| Index: webrtc/modules/audio_coding/main/acm2/acm_dump.cc
|
| diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc
|
| deleted file mode 100644
|
| index 9c624d97b67eb728c6f54729ee8a37e6359f2631..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc
|
| +++ /dev/null
|
| @@ -1,240 +0,0 @@
|
| -/*
|
| - * Copyright (c) 2015 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 "webrtc/modules/audio_coding/main/acm2/acm_dump.h"
|
| -
|
| -#include <deque>
|
| -
|
| -#include "webrtc/base/checks.h"
|
| -#include "webrtc/base/thread_annotations.h"
|
| -#include "webrtc/system_wrappers/interface/clock.h"
|
| -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
|
| -#include "webrtc/system_wrappers/interface/file_wrapper.h"
|
| -
|
| -#ifdef RTC_AUDIOCODING_DEBUG_DUMP
|
| -// Files generated at build-time by the protobuf compiler.
|
| -#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
|
| -#include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h"
|
| -#else
|
| -#include "webrtc/audio_coding/dump.pb.h"
|
| -#endif
|
| -#endif
|
| -
|
| -namespace webrtc {
|
| -
|
| -// Noop implementation if flag is not set
|
| -#ifndef RTC_AUDIOCODING_DEBUG_DUMP
|
| -class AcmDumpImpl final : public AcmDump {
|
| - public:
|
| - void StartLogging(const std::string& file_name, int duration_ms) override{};
|
| - void LogRtpPacket(bool incoming,
|
| - const uint8_t* packet,
|
| - size_t length) override{};
|
| - void LogDebugEvent(DebugEvent event_type,
|
| - const std::string& event_message) override{};
|
| - void LogDebugEvent(DebugEvent event_type) override{};
|
| -};
|
| -#else
|
| -
|
| -class AcmDumpImpl final : public AcmDump {
|
| - public:
|
| - AcmDumpImpl();
|
| -
|
| - void StartLogging(const std::string& file_name, int duration_ms) override;
|
| - void LogRtpPacket(bool incoming,
|
| - const uint8_t* packet,
|
| - size_t length) override;
|
| - void LogDebugEvent(DebugEvent event_type,
|
| - const std::string& event_message) override;
|
| - void LogDebugEvent(DebugEvent event_type) override;
|
| -
|
| - private:
|
| - // This function is identical to LogDebugEvent, but requires holding the lock.
|
| - void LogDebugEventLocked(DebugEvent event_type,
|
| - const std::string& event_message)
|
| - EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| - // Stops logging and clears the stored data and buffers.
|
| - void Clear() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| - // Adds a new event to the logfile if logging is active, or adds it to the
|
| - // list of recent log events otherwise.
|
| - void HandleEvent(ACMDumpEvent* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| - // Writes the event to the file. Note that this will destroy the state of the
|
| - // input argument.
|
| - void StoreToFile(ACMDumpEvent* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| - // Adds the event to the list of recent events, and removes any events that
|
| - // are too old and no longer fall in the time window.
|
| - void AddRecentEvent(const ACMDumpEvent& event)
|
| - EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| -
|
| - // Amount of time in microseconds to record log events, before starting the
|
| - // actual log.
|
| - const int recent_log_duration_us = 10000000;
|
| -
|
| - rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
|
| - rtc::scoped_ptr<webrtc::FileWrapper> file_ GUARDED_BY(crit_);
|
| - rtc::scoped_ptr<ACMDumpEventStream> stream_ GUARDED_BY(crit_);
|
| - std::deque<ACMDumpEvent> recent_log_events_ GUARDED_BY(crit_);
|
| - bool currently_logging_ GUARDED_BY(crit_);
|
| - int64_t start_time_us_ GUARDED_BY(crit_);
|
| - int64_t duration_us_ GUARDED_BY(crit_);
|
| - const webrtc::Clock* const clock_;
|
| -};
|
| -
|
| -namespace {
|
| -
|
| -// Convert from AcmDump's debug event enum (runtime format) to the corresponding
|
| -// protobuf enum (serialized format).
|
| -ACMDumpDebugEvent_EventType convertDebugEvent(AcmDump::DebugEvent event_type) {
|
| - switch (event_type) {
|
| - case AcmDump::DebugEvent::kLogStart:
|
| - return ACMDumpDebugEvent::LOG_START;
|
| - case AcmDump::DebugEvent::kLogEnd:
|
| - return ACMDumpDebugEvent::LOG_END;
|
| - case AcmDump::DebugEvent::kAudioPlayout:
|
| - return ACMDumpDebugEvent::AUDIO_PLAYOUT;
|
| - }
|
| - return ACMDumpDebugEvent::UNKNOWN_EVENT;
|
| -}
|
| -
|
| -} // Anonymous namespace.
|
| -
|
| -// AcmDumpImpl member functions.
|
| -AcmDumpImpl::AcmDumpImpl()
|
| - : crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
|
| - file_(webrtc::FileWrapper::Create()),
|
| - stream_(new webrtc::ACMDumpEventStream()),
|
| - currently_logging_(false),
|
| - start_time_us_(0),
|
| - duration_us_(0),
|
| - clock_(webrtc::Clock::GetRealTimeClock()) {
|
| -}
|
| -
|
| -void AcmDumpImpl::StartLogging(const std::string& file_name, int duration_ms) {
|
| - CriticalSectionScoped lock(crit_.get());
|
| - Clear();
|
| - if (file_->OpenFile(file_name.c_str(), false) != 0) {
|
| - return;
|
| - }
|
| - // Add LOG_START event to the recent event list. This call will also remove
|
| - // any events that are too old from the recent event list.
|
| - LogDebugEventLocked(DebugEvent::kLogStart, "");
|
| - currently_logging_ = true;
|
| - start_time_us_ = clock_->TimeInMicroseconds();
|
| - duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
|
| - // Write all the recent events to the log file.
|
| - for (auto&& event : recent_log_events_) {
|
| - StoreToFile(&event);
|
| - }
|
| - recent_log_events_.clear();
|
| -}
|
| -
|
| -void AcmDumpImpl::LogRtpPacket(bool incoming,
|
| - const uint8_t* packet,
|
| - size_t length) {
|
| - CriticalSectionScoped lock(crit_.get());
|
| - ACMDumpEvent rtp_event;
|
| - const int64_t timestamp = clock_->TimeInMicroseconds();
|
| - rtp_event.set_timestamp_us(timestamp);
|
| - rtp_event.set_type(webrtc::ACMDumpEvent::RTP_EVENT);
|
| - rtp_event.mutable_packet()->set_direction(
|
| - incoming ? ACMDumpRTPPacket::INCOMING : ACMDumpRTPPacket::OUTGOING);
|
| - rtp_event.mutable_packet()->set_rtp_data(packet, length);
|
| - HandleEvent(&rtp_event);
|
| -}
|
| -
|
| -void AcmDumpImpl::LogDebugEvent(DebugEvent event_type,
|
| - const std::string& event_message) {
|
| - CriticalSectionScoped lock(crit_.get());
|
| - LogDebugEventLocked(event_type, event_message);
|
| -}
|
| -
|
| -void AcmDumpImpl::LogDebugEvent(DebugEvent event_type) {
|
| - CriticalSectionScoped lock(crit_.get());
|
| - LogDebugEventLocked(event_type, "");
|
| -}
|
| -
|
| -void AcmDumpImpl::LogDebugEventLocked(DebugEvent event_type,
|
| - const std::string& event_message) {
|
| - ACMDumpEvent event;
|
| - int64_t timestamp = clock_->TimeInMicroseconds();
|
| - event.set_timestamp_us(timestamp);
|
| - event.set_type(webrtc::ACMDumpEvent::DEBUG_EVENT);
|
| - auto debug_event = event.mutable_debug_event();
|
| - debug_event->set_type(convertDebugEvent(event_type));
|
| - debug_event->set_message(event_message);
|
| - HandleEvent(&event);
|
| -}
|
| -
|
| -void AcmDumpImpl::Clear() {
|
| - if (file_->Open()) {
|
| - file_->CloseFile();
|
| - }
|
| - currently_logging_ = false;
|
| - stream_->Clear();
|
| -}
|
| -
|
| -void AcmDumpImpl::HandleEvent(ACMDumpEvent* event) {
|
| - if (currently_logging_) {
|
| - if (clock_->TimeInMicroseconds() < start_time_us_ + duration_us_) {
|
| - StoreToFile(event);
|
| - } else {
|
| - LogDebugEventLocked(DebugEvent::kLogEnd, "");
|
| - Clear();
|
| - AddRecentEvent(*event);
|
| - }
|
| - } else {
|
| - AddRecentEvent(*event);
|
| - }
|
| -}
|
| -
|
| -void AcmDumpImpl::StoreToFile(ACMDumpEvent* event) {
|
| - // Reuse the same object at every log event.
|
| - if (stream_->stream_size() < 1) {
|
| - stream_->add_stream();
|
| - }
|
| - DCHECK_EQ(stream_->stream_size(), 1);
|
| - stream_->mutable_stream(0)->Swap(event);
|
| -
|
| - std::string dump_buffer;
|
| - stream_->SerializeToString(&dump_buffer);
|
| - file_->Write(dump_buffer.data(), dump_buffer.size());
|
| -}
|
| -
|
| -void AcmDumpImpl::AddRecentEvent(const ACMDumpEvent& event) {
|
| - recent_log_events_.push_back(event);
|
| - while (recent_log_events_.front().timestamp_us() <
|
| - event.timestamp_us() - recent_log_duration_us) {
|
| - recent_log_events_.pop_front();
|
| - }
|
| -}
|
| -
|
| -bool AcmDump::ParseAcmDump(const std::string& file_name,
|
| - ACMDumpEventStream* result) {
|
| - char tmp_buffer[1024];
|
| - int bytes_read = 0;
|
| - rtc::scoped_ptr<FileWrapper> dump_file(FileWrapper::Create());
|
| - if (dump_file->OpenFile(file_name.c_str(), true) != 0) {
|
| - return false;
|
| - }
|
| - std::string dump_buffer;
|
| - while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) {
|
| - dump_buffer.append(tmp_buffer, bytes_read);
|
| - }
|
| - dump_file->CloseFile();
|
| - return result->ParseFromString(dump_buffer);
|
| -}
|
| -
|
| -#endif // RTC_AUDIOCODING_DEBUG_DUMP
|
| -
|
| -// AcmDump member functions.
|
| -rtc::scoped_ptr<AcmDump> AcmDump::Create() {
|
| - return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl());
|
| -}
|
| -} // namespace webrtc
|
|
|