| // Copyright (c) 2013 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "content/browser/webrtc/webrtc_internals.h" |
| |
| #include <stddef.h> |
| |
| #include <memory> |
| #include <utility> |
| |
| #include "base/strings/string_number_conversions.h" |
| #include "build/build_config.h" |
| #include "content/browser/renderer_host/render_process_host_impl.h" |
| #include "content/browser/web_contents/web_contents_view.h" |
| #include "content/browser/webrtc/webrtc_internals_ui_observer.h" |
| #include "content/public/browser/browser_thread.h" |
| #include "content/public/browser/content_browser_client.h" |
| #include "content/public/browser/web_contents.h" |
| #include "device/power_save_blocker/power_save_blocker.h" |
| #include "ipc/ipc_platform_file.h" |
| |
| #if defined(OS_WIN) |
| #define IntToStringType base::IntToString16 |
| #else |
| #define IntToStringType base::IntToString |
| #endif |
| |
| using base::ProcessId; |
| using std::string; |
| |
| namespace content { |
| |
| namespace { |
| |
| static base::LazyInstance<WebRTCInternals>::Leaky g_webrtc_internals = |
| LAZY_INSTANCE_INITIALIZER; |
| |
| // Makes sure that |dict| has a ListValue under path "log". |
| static base::ListValue* EnsureLogList(base::DictionaryValue* dict) { |
| base::ListValue* log = NULL; |
| if (!dict->GetList("log", &log)) { |
| log = new base::ListValue(); |
| if (log) |
| dict->Set("log", log); |
| } |
| return log; |
| } |
| |
| } // namespace |
| |
| WebRTCInternals::PendingUpdate::PendingUpdate( |
| const std::string& command, |
| std::unique_ptr<base::Value> value) |
| : command_(command), value_(std::move(value)) {} |
| |
| WebRTCInternals::PendingUpdate::PendingUpdate(PendingUpdate&& other) |
| : command_(std::move(other.command_)), |
| value_(std::move(other.value_)) {} |
| |
| WebRTCInternals::PendingUpdate::~PendingUpdate() { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| } |
| |
| const std::string& WebRTCInternals::PendingUpdate::command() const { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| return command_; |
| } |
| |
| const base::Value* WebRTCInternals::PendingUpdate::value() const { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| return value_.get(); |
| } |
| |
| WebRTCInternals::WebRTCInternals() : WebRTCInternals(500, true) {} |
| |
| WebRTCInternals::WebRTCInternals(int aggregate_updates_ms, |
| bool should_block_power_saving) |
| : audio_debug_recordings_(false), |
| event_log_recordings_(false), |
| selecting_event_log_(false), |
| should_block_power_saving_(should_block_power_saving), |
| aggregate_updates_ms_(aggregate_updates_ms), |
| weak_factory_(this) { |
| // TODO(grunell): Shouldn't all the webrtc_internals* files be excluded from the |
| // build if WebRTC is disabled? |
| #if defined(ENABLE_WEBRTC) |
| audio_debug_recordings_file_path_ = |
| GetContentClient()->browser()->GetDefaultDownloadDirectory(); |
| event_log_recordings_file_path_ = audio_debug_recordings_file_path_; |
| |
| if (audio_debug_recordings_file_path_.empty()) { |
| // In this case the default path (|audio_debug_recordings_file_path_|) will |
| // be empty and the platform default path will be used in the file dialog |
| // (with no default file name). See SelectFileDialog::SelectFile. On Android |
| // where there's no dialog we'll fail to open the file. |
| VLOG(1) << "Could not get the download directory."; |
| } else { |
| audio_debug_recordings_file_path_ = |
| audio_debug_recordings_file_path_.Append( |
| FILE_PATH_LITERAL("audio_debug")); |
| event_log_recordings_file_path_ = |
| event_log_recordings_file_path_.Append(FILE_PATH_LITERAL("event_log")); |
| } |
| #endif // defined(ENABLE_WEBRTC) |
| } |
| |
| WebRTCInternals::~WebRTCInternals() { |
| } |
| |
| WebRTCInternals* WebRTCInternals::GetInstance() { |
| return g_webrtc_internals.Pointer(); |
| } |
| |
| void WebRTCInternals::OnAddPeerConnection(int render_process_id, |
| ProcessId pid, |
| int lid, |
| const string& url, |
| const string& rtc_configuration, |
| const string& constraints) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| |
| std::unique_ptr<base::DictionaryValue> dict(new base::DictionaryValue()); |
| dict->SetInteger("rid", render_process_id); |
| dict->SetInteger("pid", static_cast<int>(pid)); |
| dict->SetInteger("lid", lid); |
| dict->SetString("rtcConfiguration", rtc_configuration); |
| dict->SetString("constraints", constraints); |
| dict->SetString("url", url); |
| |
| if (observers_.might_have_observers()) |
| SendUpdate("addPeerConnection", dict->CreateDeepCopy()); |
| |
| peer_connection_data_.Append(std::move(dict)); |
| CreateOrReleasePowerSaveBlocker(); |
| |
| if (render_process_id_set_.insert(render_process_id).second) { |
| RenderProcessHost* host = RenderProcessHost::FromID(render_process_id); |
| if (host) |
| host->AddObserver(this); |
| } |
| } |
| |
| void WebRTCInternals::OnRemovePeerConnection(ProcessId pid, int lid) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| for (size_t i = 0; i < peer_connection_data_.GetSize(); ++i) { |
| base::DictionaryValue* dict = NULL; |
| peer_connection_data_.GetDictionary(i, &dict); |
| |
| int this_pid = 0; |
| int this_lid = 0; |
| dict->GetInteger("pid", &this_pid); |
| dict->GetInteger("lid", &this_lid); |
| |
| if (this_pid != static_cast<int>(pid) || this_lid != lid) |
| continue; |
| |
| peer_connection_data_.Remove(i, NULL); |
| CreateOrReleasePowerSaveBlocker(); |
| |
| if (observers_.might_have_observers()) { |
| std::unique_ptr<base::DictionaryValue> id(new base::DictionaryValue()); |
| id->SetInteger("pid", static_cast<int>(pid)); |
| id->SetInteger("lid", lid); |
| SendUpdate("removePeerConnection", std::move(id)); |
| } |
| break; |
| } |
| } |
| |
| void WebRTCInternals::OnUpdatePeerConnection( |
| ProcessId pid, int lid, const string& type, const string& value) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| |
| for (size_t i = 0; i < peer_connection_data_.GetSize(); ++i) { |
| base::DictionaryValue* record = NULL; |
| peer_connection_data_.GetDictionary(i, &record); |
| |
| int this_pid = 0, this_lid = 0; |
| record->GetInteger("pid", &this_pid); |
| record->GetInteger("lid", &this_lid); |
| |
| if (this_pid != static_cast<int>(pid) || this_lid != lid) |
| continue; |
| |
| // Append the update to the end of the log. |
| base::ListValue* log = EnsureLogList(record); |
| if (!log) |
| return; |
| |
| std::unique_ptr<base::DictionaryValue> log_entry( |
| new base::DictionaryValue()); |
| |
| double epoch_time = base::Time::Now().ToJsTime(); |
| string time = base::DoubleToString(epoch_time); |
| log_entry->SetString("time", time); |
| log_entry->SetString("type", type); |
| log_entry->SetString("value", value); |
| |
| if (observers_.might_have_observers()) { |
| std::unique_ptr<base::DictionaryValue> update( |
| new base::DictionaryValue()); |
| update->SetInteger("pid", static_cast<int>(pid)); |
| update->SetInteger("lid", lid); |
| update->MergeDictionary(log_entry.get()); |
| |
| SendUpdate("updatePeerConnection", std::move(update)); |
| } |
| |
| log->Append(std::move(log_entry)); |
| |
| return; |
| } |
| } |
| |
| void WebRTCInternals::OnAddStats(base::ProcessId pid, int lid, |
| const base::ListValue& value) { |
| if (!observers_.might_have_observers()) |
| return; |
| |
| std::unique_ptr<base::DictionaryValue> dict(new base::DictionaryValue()); |
| dict->SetInteger("pid", static_cast<int>(pid)); |
| dict->SetInteger("lid", lid); |
| |
| dict->Set("reports", value.CreateDeepCopy()); |
| |
| SendUpdate("addStats", std::move(dict)); |
| } |
| |
| void WebRTCInternals::OnGetUserMedia(int rid, |
| base::ProcessId pid, |
| const std::string& origin, |
| bool audio, |
| bool video, |
| const std::string& audio_constraints, |
| const std::string& video_constraints) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| |
| std::unique_ptr<base::DictionaryValue> dict(new base::DictionaryValue()); |
| dict->SetInteger("rid", rid); |
| dict->SetInteger("pid", static_cast<int>(pid)); |
| dict->SetString("origin", origin); |
| if (audio) |
| dict->SetString("audio", audio_constraints); |
| if (video) |
| dict->SetString("video", video_constraints); |
| |
| if (observers_.might_have_observers()) |
| SendUpdate("addGetUserMedia", dict->CreateDeepCopy()); |
| |
| get_user_media_requests_.Append(std::move(dict)); |
| |
| if (render_process_id_set_.insert(rid).second) { |
| RenderProcessHost* host = RenderProcessHost::FromID(rid); |
| if (host) |
| host->AddObserver(this); |
| } |
| } |
| |
| void WebRTCInternals::AddObserver(WebRTCInternalsUIObserver* observer) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| observers_.AddObserver(observer); |
| } |
| |
| void WebRTCInternals::RemoveObserver(WebRTCInternalsUIObserver* observer) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| observers_.RemoveObserver(observer); |
| |
| // Disables event log and audio debug recordings if enabled and the last |
| // webrtc-internals page is going away. |
| if (!observers_.might_have_observers()) { |
| if (audio_debug_recordings_) |
| DisableAudioDebugRecordings(); |
| DisableEventLogRecordings(); |
| } |
| } |
| |
| void WebRTCInternals::UpdateObserver(WebRTCInternalsUIObserver* observer) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| if (peer_connection_data_.GetSize() > 0) |
| observer->OnUpdate("updateAllPeerConnections", &peer_connection_data_); |
| |
| for (const auto& request : get_user_media_requests_) { |
| observer->OnUpdate("addGetUserMedia", request.get()); |
| } |
| } |
| |
| void WebRTCInternals::EnableAudioDebugRecordings( |
| content::WebContents* web_contents) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| #if defined(ENABLE_WEBRTC) |
| #if defined(OS_ANDROID) |
| EnableAudioDebugRecordingsOnAllRenderProcessHosts(); |
| #else |
| selecting_event_log_ = false; |
| DCHECK(!select_file_dialog_); |
| select_file_dialog_ = ui::SelectFileDialog::Create(this, NULL); |
| select_file_dialog_->SelectFile( |
| ui::SelectFileDialog::SELECT_SAVEAS_FILE, |
| base::string16(), |
| audio_debug_recordings_file_path_, |
| NULL, |
| 0, |
| FILE_PATH_LITERAL(""), |
| web_contents->GetTopLevelNativeWindow(), |
| NULL); |
| #endif |
| #endif |
| } |
| |
| void WebRTCInternals::DisableAudioDebugRecordings() { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| #if defined(ENABLE_WEBRTC) |
| audio_debug_recordings_ = false; |
| |
| // Tear down the dialog since the user has unchecked the audio debug |
| // recordings box. |
| select_file_dialog_ = NULL; |
| |
| for (RenderProcessHost::iterator i( |
| content::RenderProcessHost::AllHostsIterator()); |
| !i.IsAtEnd(); i.Advance()) { |
| i.GetCurrentValue()->DisableAudioDebugRecordings(); |
| } |
| #endif |
| } |
| |
| bool WebRTCInternals::IsAudioDebugRecordingsEnabled() const { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| return audio_debug_recordings_; |
| } |
| |
| const base::FilePath& WebRTCInternals::GetAudioDebugRecordingsFilePath() const { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| return audio_debug_recordings_file_path_; |
| } |
| |
| void WebRTCInternals::EnableEventLogRecordings( |
| content::WebContents* web_contents) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| #if defined(ENABLE_WEBRTC) |
| #if defined(OS_ANDROID) |
| EnableEventLogRecordingsOnAllRenderProcessHosts(); |
| #else |
| DCHECK(web_contents); |
| DCHECK(!select_file_dialog_); |
| selecting_event_log_ = true; |
| select_file_dialog_ = ui::SelectFileDialog::Create(this, nullptr); |
| select_file_dialog_->SelectFile( |
| ui::SelectFileDialog::SELECT_SAVEAS_FILE, base::string16(), |
| event_log_recordings_file_path_, nullptr, 0, FILE_PATH_LITERAL(""), |
| web_contents->GetTopLevelNativeWindow(), nullptr); |
| #endif |
| #endif |
| } |
| |
| void WebRTCInternals::DisableEventLogRecordings() { |
| #if defined(ENABLE_WEBRTC) |
| event_log_recordings_ = false; |
| // Tear down the dialog since the user has unchecked the event log checkbox. |
| select_file_dialog_ = nullptr; |
| for (RenderProcessHost::iterator i( |
| content::RenderProcessHost::AllHostsIterator()); |
| !i.IsAtEnd(); i.Advance()) |
| i.GetCurrentValue()->StopWebRTCEventLog(); |
| #endif |
| } |
| |
| bool WebRTCInternals::IsEventLogRecordingsEnabled() const { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| return event_log_recordings_; |
| } |
| |
| const base::FilePath& WebRTCInternals::GetEventLogFilePath() const { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| return event_log_recordings_file_path_; |
| } |
| |
| void WebRTCInternals::SendUpdate(const string& command, |
| std::unique_ptr<base::Value> value) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| DCHECK(observers_.might_have_observers()); |
| |
| bool queue_was_empty = pending_updates_.empty(); |
| pending_updates_.push(PendingUpdate(command, std::move(value))); |
| |
| if (queue_was_empty) { |
| BrowserThread::PostDelayedTask(BrowserThread::UI, FROM_HERE, |
| base::Bind(&WebRTCInternals::ProcessPendingUpdates, |
| weak_factory_.GetWeakPtr()), |
| base::TimeDelta::FromMilliseconds(aggregate_updates_ms_)); |
| } |
| } |
| |
| void WebRTCInternals::RenderProcessHostDestroyed(RenderProcessHost* host) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| OnRendererExit(host->GetID()); |
| |
| render_process_id_set_.erase(host->GetID()); |
| host->RemoveObserver(this); |
| } |
| |
| void WebRTCInternals::FileSelected(const base::FilePath& path, |
| int /* unused_index */, |
| void* /*unused_params */) { |
| #if defined(ENABLE_WEBRTC) |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| if (selecting_event_log_) { |
| event_log_recordings_file_path_ = path; |
| EnableEventLogRecordingsOnAllRenderProcessHosts(); |
| } else { |
| audio_debug_recordings_file_path_ = path; |
| EnableAudioDebugRecordingsOnAllRenderProcessHosts(); |
| } |
| #endif |
| } |
| |
| void WebRTCInternals::FileSelectionCanceled(void* params) { |
| #if defined(ENABLE_WEBRTC) |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| if (selecting_event_log_) { |
| SendUpdate("eventLogRecordingsFileSelectionCancelled", nullptr); |
| } else { |
| SendUpdate("audioDebugRecordingsFileSelectionCancelled", nullptr); |
| } |
| #endif |
| } |
| |
| void WebRTCInternals::OnRendererExit(int render_process_id) { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| |
| // Iterates from the end of the list to remove the PeerConnections created |
| // by the exitting renderer. |
| for (int i = peer_connection_data_.GetSize() - 1; i >= 0; --i) { |
| base::DictionaryValue* record = NULL; |
| peer_connection_data_.GetDictionary(i, &record); |
| |
| int this_rid = 0; |
| record->GetInteger("rid", &this_rid); |
| |
| if (this_rid == render_process_id) { |
| if (observers_.might_have_observers()) { |
| int lid = 0, pid = 0; |
| record->GetInteger("lid", &lid); |
| record->GetInteger("pid", &pid); |
| |
| std::unique_ptr<base::DictionaryValue> update( |
| new base::DictionaryValue()); |
| update->SetInteger("lid", lid); |
| update->SetInteger("pid", pid); |
| SendUpdate("removePeerConnection", std::move(update)); |
| } |
| peer_connection_data_.Remove(i, NULL); |
| } |
| } |
| CreateOrReleasePowerSaveBlocker(); |
| |
| bool found_any = false; |
| // Iterates from the end of the list to remove the getUserMedia requests |
| // created by the exiting renderer. |
| for (int i = get_user_media_requests_.GetSize() - 1; i >= 0; --i) { |
| base::DictionaryValue* record = NULL; |
| get_user_media_requests_.GetDictionary(i, &record); |
| |
| int this_rid = 0; |
| record->GetInteger("rid", &this_rid); |
| |
| if (this_rid == render_process_id) { |
| get_user_media_requests_.Remove(i, NULL); |
| found_any = true; |
| } |
| } |
| |
| if (found_any && observers_.might_have_observers()) { |
| std::unique_ptr<base::DictionaryValue> update(new base::DictionaryValue()); |
| update->SetInteger("rid", render_process_id); |
| SendUpdate("removeGetUserMediaForRenderer", std::move(update)); |
| } |
| } |
| |
| #if defined(ENABLE_WEBRTC) |
| void WebRTCInternals::EnableAudioDebugRecordingsOnAllRenderProcessHosts() { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| |
| audio_debug_recordings_ = true; |
| for (RenderProcessHost::iterator i( |
| content::RenderProcessHost::AllHostsIterator()); |
| !i.IsAtEnd(); i.Advance()) { |
| i.GetCurrentValue()->EnableAudioDebugRecordings( |
| audio_debug_recordings_file_path_); |
| } |
| } |
| |
| void WebRTCInternals::EnableEventLogRecordingsOnAllRenderProcessHosts() { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| |
| event_log_recordings_ = true; |
| for (RenderProcessHost::iterator i( |
| content::RenderProcessHost::AllHostsIterator()); |
| !i.IsAtEnd(); i.Advance()) |
| i.GetCurrentValue()->StartWebRTCEventLog(event_log_recordings_file_path_); |
| } |
| #endif |
| |
| void WebRTCInternals::CreateOrReleasePowerSaveBlocker() { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| if (!should_block_power_saving_) |
| return; |
| |
| if (peer_connection_data_.empty() && power_save_blocker_) { |
| DVLOG(1) << ("Releasing the block on application suspension since no " |
| "PeerConnections are active anymore."); |
| power_save_blocker_.reset(); |
| } else if (!peer_connection_data_.empty() && !power_save_blocker_) { |
| DVLOG(1) << ("Preventing the application from being suspended while one or " |
| "more PeerConnections are active."); |
| power_save_blocker_.reset(new device::PowerSaveBlocker( |
| device::PowerSaveBlocker::kPowerSaveBlockPreventAppSuspension, |
| device::PowerSaveBlocker::kReasonOther, |
| "WebRTC has active PeerConnections", |
| BrowserThread::GetTaskRunnerForThread(BrowserThread::UI), |
| BrowserThread::GetTaskRunnerForThread(BrowserThread::FILE))); |
| } |
| } |
| |
| void WebRTCInternals::ProcessPendingUpdates() { |
| DCHECK_CURRENTLY_ON(BrowserThread::UI); |
| while (!pending_updates_.empty()) { |
| const auto& update = pending_updates_.front(); |
| FOR_EACH_OBSERVER(WebRTCInternalsUIObserver, |
| observers_, |
| OnUpdate(update.command(), update.value())); |
| pending_updates_.pop(); |
| } |
| } |
| |
| } // namespace content |