams_[0]->StopSending();
streams_[1]->StopSending();
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_EQ(kEventSignaled, receiver_log_.Wait());
}
} // namespace webrtc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 441de87..4156765 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -18,6 +18,7 @@
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
@@ -35,7 +36,6 @@
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/call/congestion_controller.cc b/webrtc/call/congestion_controller.cc
index 387792f..18c880c 100644
--- a/webrtc/call/congestion_controller.cc
+++ b/webrtc/call/congestion_controller.cc
@@ -11,6 +11,7 @@
#include "webrtc/call/congestion_controller.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
@@ -24,7 +25,6 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/video_engine/call_stats.h"
#include "webrtc/video_engine/payload_router.h"
#include "webrtc/video_engine/vie_encoder.h"
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index c1c10e8..96a3b47 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -13,8 +13,8 @@
#include <cmath>
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/field_trial.h"
-#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/call/rtc_event_log.h"
diff --git a/webrtc/modules/pacing/bitrate_prober.cc b/webrtc/modules/pacing/bitrate_prober.cc
index db903f06..41ad5fa 100644
--- a/webrtc/modules/pacing/bitrate_prober.cc
+++ b/webrtc/modules/pacing/bitrate_prober.cc
@@ -15,8 +15,8 @@
#include <limits>
#include <sstream>
+#include "webrtc/base/logging.h"
#include "webrtc/modules/pacing/paced_sender.h"
-#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index dcdf64e..37e8ed7 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -15,12 +15,12 @@
#include <set>
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/pacing/bitrate_prober.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/field_trial.h"
-#include "webrtc/system_wrappers/include/logging.h"
namespace {
// Time limit in milliseconds between packet bursts.
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc b/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
index 4be7b74..36c3acd 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
@@ -16,8 +16,8 @@
#include <stdlib.h>
#include <string.h>
+#include "webrtc/base/logging.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
-#include "webrtc/system_wrappers/include/logging.h"
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 51f7d43..8006fe8 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
@@ -13,13 +13,13 @@
#include <math.h>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -241,6 +241,7 @@
if (!header.extension.hasAbsoluteSendTime) {
LOG(LS_WARNING) << "RemoteBitrateEstimatorAbsSendTimeImpl: Incoming packet "
"is missing absolute send time extension!";
+ return;
}
IncomingPacketInfo(arrival_time_ms, header.extension.absoluteSendTime,
payload_size, header.ssrc, was_paced);
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 f1a1cb6..a84377b 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
@@ -11,15 +11,15 @@
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
#include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
#include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
-#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/typedefs.h"
namespace webrtc {
