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| 1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #include "chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_pr
ivate_api.h" | 5 #include "chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_pr
ivate_api.h" |
| 6 | 6 |
| 7 #include "base/command_line.h" | 7 #include "base/command_line.h" |
| 8 #include "base/hash.h" | 8 #include "base/hash.h" |
| 9 #include "base/logging.h" | 9 #include "base/logging.h" |
| 10 #include "base/strings/string_number_conversions.h" | 10 #include "base/strings/string_number_conversions.h" |
| (...skipping 24 matching lines...) Expand all Loading... |
| 35 namespace StartRtpDump = api::webrtc_logging_private::StartRtpDump; | 35 namespace StartRtpDump = api::webrtc_logging_private::StartRtpDump; |
| 36 namespace Stop = api::webrtc_logging_private::Stop; | 36 namespace Stop = api::webrtc_logging_private::Stop; |
| 37 namespace StopRtpDump = api::webrtc_logging_private::StopRtpDump; | 37 namespace StopRtpDump = api::webrtc_logging_private::StopRtpDump; |
| 38 namespace Store = api::webrtc_logging_private::Store; | 38 namespace Store = api::webrtc_logging_private::Store; |
| 39 namespace Upload = api::webrtc_logging_private::Upload; | 39 namespace Upload = api::webrtc_logging_private::Upload; |
| 40 namespace UploadStored = api::webrtc_logging_private::UploadStored; | 40 namespace UploadStored = api::webrtc_logging_private::UploadStored; |
| 41 namespace StartAudioDebugRecordings = | 41 namespace StartAudioDebugRecordings = |
| 42 api::webrtc_logging_private::StartAudioDebugRecordings; | 42 api::webrtc_logging_private::StartAudioDebugRecordings; |
| 43 namespace StopAudioDebugRecordings = | 43 namespace StopAudioDebugRecordings = |
| 44 api::webrtc_logging_private::StopAudioDebugRecordings; | 44 api::webrtc_logging_private::StopAudioDebugRecordings; |
| 45 namespace StartWebRtcEventLogging = |
| 46 api::webrtc_logging_private::StartWebRtcEventLogging; |
| 47 namespace StopWebRtcEventLogging = |
| 48 api::webrtc_logging_private::StopWebRtcEventLogging; |
| 45 | 49 |
| 46 namespace { | 50 namespace { |
| 47 std::string HashIdWithOrigin(const std::string& security_origin, | 51 std::string HashIdWithOrigin(const std::string& security_origin, |
| 48 const std::string& log_id) { | 52 const std::string& log_id) { |
| 49 return base::UintToString(base::Hash(security_origin + log_id)); | 53 return base::UintToString(base::Hash(security_origin + log_id)); |
| 50 } | 54 } |
| 51 } // namespace | 55 } // namespace |
| 52 | 56 |
| 53 content::RenderProcessHost* WebrtcLoggingPrivateFunction::RphFromRequest( | 57 content::RenderProcessHost* WebrtcLoggingPrivateFunction::RphFromRequest( |
| 54 const RequestInfo& request, const std::string& security_origin) { | 58 const RequestInfo& request, const std::string& security_origin) { |
| (...skipping 34 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 89 } | 93 } |
| 90 | 94 |
| 91 scoped_refptr<WebRtcLoggingHandlerHost> | 95 scoped_refptr<WebRtcLoggingHandlerHost> |
| 92 WebrtcLoggingPrivateFunction::LoggingHandlerFromRequest( | 96 WebrtcLoggingPrivateFunction::LoggingHandlerFromRequest( |
| 93 const api::webrtc_logging_private::RequestInfo& request, | 97 const api::webrtc_logging_private::RequestInfo& request, |
| 94 const std::string& security_origin) { | 98 const std::string& security_origin) { |
| 95 content::RenderProcessHost* host = RphFromRequest(request, security_origin); | 99 content::RenderProcessHost* host = RphFromRequest(request, security_origin); |
| 96 if (!host) | 100 if (!host) |
| 97 return nullptr; | 101 return nullptr; |
| 98 | 102 |
| 99 return base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host); | 103 return base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get( |
| 104 host, WebRtcLoggingHandlerHost::kWebRtcLoggingHandlerHostKey); |
| 100 } | 105 } |
| 101 | 106 |
| 102 scoped_refptr<WebRtcLoggingHandlerHost> | 107 scoped_refptr<WebRtcLoggingHandlerHost> |
| 103 WebrtcLoggingPrivateFunctionWithGenericCallback::PrepareTask( | 108 WebrtcLoggingPrivateFunctionWithGenericCallback::PrepareTask( |
| 104 const RequestInfo& request, const std::string& security_origin, | 109 const RequestInfo& request, const std::string& security_origin, |
| 105 WebRtcLoggingHandlerHost::GenericDoneCallback* callback) { | 110 WebRtcLoggingHandlerHost::GenericDoneCallback* callback) { |
| 106 *callback = base::Bind( | 111 *callback = base::Bind( |
| 107 &WebrtcLoggingPrivateFunctionWithGenericCallback::FireCallback, this); | 112 &WebrtcLoggingPrivateFunctionWithGenericCallback::FireCallback, this); |
| 108 return LoggingHandlerFromRequest(request, security_origin); | 113 return LoggingHandlerFromRequest(request, security_origin); |
| 109 } | 114 } |
| (...skipping 13 matching lines...) Expand all Loading... |
| 123 if (success) { | 128 if (success) { |
| 124 api::webrtc_logging_private::UploadResult result; | 129 api::webrtc_logging_private::UploadResult result; |
| 125 result.report_id = report_id; | 130 result.report_id = report_id; |
| 126 SetResult(result.ToValue().release()); | 131 SetResult(result.ToValue().release()); |
| 127 } else { | 132 } else { |
| 128 SetError(error_message); | 133 SetError(error_message); |
| 129 } | 134 } |
| 130 SendResponse(success); | 135 SendResponse(success); |
| 131 } | 136 } |
| 132 | 137 |
| 133 void WebrtcLoggingPrivateFunctionWithAudioDebugRecordingsCallback:: | 138 void WebrtcLoggingPrivateFunctionWithRecordingDoneCallback::FireErrorCallback( |
| 134 FireErrorCallback(const std::string& error_message) { | 139 const std::string& error_message) { |
| 135 DCHECK_CURRENTLY_ON(content::BrowserThread::UI); | 140 DCHECK_CURRENTLY_ON(content::BrowserThread::UI); |
| 136 SetError(error_message); | 141 SetError(error_message); |
| 137 SendResponse(false); | 142 SendResponse(false); |
| 138 } | 143 } |
| 139 | 144 |
| 140 void WebrtcLoggingPrivateFunctionWithAudioDebugRecordingsCallback::FireCallback( | 145 void WebrtcLoggingPrivateFunctionWithRecordingDoneCallback::FireCallback( |
| 141 const std::string& prefix_path, | 146 const std::string& prefix_path, |
| 142 bool did_stop, | 147 bool did_stop, |
| 143 bool did_manual_stop) { | 148 bool did_manual_stop) { |
| 144 DCHECK_CURRENTLY_ON(content::BrowserThread::UI); | 149 DCHECK_CURRENTLY_ON(content::BrowserThread::UI); |
| 145 api::webrtc_logging_private::AudioDebugRecordingsInfo result; | 150 api::webrtc_logging_private::RecordingInfo result; |
| 146 result.prefix_path = prefix_path; | 151 result.prefix_path = prefix_path; |
| 147 result.did_stop = did_stop; | 152 result.did_stop = did_stop; |
| 148 result.did_manual_stop = did_manual_stop; | 153 result.did_manual_stop = did_manual_stop; |
| 149 SetResult(result.ToValue().release()); | 154 SetResult(result.ToValue().release()); |
| 150 SendResponse(true); | 155 SendResponse(true); |
| 151 } | 156 } |
| 152 | 157 |
| 153 bool WebrtcLoggingPrivateSetMetaDataFunction::RunAsync() { | 158 bool WebrtcLoggingPrivateSetMetaDataFunction::RunAsync() { |
| 154 std::unique_ptr<SetMetaData::Params> params( | 159 std::unique_ptr<SetMetaData::Params> params( |
| 155 SetMetaData::Params::Create(*args_)); | 160 SetMetaData::Params::Create(*args_)); |
| (...skipping 158 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 314 (params->incoming && params->outgoing) | 319 (params->incoming && params->outgoing) |
| 315 ? RTP_DUMP_BOTH | 320 ? RTP_DUMP_BOTH |
| 316 : (params->incoming ? RTP_DUMP_INCOMING : RTP_DUMP_OUTGOING); | 321 : (params->incoming ? RTP_DUMP_INCOMING : RTP_DUMP_OUTGOING); |
| 317 | 322 |
| 318 content::RenderProcessHost* host = | 323 content::RenderProcessHost* host = |
| 319 RphFromRequest(params->request, params->security_origin); | 324 RphFromRequest(params->request, params->security_origin); |
| 320 if (!host) | 325 if (!host) |
| 321 return false; | 326 return false; |
| 322 | 327 |
| 323 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( | 328 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( |
| 324 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host)); | 329 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get( |
| 330 host, WebRtcLoggingHandlerHost::kWebRtcLoggingHandlerHostKey)); |
| 325 | 331 |
| 326 WebRtcLoggingHandlerHost::GenericDoneCallback callback = base::Bind( | 332 WebRtcLoggingHandlerHost::GenericDoneCallback callback = base::Bind( |
| 327 &WebrtcLoggingPrivateStartRtpDumpFunction::FireCallback, this); | 333 &WebrtcLoggingPrivateStartRtpDumpFunction::FireCallback, this); |
| 328 | 334 |
| 329 // This call cannot fail. | 335 // This call cannot fail. |
| 330 content::RenderProcessHost::WebRtcStopRtpDumpCallback stop_callback = | 336 content::RenderProcessHost::WebRtcStopRtpDumpCallback stop_callback = |
| 331 host->StartRtpDump(params->incoming, | 337 host->StartRtpDump(params->incoming, |
| 332 params->outgoing, | 338 params->outgoing, |
| 333 base::Bind(&WebRtcLoggingHandlerHost::OnRtpPacket, | 339 base::Bind(&WebRtcLoggingHandlerHost::OnRtpPacket, |
| 334 webrtc_logging_handler_host)); | 340 webrtc_logging_handler_host)); |
| (...skipping 22 matching lines...) Expand all Loading... |
| 357 (params->incoming && params->outgoing) | 363 (params->incoming && params->outgoing) |
| 358 ? RTP_DUMP_BOTH | 364 ? RTP_DUMP_BOTH |
| 359 : (params->incoming ? RTP_DUMP_INCOMING : RTP_DUMP_OUTGOING); | 365 : (params->incoming ? RTP_DUMP_INCOMING : RTP_DUMP_OUTGOING); |
| 360 | 366 |
| 361 content::RenderProcessHost* host = | 367 content::RenderProcessHost* host = |
| 362 RphFromRequest(params->request, params->security_origin); | 368 RphFromRequest(params->request, params->security_origin); |
| 363 if (!host) | 369 if (!host) |
| 364 return false; | 370 return false; |
| 365 | 371 |
| 366 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( | 372 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( |
| 367 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host)); | 373 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get( |
| 374 host, WebRtcLoggingHandlerHost::kWebRtcLoggingHandlerHostKey)); |
| 368 | 375 |
| 369 WebRtcLoggingHandlerHost::GenericDoneCallback callback = base::Bind( | 376 WebRtcLoggingHandlerHost::GenericDoneCallback callback = base::Bind( |
| 370 &WebrtcLoggingPrivateStopRtpDumpFunction::FireCallback, this); | 377 &WebrtcLoggingPrivateStopRtpDumpFunction::FireCallback, this); |
| 371 | 378 |
| 372 BrowserThread::PostTask(BrowserThread::IO, | 379 BrowserThread::PostTask(BrowserThread::IO, |
| 373 FROM_HERE, | 380 FROM_HERE, |
| 374 base::Bind(&WebRtcLoggingHandlerHost::StopRtpDump, | 381 base::Bind(&WebRtcLoggingHandlerHost::StopRtpDump, |
| 375 webrtc_logging_handler_host, | 382 webrtc_logging_handler_host, |
| 376 type, | 383 type, |
| 377 callback)); | 384 callback)); |
| (...skipping 13 matching lines...) Expand all Loading... |
| 391 if (params->seconds < 0) { | 398 if (params->seconds < 0) { |
| 392 FireErrorCallback("seconds must be greater than or equal to 0"); | 399 FireErrorCallback("seconds must be greater than or equal to 0"); |
| 393 return true; | 400 return true; |
| 394 } | 401 } |
| 395 | 402 |
| 396 content::RenderProcessHost* host = | 403 content::RenderProcessHost* host = |
| 397 RphFromRequest(params->request, params->security_origin); | 404 RphFromRequest(params->request, params->security_origin); |
| 398 if (!host) | 405 if (!host) |
| 399 return false; | 406 return false; |
| 400 | 407 |
| 401 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( | 408 scoped_refptr<AudioDebugRecordingsHandler> audio_debug_recordings_handler( |
| 402 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host)); | 409 base::UserDataAdapter<AudioDebugRecordingsHandler>::Get( |
| 410 host, AudioDebugRecordingsHandler::kAudioDebugRecordingsHandlerKey)); |
| 403 | 411 |
| 404 webrtc_logging_handler_host->StartAudioDebugRecordings( | 412 audio_debug_recordings_handler->StartAudioDebugRecordings( |
| 405 host, base::TimeDelta::FromSeconds(params->seconds), | 413 host, base::TimeDelta::FromSeconds(params->seconds), |
| 406 base::Bind( | 414 base::Bind( |
| 407 &WebrtcLoggingPrivateStartAudioDebugRecordingsFunction::FireCallback, | 415 &WebrtcLoggingPrivateStartAudioDebugRecordingsFunction::FireCallback, |
| 408 this), | 416 this), |
| 409 base::Bind(&WebrtcLoggingPrivateStartAudioDebugRecordingsFunction:: | 417 base::Bind(&WebrtcLoggingPrivateStartAudioDebugRecordingsFunction:: |
| 410 FireErrorCallback, | 418 FireErrorCallback, |
| 411 this)); | 419 this)); |
| 412 return true; | 420 return true; |
| 413 } | 421 } |
| 414 | 422 |
| 415 bool WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::RunAsync() { | 423 bool WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::RunAsync() { |
| 416 if (!base::CommandLine::ForCurrentProcess()->HasSwitch( | 424 if (!base::CommandLine::ForCurrentProcess()->HasSwitch( |
| 417 switches::kEnableAudioDebugRecordingsFromExtension)) { | 425 switches::kEnableAudioDebugRecordingsFromExtension)) { |
| 418 return false; | 426 return false; |
| 419 } | 427 } |
| 420 | 428 |
| 421 std::unique_ptr<StopAudioDebugRecordings::Params> params( | 429 std::unique_ptr<StopAudioDebugRecordings::Params> params( |
| 422 StopAudioDebugRecordings::Params::Create(*args_)); | 430 StopAudioDebugRecordings::Params::Create(*args_)); |
| 423 EXTENSION_FUNCTION_VALIDATE(params.get()); | 431 EXTENSION_FUNCTION_VALIDATE(params.get()); |
| 424 | 432 |
| 425 content::RenderProcessHost* host = | 433 content::RenderProcessHost* host = |
| 426 RphFromRequest(params->request, params->security_origin); | 434 RphFromRequest(params->request, params->security_origin); |
| 427 if (!host) | 435 if (!host) |
| 428 return false; | 436 return false; |
| 429 | 437 |
| 430 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( | 438 scoped_refptr<AudioDebugRecordingsHandler> audio_debug_recordings_handler( |
| 431 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host)); | 439 base::UserDataAdapter<AudioDebugRecordingsHandler>::Get( |
| 440 host, AudioDebugRecordingsHandler::kAudioDebugRecordingsHandlerKey)); |
| 432 | 441 |
| 433 webrtc_logging_handler_host->StopAudioDebugRecordings( | 442 audio_debug_recordings_handler->StopAudioDebugRecordings( |
| 434 host, | 443 host, |
| 435 base::Bind( | 444 base::Bind( |
| 436 &WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::FireCallback, | 445 &WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::FireCallback, |
| 437 this), | 446 this), |
| 438 base::Bind(&WebrtcLoggingPrivateStopAudioDebugRecordingsFunction:: | 447 base::Bind(&WebrtcLoggingPrivateStopAudioDebugRecordingsFunction:: |
| 439 FireErrorCallback, | 448 FireErrorCallback, |
| 440 this)); | 449 this)); |
| 441 return true; | 450 return true; |
| 442 } | 451 } |
| 443 | 452 |
| 453 bool WebrtcLoggingPrivateStartWebRtcEventLoggingFunction::RunAsync() { |
| 454 if (!base::CommandLine::ForCurrentProcess()->HasSwitch( |
| 455 switches::kEnableWebRtcEventLoggingFromExtension)) { |
| 456 return false; |
| 457 } |
| 458 |
| 459 scoped_ptr<StartWebRtcEventLogging::Params> params( |
| 460 StartWebRtcEventLogging::Params::Create(*args_)); |
| 461 EXTENSION_FUNCTION_VALIDATE(params.get()); |
| 462 |
| 463 if (params->seconds < 0) { |
| 464 FireErrorCallback("seconds must be greater than or equal to 0"); |
| 465 return true; |
| 466 } |
| 467 |
| 468 content::RenderProcessHost* host = |
| 469 RphFromRequest(params->request, params->security_origin); |
| 470 if (!host) |
| 471 return false; |
| 472 |
| 473 scoped_refptr<WebRtcEventLogHandler> webrtc_event_log_handler( |
| 474 base::UserDataAdapter<WebRtcEventLogHandler>::Get( |
| 475 host, WebRtcEventLogHandler::kWebRtcEventLogHandlerKey)); |
| 476 |
| 477 webrtc_event_log_handler->StartWebRtcEventLogging( |
| 478 host, base::TimeDelta::FromSeconds(params->seconds), |
| 479 base::Bind( |
| 480 &WebrtcLoggingPrivateStartWebRtcEventLoggingFunction::FireCallback, |
| 481 this), |
| 482 base::Bind( |
| 483 &WebrtcLoggingPrivateStartWebRtcEventLoggingFunction::FireErrorCallback, |
| 484 this)); |
| 485 return true; |
| 486 } |
| 487 |
| 488 bool WebrtcLoggingPrivateStopWebRtcEventLoggingFunction::RunAsync() { |
| 489 if (!base::CommandLine::ForCurrentProcess()->HasSwitch( |
| 490 switches::kEnableWebRtcEventLoggingFromExtension)) { |
| 491 return false; |
| 492 } |
| 493 |
| 494 scoped_ptr<StopWebRtcEventLogging::Params> params( |
| 495 StopWebRtcEventLogging::Params::Create(*args_)); |
| 496 EXTENSION_FUNCTION_VALIDATE(params.get()); |
| 497 |
| 498 content::RenderProcessHost* host = |
| 499 RphFromRequest(params->request, params->security_origin); |
| 500 if (!host) |
| 501 return false; |
| 502 |
| 503 scoped_refptr<WebRtcEventLogHandler> webrtc_event_log_handler( |
| 504 base::UserDataAdapter<WebRtcEventLogHandler>::Get( |
| 505 host, WebRtcEventLogHandler::kWebRtcEventLogHandlerKey)); |
| 506 |
| 507 webrtc_event_log_handler->StopWebRtcEventLogging( |
| 508 host, |
| 509 base::Bind( |
| 510 &WebrtcLoggingPrivateStopWebRtcEventLoggingFunction::FireCallback, |
| 511 this), |
| 512 base::Bind( |
| 513 &WebrtcLoggingPrivateStopWebRtcEventLoggingFunction::FireErrorCallback, |
| 514 this)); |
| 515 return true; |
| 516 } |
| 517 |
| 444 } // namespace extensions | 518 } // namespace extensions |
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