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Use webrtc/base/logging.h in stefan@'s ownership.

Browse Source 
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}
This commit is contained in:
Peter Boström 2015-12-04 16:13:05 +01:00
parent b572768efb
commit 7c704b8289
9 changed files with 61 additions and 69 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

109
webrtc/call/bitrate_estimator_tests.cc
View File

@ -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 @@ class TraceObserver {
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 @@ class TraceObserver {
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 @@ class BitrateEstimatorTest : public test::CallTest {
};
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 @@ static const char* kSingleStreamLog =
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

2
webrtc/call/call.cc
View File

@ -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"

2
webrtc/call/congestion_controller.cc
View File

@ -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"

2
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
View File

@ -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"

2
webrtc/modules/pacing/bitrate_prober.cc
View File

@ -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 {

2
webrtc/modules/pacing/paced_sender.cc
View File

@ -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.

2
webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
View File

@ -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 {

5
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
View File

@ -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 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(int64_t arrival_time_ms,
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);

4
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
View File

@ -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 {