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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 StartRtcEventLogging = | |
| 46 api::webrtc_logging_private::StartRtcEventLogging; | |
| 47 namespace StopRtcEventLogging = | |
| 48 api::webrtc_logging_private::StopRtcEventLogging; | |
| 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 WebrtcLoggingPrivateFunctionWithTimeLimitedRecordingCallback:: |
| 134 FireErrorCallback(const std::string& error_message) { | 139 FireErrorCallback(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 WebrtcLoggingPrivateFunctionWithTimeLimitedRecordingCallback::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::TimeLimitedRecordingInfo 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 scoped_ptr<SetMetaData::Params> params(SetMetaData::Params::Create(*args_)); | 159 scoped_ptr<SetMetaData::Params> params(SetMetaData::Params::Create(*args_)); |
| 155 EXTENSION_FUNCTION_VALIDATE(params.get()); | 160 EXTENSION_FUNCTION_VALIDATE(params.get()); |
| (...skipping 155 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 311 (params->incoming && params->outgoing) | 316 (params->incoming && params->outgoing) |
| 312 ? RTP_DUMP_BOTH | 317 ? RTP_DUMP_BOTH |
| 313 : (params->incoming ? RTP_DUMP_INCOMING : RTP_DUMP_OUTGOING); | 318 : (params->incoming ? RTP_DUMP_INCOMING : RTP_DUMP_OUTGOING); |
| 314 | 319 |
| 315 content::RenderProcessHost* host = | 320 content::RenderProcessHost* host = |
| 316 RphFromRequest(params->request, params->security_origin); | 321 RphFromRequest(params->request, params->security_origin); |
| 317 if (!host) | 322 if (!host) |
| 318 return false; | 323 return false; |
| 319 | 324 |
| 320 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( | 325 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( |
| 321 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host)); | 326 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get( |
| 327 host, WebRtcLoggingHandlerHost::kWebRtcLoggingHandlerHostKey)); | |
| 322 | 328 |
| 323 WebRtcLoggingHandlerHost::GenericDoneCallback callback = base::Bind( | 329 WebRtcLoggingHandlerHost::GenericDoneCallback callback = base::Bind( |
| 324 &WebrtcLoggingPrivateStartRtpDumpFunction::FireCallback, this); | 330 &WebrtcLoggingPrivateStartRtpDumpFunction::FireCallback, this); |
| 325 | 331 |
| 326 // This call cannot fail. | 332 // This call cannot fail. |
| 327 content::RenderProcessHost::WebRtcStopRtpDumpCallback stop_callback = | 333 content::RenderProcessHost::WebRtcStopRtpDumpCallback stop_callback = |
| 328 host->StartRtpDump(params->incoming, | 334 host->StartRtpDump(params->incoming, |
| 329 params->outgoing, | 335 params->outgoing, |
| 330 base::Bind(&WebRtcLoggingHandlerHost::OnRtpPacket, | 336 base::Bind(&WebRtcLoggingHandlerHost::OnRtpPacket, |
| 331 webrtc_logging_handler_host)); | 337 webrtc_logging_handler_host)); |
| (...skipping 21 matching lines...) Expand all Loading... | |
| 353 (params->incoming && params->outgoing) | 359 (params->incoming && params->outgoing) |
| 354 ? RTP_DUMP_BOTH | 360 ? RTP_DUMP_BOTH |
| 355 : (params->incoming ? RTP_DUMP_INCOMING : RTP_DUMP_OUTGOING); | 361 : (params->incoming ? RTP_DUMP_INCOMING : RTP_DUMP_OUTGOING); |
| 356 | 362 |
| 357 content::RenderProcessHost* host = | 363 content::RenderProcessHost* host = |
| 358 RphFromRequest(params->request, params->security_origin); | 364 RphFromRequest(params->request, params->security_origin); |
| 359 if (!host) | 365 if (!host) |
| 360 return false; | 366 return false; |
| 361 | 367 |
| 362 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( | 368 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( |
| 363 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host)); | 369 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get( |
| 370 host, WebRtcLoggingHandlerHost::kWebRtcLoggingHandlerHostKey)); | |
| 364 | 371 |
| 365 WebRtcLoggingHandlerHost::GenericDoneCallback callback = base::Bind( | 372 WebRtcLoggingHandlerHost::GenericDoneCallback callback = base::Bind( |
| 366 &WebrtcLoggingPrivateStopRtpDumpFunction::FireCallback, this); | 373 &WebrtcLoggingPrivateStopRtpDumpFunction::FireCallback, this); |
| 367 | 374 |
| 368 BrowserThread::PostTask(BrowserThread::IO, | 375 BrowserThread::PostTask(BrowserThread::IO, |
| 369 FROM_HERE, | 376 FROM_HERE, |
| 370 base::Bind(&WebRtcLoggingHandlerHost::StopRtpDump, | 377 base::Bind(&WebRtcLoggingHandlerHost::StopRtpDump, |
| 371 webrtc_logging_handler_host, | 378 webrtc_logging_handler_host, |
| 372 type, | 379 type, |
| 373 callback)); | 380 callback)); |
| (...skipping 13 matching lines...) Expand all Loading... | |
| 387 if (params->seconds < 0) { | 394 if (params->seconds < 0) { |
| 388 FireErrorCallback("seconds must be greater than or equal to 0"); | 395 FireErrorCallback("seconds must be greater than or equal to 0"); |
| 389 return true; | 396 return true; |
| 390 } | 397 } |
| 391 | 398 |
| 392 content::RenderProcessHost* host = | 399 content::RenderProcessHost* host = |
| 393 RphFromRequest(params->request, params->security_origin); | 400 RphFromRequest(params->request, params->security_origin); |
| 394 if (!host) | 401 if (!host) |
| 395 return false; | 402 return false; |
| 396 | 403 |
| 397 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( | 404 scoped_refptr<AudioDebugRecordingsHandler> audio_debug_recrdings_handler( |
|
Guido Urdaneta
2016/02/23 13:57:34
typo: recrdings -> recordings
terelius-chromium
2016/03/02 10:01:09
Done. Thanks!
| |
| 398 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host)); | 405 base::UserDataAdapter<AudioDebugRecordingsHandler>::Get( |
| 406 host, AudioDebugRecordingsHandler::kAudioDebugRecordingsHandlerKey)); | |
| 399 | 407 |
| 400 webrtc_logging_handler_host->StartAudioDebugRecordings( | 408 audio_debug_recrdings_handler->StartAudioDebugRecordings( |
| 401 host, base::TimeDelta::FromSeconds(params->seconds), | 409 host, base::TimeDelta::FromSeconds(params->seconds), |
| 402 base::Bind( | 410 base::Bind( |
| 403 &WebrtcLoggingPrivateStartAudioDebugRecordingsFunction::FireCallback, | 411 &WebrtcLoggingPrivateStartAudioDebugRecordingsFunction::FireCallback, |
| 404 this), | 412 this), |
| 405 base::Bind(&WebrtcLoggingPrivateStartAudioDebugRecordingsFunction:: | 413 base::Bind(&WebrtcLoggingPrivateStartAudioDebugRecordingsFunction:: |
| 406 FireErrorCallback, | 414 FireErrorCallback, |
| 407 this)); | 415 this)); |
| 408 return true; | 416 return true; |
| 409 } | 417 } |
| 410 | 418 |
| 411 bool WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::RunAsync() { | 419 bool WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::RunAsync() { |
| 412 if (!base::CommandLine::ForCurrentProcess()->HasSwitch( | 420 if (!base::CommandLine::ForCurrentProcess()->HasSwitch( |
| 413 switches::kEnableAudioDebugRecordingsFromExtension)) { | 421 switches::kEnableAudioDebugRecordingsFromExtension)) { |
| 414 return false; | 422 return false; |
| 415 } | 423 } |
| 416 | 424 |
| 417 scoped_ptr<StopAudioDebugRecordings::Params> params( | 425 scoped_ptr<StopAudioDebugRecordings::Params> params( |
| 418 StopAudioDebugRecordings::Params::Create(*args_)); | 426 StopAudioDebugRecordings::Params::Create(*args_)); |
| 419 EXTENSION_FUNCTION_VALIDATE(params.get()); | 427 EXTENSION_FUNCTION_VALIDATE(params.get()); |
| 420 | 428 |
| 421 content::RenderProcessHost* host = | 429 content::RenderProcessHost* host = |
| 422 RphFromRequest(params->request, params->security_origin); | 430 RphFromRequest(params->request, params->security_origin); |
| 423 if (!host) | 431 if (!host) |
| 424 return false; | 432 return false; |
| 425 | 433 |
| 426 scoped_refptr<WebRtcLoggingHandlerHost> webrtc_logging_handler_host( | 434 scoped_refptr<AudioDebugRecordingsHandler> audio_debug_recrdings_handler( |
| 427 base::UserDataAdapter<WebRtcLoggingHandlerHost>::Get(host, host)); | 435 base::UserDataAdapter<AudioDebugRecordingsHandler>::Get( |
| 436 host, AudioDebugRecordingsHandler::kAudioDebugRecordingsHandlerKey)); | |
| 428 | 437 |
| 429 webrtc_logging_handler_host->StopAudioDebugRecordings( | 438 audio_debug_recrdings_handler->StopAudioDebugRecordings( |
| 430 host, | 439 host, |
| 431 base::Bind( | 440 base::Bind( |
| 432 &WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::FireCallback, | 441 &WebrtcLoggingPrivateStopAudioDebugRecordingsFunction::FireCallback, |
| 433 this), | 442 this), |
| 434 base::Bind(&WebrtcLoggingPrivateStopAudioDebugRecordingsFunction:: | 443 base::Bind(&WebrtcLoggingPrivateStopAudioDebugRecordingsFunction:: |
| 435 FireErrorCallback, | 444 FireErrorCallback, |
| 436 this)); | 445 this)); |
| 437 return true; | 446 return true; |
| 438 } | 447 } |
| 439 | 448 |
| 449 bool WebrtcLoggingPrivateStartRtcEventLoggingFunction::RunAsync() { | |
| 450 if (!base::CommandLine::ForCurrentProcess()->HasSwitch( | |
| 451 switches::kEnableRtcEventLoggingFromExtension)) { | |
|
Henrik Grunell
2016/02/23 15:29:10
kEnableRtcEventLoggingFromExtension -> kEnableWebR
terelius-chromium
2016/03/02 10:01:09
Done.
| |
| 452 return false; | |
| 453 } | |
| 454 | |
| 455 scoped_ptr<StartRtcEventLogging::Params> params( | |
| 456 StartRtcEventLogging::Params::Create(*args_)); | |
| 457 EXTENSION_FUNCTION_VALIDATE(params.get()); | |
| 458 | |
| 459 if (params->seconds < 0) { | |
| 460 FireErrorCallback("seconds must be greater than or equal to 0"); | |
| 461 return true; | |
| 462 } | |
| 463 | |
| 464 content::RenderProcessHost* host = | |
| 465 RphFromRequest(params->request, params->security_origin); | |
| 466 if (!host) | |
| 467 return false; | |
| 468 | |
| 469 scoped_refptr<RtcEventLogHandler> rtc_event_log_handler( | |
| 470 base::UserDataAdapter<RtcEventLogHandler>::Get( | |
| 471 host, RtcEventLogHandler::kRtcEventLogHandlerKey)); | |
| 472 | |
| 473 rtc_event_log_handler->StartRtcEventLogging( | |
| 474 host, base::TimeDelta::FromSeconds(params->seconds), | |
| 475 base::Bind( | |
| 476 &WebrtcLoggingPrivateStartRtcEventLoggingFunction::FireCallback, | |
| 477 this), | |
| 478 base::Bind( | |
| 479 &WebrtcLoggingPrivateStartRtcEventLoggingFunction::FireErrorCallback, | |
| 480 this)); | |
| 481 return true; | |
| 482 } | |
| 483 | |
| 484 bool WebrtcLoggingPrivateStopRtcEventLoggingFunction::RunAsync() { | |
| 485 if (!base::CommandLine::ForCurrentProcess()->HasSwitch( | |
| 486 switches::kEnableRtcEventLoggingFromExtension)) { | |
| 487 return false; | |
| 488 } | |
| 489 | |
| 490 scoped_ptr<StopRtcEventLogging::Params> params( | |
| 491 StopRtcEventLogging::Params::Create(*args_)); | |
| 492 EXTENSION_FUNCTION_VALIDATE(params.get()); | |
| 493 | |
| 494 content::RenderProcessHost* host = | |
| 495 RphFromRequest(params->request, params->security_origin); | |
| 496 if (!host) | |
| 497 return false; | |
| 498 | |
| 499 scoped_refptr<RtcEventLogHandler> rtc_event_log_handler( | |
| 500 base::UserDataAdapter<RtcEventLogHandler>::Get( | |
| 501 host, RtcEventLogHandler::kRtcEventLogHandlerKey)); | |
|
Henrik Grunell
2016/02/23 15:29:10
Same here. Rtc -> WebRtc
terelius-chromium
2016/03/02 10:01:09
Done.
| |
| 502 | |
| 503 rtc_event_log_handler->StopRtcEventLogging( | |
| 504 host, | |
| 505 base::Bind(&WebrtcLoggingPrivateStopRtcEventLoggingFunction::FireCallback, | |
| 506 this), | |
| 507 base::Bind( | |
| 508 &WebrtcLoggingPrivateStopRtcEventLoggingFunction::FireErrorCallback, | |
| 509 this)); | |
| 510 return true; | |
| 511 } | |
| 512 | |
| 440 } // namespace extensions | 513 } // namespace extensions |
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