diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/rtc_base/logging.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/rtc_base/logging.cc')
-rw-r--r-- | third_party/libwebrtc/rtc_base/logging.cc | 608 |
1 files changed, 608 insertions, 0 deletions
diff --git a/third_party/libwebrtc/rtc_base/logging.cc b/third_party/libwebrtc/rtc_base/logging.cc new file mode 100644 index 0000000000..9c2d3b0a39 --- /dev/null +++ b/third_party/libwebrtc/rtc_base/logging.cc @@ -0,0 +1,608 @@ +/* + * Copyright 2004 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 "rtc_base/logging.h" + +#include <string.h> + +#if RTC_LOG_ENABLED() + +#if defined(WEBRTC_WIN) +#include <windows.h> +#if _MSC_VER < 1900 +#define snprintf _snprintf +#endif +#undef ERROR // wingdi.h +#endif + +#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) +#include <CoreServices/CoreServices.h> +#elif defined(WEBRTC_ANDROID) +#include <android/log.h> + +// Android has a 1024 limit on log inputs. We use 60 chars as an +// approx for the header/tag portion. +// See android/system/core/liblog/logd_write.c +static const int kMaxLogLineSize = 1024 - 60; +#endif // WEBRTC_MAC && !defined(WEBRTC_IOS) || WEBRTC_ANDROID + +#include <inttypes.h> +#include <stdio.h> +#include <time.h> + +#include <algorithm> +#include <cstdarg> +#include <vector> + +#include "absl/base/attributes.h" +#include "absl/strings/string_view.h" +#include "api/units/timestamp.h" +#include "modules/audio_processing/logging/apm_data_dumper.h" +#include "rtc_base/checks.h" +#include "rtc_base/platform_thread_types.h" +#include "rtc_base/string_encode.h" +#include "rtc_base/string_utils.h" +#include "rtc_base/strings/string_builder.h" +#include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" +#include "rtc_base/time_utils.h" + +namespace rtc { + +bool LogMessage::aec_debug_ = false; +std::string LogMessage::aec_filename_base_; + +void LogMessage::set_aec_debug(bool enable) { + aec_debug_ = enable; + webrtc::ApmDataDumper::SetActivated(aec_debug_); +} + +std::string LogMessage::aec_debug_filename() { + return aec_filename_base_; +} + +void LogMessage::set_aec_debug_filename(const char* filename) { + aec_filename_base_ = filename; + webrtc::ApmDataDumper::SetOutputDirectory(aec_filename_base_); +} + +namespace { + +// By default, release builds don't log, debug builds at info level +#if !defined(NDEBUG) +constexpr LoggingSeverity kDefaultLoggingSeverity = LS_INFO; +#else +constexpr LoggingSeverity kDefaultLoggingSeverity = LS_NONE; +#endif + +// Note: `g_min_sev` and `g_dbg_sev` can be changed while running. +LoggingSeverity g_min_sev = kDefaultLoggingSeverity; +LoggingSeverity g_dbg_sev = kDefaultLoggingSeverity; + +// Return the filename portion of the string (that following the last slash). +const char* FilenameFromPath(const char* file) { + const char* end1 = ::strrchr(file, '/'); + const char* end2 = ::strrchr(file, '\\'); + if (!end1 && !end2) + return file; + else + return (end1 > end2) ? end1 + 1 : end2 + 1; +} + +// Global lock for log subsystem, only needed to serialize access to streams_. +webrtc::Mutex& GetLoggingLock() { + static webrtc::Mutex& mutex = *new webrtc::Mutex(); + return mutex; +} + +} // namespace + +std::string LogLineRef::DefaultLogLine() const { + rtc::StringBuilder log_output; + if (timestamp_ != webrtc::Timestamp::MinusInfinity()) { + // TODO(kwiberg): Switch to absl::StrFormat, if binary size is ok. + char timestamp[50]; // Maximum string length of an int64_t is 20. + int len = + snprintf(timestamp, sizeof(timestamp), "[%03" PRId64 ":%03" PRId64 "]", + timestamp_.ms() / 1000, timestamp_.ms() % 1000); + RTC_DCHECK_LT(len, sizeof(timestamp)); + log_output << timestamp; + } + if (thread_id_.has_value()) { + log_output << "[" << *thread_id_ << "] "; + } + if (!filename_.empty()) { +#if defined(WEBRTC_ANDROID) + log_output << "(line " << line_ << "): "; +#else + log_output << "(" << filename_ << ":" << line_ << "): "; +#endif + } + log_output << message_; + return log_output.Release(); +} + +///////////////////////////////////////////////////////////////////////////// +// LogMessage +///////////////////////////////////////////////////////////////////////////// + +bool LogMessage::log_to_stderr_ = false; + +// The list of logging streams currently configured. +// Note: we explicitly do not clean this up, because of the uncertain ordering +// of destructors at program exit. Let the person who sets the stream trigger +// cleanup by setting to null, or let it leak (safe at program exit). +ABSL_CONST_INIT LogSink* LogMessage::streams_ RTC_GUARDED_BY(GetLoggingLock()) = + nullptr; +ABSL_CONST_INIT std::atomic<bool> LogMessage::streams_empty_ = {true}; + +// Boolean options default to false. +ABSL_CONST_INIT bool LogMessage::log_thread_ = false; +ABSL_CONST_INIT bool LogMessage::log_timestamp_ = false; + +LogMessage::LogMessage(const char* file, int line, LoggingSeverity sev) + : LogMessage(file, line, sev, ERRCTX_NONE, 0) {} + +LogMessage::LogMessage(const char* file, + int line, + LoggingSeverity sev, + LogErrorContext err_ctx, + int err) { + log_line_.set_severity(sev); + if (log_timestamp_) { + int64_t log_start_time = LogStartTime(); + // Use SystemTimeMillis so that even if tests use fake clocks, the timestamp + // in log messages represents the real system time. + int64_t time = TimeDiff(SystemTimeMillis(), log_start_time); + // Also ensure WallClockStartTime is initialized, so that it matches + // LogStartTime. + WallClockStartTime(); + log_line_.set_timestamp(webrtc::Timestamp::Millis(time)); + } + + if (log_thread_) { + log_line_.set_thread_id(CurrentThreadId()); + } + + if (file != nullptr) { + log_line_.set_filename(FilenameFromPath(file)); + log_line_.set_line(line); +#if defined(WEBRTC_ANDROID) + log_line_.set_tag(log_line_.filename()); +#endif + } + + if (err_ctx != ERRCTX_NONE) { + char tmp_buf[1024]; + SimpleStringBuilder tmp(tmp_buf); + tmp.AppendFormat("[0x%08X]", err); + switch (err_ctx) { + case ERRCTX_ERRNO: + tmp << " " << strerror(err); + break; +#ifdef WEBRTC_WIN + case ERRCTX_HRESULT: { + char msgbuf[256]; + DWORD flags = + FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS; + if (DWORD len = FormatMessageA( + flags, nullptr, err, MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT), + msgbuf, sizeof(msgbuf) / sizeof(msgbuf[0]), nullptr)) { + while ((len > 0) && + isspace(static_cast<unsigned char>(msgbuf[len - 1]))) { + msgbuf[--len] = 0; + } + tmp << " " << msgbuf; + } + break; + } +#endif // WEBRTC_WIN + default: + break; + } + extra_ = tmp.str(); + } +} + +#if defined(WEBRTC_ANDROID) +LogMessage::LogMessage(const char* file, + int line, + LoggingSeverity sev, + const char* tag) + : LogMessage(file, line, sev, ERRCTX_NONE, /*err=*/0) { + log_line_.set_tag(tag); + print_stream_ << tag << ": "; +} +#endif + +LogMessage::~LogMessage() { + FinishPrintStream(); + + log_line_.set_message(print_stream_.Release()); + + if (log_line_.severity() >= g_dbg_sev) { + OutputToDebug(log_line_); + } + + webrtc::MutexLock lock(&GetLoggingLock()); + for (LogSink* entry = streams_; entry != nullptr; entry = entry->next_) { + if (log_line_.severity() >= entry->min_severity_) { + entry->OnLogMessage(log_line_); + } + } +} + +void LogMessage::AddTag(const char* tag) { +#ifdef WEBRTC_ANDROID + log_line_.set_tag(tag); +#endif +} + +rtc::StringBuilder& LogMessage::stream() { + return print_stream_; +} + +int LogMessage::GetMinLogSeverity() { + return g_min_sev; +} + +LoggingSeverity LogMessage::GetLogToDebug() { + return g_dbg_sev; +} +int64_t LogMessage::LogStartTime() { + static const int64_t g_start = SystemTimeMillis(); + return g_start; +} + +uint32_t LogMessage::WallClockStartTime() { + static const uint32_t g_start_wallclock = time(nullptr); + return g_start_wallclock; +} + +void LogMessage::LogThreads(bool on) { + log_thread_ = on; +} + +void LogMessage::LogTimestamps(bool on) { + log_timestamp_ = on; +} + +void LogMessage::LogToDebug(LoggingSeverity min_sev) { + webrtc::MutexLock lock(&GetLoggingLock()); + g_dbg_sev = min_sev; + UpdateMinLogSeverity(); +} + +void LogMessage::SetLogToStderr(bool log_to_stderr) { + log_to_stderr_ = log_to_stderr; +} + +int LogMessage::GetLogToStream(LogSink* stream) { + webrtc::MutexLock lock(&GetLoggingLock()); + LoggingSeverity sev = LS_NONE; + for (LogSink* entry = streams_; entry != nullptr; entry = entry->next_) { + if (stream == nullptr || stream == entry) { + sev = std::min(sev, entry->min_severity_); + } + } + return sev; +} + +void LogMessage::AddLogToStream(LogSink* stream, LoggingSeverity min_sev) { + webrtc::MutexLock lock(&GetLoggingLock()); + stream->min_severity_ = min_sev; + stream->next_ = streams_; + streams_ = stream; + streams_empty_.store(false, std::memory_order_relaxed); + UpdateMinLogSeverity(); +} + +void LogMessage::RemoveLogToStream(LogSink* stream) { + webrtc::MutexLock lock(&GetLoggingLock()); + for (LogSink** entry = &streams_; *entry != nullptr; + entry = &(*entry)->next_) { + if (*entry == stream) { + *entry = (*entry)->next_; + break; + } + } + streams_empty_.store(streams_ == nullptr, std::memory_order_relaxed); + UpdateMinLogSeverity(); +} + +void LogMessage::ConfigureLogging(absl::string_view params) { + LoggingSeverity current_level = LS_VERBOSE; + LoggingSeverity debug_level = GetLogToDebug(); + + std::vector<std::string> tokens; + tokenize(params, ' ', &tokens); + + for (const std::string& token : tokens) { + if (token.empty()) + continue; + + // Logging features + if (token == "tstamp") { + LogTimestamps(); + } else if (token == "thread") { + LogThreads(); + + // Logging levels + } else if (token == "verbose") { + current_level = LS_VERBOSE; + } else if (token == "info") { + current_level = LS_INFO; + } else if (token == "warning") { + current_level = LS_WARNING; + } else if (token == "error") { + current_level = LS_ERROR; + } else if (token == "none") { + current_level = LS_NONE; + + // Logging targets + } else if (token == "debug") { + debug_level = current_level; + } + } + +#if defined(WEBRTC_WIN) && !defined(WINUWP) + if ((LS_NONE != debug_level) && !::IsDebuggerPresent()) { + // First, attempt to attach to our parent's console... so if you invoke + // from the command line, we'll see the output there. Otherwise, create + // our own console window. + // Note: These methods fail if a console already exists, which is fine. + if (!AttachConsole(ATTACH_PARENT_PROCESS)) + ::AllocConsole(); + } +#endif // defined(WEBRTC_WIN) && !defined(WINUWP) + + LogToDebug(debug_level); +} + +void LogMessage::UpdateMinLogSeverity() + RTC_EXCLUSIVE_LOCKS_REQUIRED(GetLoggingLock()) { + LoggingSeverity min_sev = g_dbg_sev; + for (LogSink* entry = streams_; entry != nullptr; entry = entry->next_) { + min_sev = std::min(min_sev, entry->min_severity_); + } + g_min_sev = min_sev; +} + +void LogMessage::OutputToDebug(const LogLineRef& log_line) { + std::string msg_str = log_line.DefaultLogLine(); + bool log_to_stderr = log_to_stderr_; +#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && defined(NDEBUG) + // On the Mac, all stderr output goes to the Console log and causes clutter. + // So in opt builds, don't log to stderr unless the user specifically sets + // a preference to do so. + CFStringRef domain = CFBundleGetIdentifier(CFBundleGetMainBundle()); + if (domain != nullptr) { + Boolean exists_and_is_valid; + Boolean should_log = CFPreferencesGetAppBooleanValue( + CFSTR("logToStdErr"), domain, &exists_and_is_valid); + // If the key doesn't exist or is invalid or is false, we will not log to + // stderr. + log_to_stderr = exists_and_is_valid && should_log; + } +#endif // defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && defined(NDEBUG) + +#if defined(WEBRTC_WIN) + // Always log to the debugger. + // Perhaps stderr should be controlled by a preference, as on Mac? + OutputDebugStringA(msg_str.c_str()); + if (log_to_stderr) { + // This handles dynamically allocated consoles, too. + if (HANDLE error_handle = ::GetStdHandle(STD_ERROR_HANDLE)) { + log_to_stderr = false; + DWORD written = 0; + ::WriteFile(error_handle, msg_str.c_str(), + static_cast<DWORD>(msg_str.size()), &written, 0); + } + } +#endif // WEBRTC_WIN + +#if defined(WEBRTC_ANDROID) + // Android's logging facility uses severity to log messages but we + // need to map libjingle's severity levels to Android ones first. + // Also write to stderr which maybe available to executable started + // from the shell. + int prio; + switch (log_line.severity()) { + case LS_VERBOSE: + prio = ANDROID_LOG_VERBOSE; + break; + case LS_INFO: + prio = ANDROID_LOG_INFO; + break; + case LS_WARNING: + prio = ANDROID_LOG_WARN; + break; + case LS_ERROR: + prio = ANDROID_LOG_ERROR; + break; + default: + prio = ANDROID_LOG_UNKNOWN; + } + + int size = msg_str.size(); + int current_line = 0; + int idx = 0; + const int max_lines = size / kMaxLogLineSize + 1; + if (max_lines == 1) { + __android_log_print(prio, log_line.tag().data(), "%.*s", size, + msg_str.c_str()); + } else { + while (size > 0) { + const int len = std::min(size, kMaxLogLineSize); + // Use the size of the string in the format (msg may have \0 in the + // middle). + __android_log_print(prio, log_line.tag().data(), "[%d/%d] %.*s", + current_line + 1, max_lines, len, + msg_str.c_str() + idx); + idx += len; + size -= len; + ++current_line; + } + } +#endif // WEBRTC_ANDROID + if (log_to_stderr) { + fprintf(stderr, "%s", msg_str.c_str()); + fflush(stderr); + } +} + +// static +bool LogMessage::IsNoop(LoggingSeverity severity) { + // Added MutexLock to fix tsan warnings on accessing g_dbg_sev. (mjf) + // See https://bugs.chromium.org/p/chromium/issues/detail?id=1228729 + webrtc::MutexLock lock(&GetLoggingLock()); + if (severity >= g_dbg_sev || severity >= g_min_sev) + return false; + return streams_empty_.load(std::memory_order_relaxed); +} + +void LogMessage::FinishPrintStream() { + if (!extra_.empty()) + print_stream_ << " : " << extra_; + print_stream_ << "\n"; +} + +namespace webrtc_logging_impl { + +void Log(const LogArgType* fmt, ...) { + va_list args; + va_start(args, fmt); + + LogMetadataErr meta; + const char* tag = nullptr; + switch (*fmt) { + case LogArgType::kLogMetadata: { + meta = {va_arg(args, LogMetadata), ERRCTX_NONE, 0}; + break; + } + case LogArgType::kLogMetadataErr: { + meta = va_arg(args, LogMetadataErr); + break; + } +#ifdef WEBRTC_ANDROID + case LogArgType::kLogMetadataTag: { + const LogMetadataTag tag_meta = va_arg(args, LogMetadataTag); + meta = {{nullptr, 0, tag_meta.severity}, ERRCTX_NONE, 0}; + tag = tag_meta.tag; + break; + } +#endif + default: { + RTC_DCHECK_NOTREACHED(); + va_end(args); + return; + } + } + + LogMessage log_message(meta.meta.File(), meta.meta.Line(), + meta.meta.Severity(), meta.err_ctx, meta.err); + if (tag) { + log_message.AddTag(tag); + } + + for (++fmt; *fmt != LogArgType::kEnd; ++fmt) { + switch (*fmt) { + case LogArgType::kInt: + log_message.stream() << va_arg(args, int); + break; + case LogArgType::kLong: + log_message.stream() << va_arg(args, long); + break; + case LogArgType::kLongLong: + log_message.stream() << va_arg(args, long long); + break; + case LogArgType::kUInt: + log_message.stream() << va_arg(args, unsigned); + break; + case LogArgType::kULong: + log_message.stream() << va_arg(args, unsigned long); + break; + case LogArgType::kULongLong: + log_message.stream() << va_arg(args, unsigned long long); + break; + case LogArgType::kDouble: + log_message.stream() << va_arg(args, double); + break; + case LogArgType::kLongDouble: + log_message.stream() << va_arg(args, long double); + break; + case LogArgType::kCharP: { + const char* s = va_arg(args, const char*); + log_message.stream() << (s ? s : "(null)"); + break; + } + case LogArgType::kStdString: + log_message.stream() << *va_arg(args, const std::string*); + break; + case LogArgType::kStringView: + log_message.stream() << *va_arg(args, const absl::string_view*); + break; + case LogArgType::kVoidP: + log_message.stream() << rtc::ToHex( + reinterpret_cast<uintptr_t>(va_arg(args, const void*))); + break; + default: + RTC_DCHECK_NOTREACHED(); + va_end(args); + return; + } + } + + va_end(args); +} + +} // namespace webrtc_logging_impl +} // namespace rtc +#endif + +namespace rtc { +// Default implementation, override is recomended. +void LogSink::OnLogMessage(const LogLineRef& log_line) { +#if defined(WEBRTC_ANDROID) + OnLogMessage(log_line.DefaultLogLine(), log_line.severity(), + log_line.tag().data()); +#else + OnLogMessage(log_line.DefaultLogLine(), log_line.severity()); +#endif +} + +// Inefficient default implementation, override is recommended. +void LogSink::OnLogMessage(const std::string& msg, + LoggingSeverity severity, + const char* tag) { + OnLogMessage(tag + (": " + msg), severity); +} + +void LogSink::OnLogMessage(const std::string& msg, + LoggingSeverity /* severity */) { + OnLogMessage(msg); +} + +// Inefficient default implementation, override is recommended. +void LogSink::OnLogMessage(absl::string_view msg, + LoggingSeverity severity, + const char* tag) { + OnLogMessage(tag + (": " + std::string(msg)), severity); +} + +void LogSink::OnLogMessage(absl::string_view msg, + LoggingSeverity /* severity */) { + OnLogMessage(msg); +} + +void LogSink::OnLogMessage(absl::string_view msg) { + OnLogMessage(std::string(msg)); +} +} // namespace rtc |