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Side by Side Diff: webrtc/modules/audio_processing/audio_processing_impl.cc

Issue 1541633002: Reland "Added option to specify a maximum file size when recording an AEC dump." (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix to prevent Chrome from breaking. Created 5 years ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 614 matching lines...) Expand 10 before | Expand all | Expand 10 after
625 capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest); 625 capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
626 626
627 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 627 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
628 if (debug_dump_.debug_file->Open()) { 628 if (debug_dump_.debug_file->Open()) {
629 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); 629 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
630 const size_t channel_size = 630 const size_t channel_size =
631 sizeof(float) * formats_.api_format.output_stream().num_frames(); 631 sizeof(float) * formats_.api_format.output_stream().num_frames();
632 for (int i = 0; i < formats_.api_format.output_stream().num_channels(); ++i) 632 for (int i = 0; i < formats_.api_format.output_stream().num_channels(); ++i)
633 msg->add_output_channel(dest[i], channel_size); 633 msg->add_output_channel(dest[i], channel_size);
634 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 634 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
635 &debug_dump_.num_bytes_left_for_log_,
635 &crit_debug_, &debug_dump_.capture)); 636 &crit_debug_, &debug_dump_.capture));
636 } 637 }
637 #endif 638 #endif
638 639
639 return kNoError; 640 return kNoError;
640 } 641 }
641 642
642 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) { 643 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
643 TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame"); 644 TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
644 { 645 {
(...skipping 67 matching lines...) Expand 10 before | Expand all | Expand 10 after
712 capture_.capture_audio->InterleaveTo(frame, 713 capture_.capture_audio->InterleaveTo(frame,
713 output_copy_needed(is_data_processed())); 714 output_copy_needed(is_data_processed()));
714 715
715 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 716 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
716 if (debug_dump_.debug_file->Open()) { 717 if (debug_dump_.debug_file->Open()) {
717 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); 718 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
718 const size_t data_size = 719 const size_t data_size =
719 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; 720 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
720 msg->set_output_data(frame->data_, data_size); 721 msg->set_output_data(frame->data_, data_size);
721 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 722 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
723 &debug_dump_.num_bytes_left_for_log_,
722 &crit_debug_, &debug_dump_.capture)); 724 &crit_debug_, &debug_dump_.capture));
723 } 725 }
724 #endif 726 #endif
725 727
726 return kNoError; 728 return kNoError;
727 } 729 }
728 730
729 int AudioProcessingImpl::ProcessStreamLocked() { 731 int AudioProcessingImpl::ProcessStreamLocked() {
730 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 732 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
731 if (debug_dump_.debug_file->Open()) { 733 if (debug_dump_.debug_file->Open()) {
(...skipping 147 matching lines...) Expand 10 before | Expand all | Expand 10 after
879 if (debug_dump_.debug_file->Open()) { 881 if (debug_dump_.debug_file->Open()) {
880 debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM); 882 debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
881 audioproc::ReverseStream* msg = 883 audioproc::ReverseStream* msg =
882 debug_dump_.render.event_msg->mutable_reverse_stream(); 884 debug_dump_.render.event_msg->mutable_reverse_stream();
883 const size_t channel_size = 885 const size_t channel_size =
884 sizeof(float) * formats_.api_format.reverse_input_stream().num_frames(); 886 sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
885 for (int i = 0; 887 for (int i = 0;
886 i < formats_.api_format.reverse_input_stream().num_channels(); ++i) 888 i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
887 msg->add_channel(src[i], channel_size); 889 msg->add_channel(src[i], channel_size);
888 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 890 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
891 &debug_dump_.num_bytes_left_for_log_,
889 &crit_debug_, &debug_dump_.render)); 892 &crit_debug_, &debug_dump_.render));
890 } 893 }
891 #endif 894 #endif
892 895
893 render_.render_audio->CopyFrom(src, 896 render_.render_audio->CopyFrom(src,
894 formats_.api_format.reverse_input_stream()); 897 formats_.api_format.reverse_input_stream());
895 return ProcessReverseStreamLocked(); 898 return ProcessReverseStreamLocked();
896 } 899 }
897 900
898 int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) { 901 int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
(...skipping 48 matching lines...) Expand 10 before | Expand all | Expand 10 after
947 950
948 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 951 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
949 if (debug_dump_.debug_file->Open()) { 952 if (debug_dump_.debug_file->Open()) {
950 debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM); 953 debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
951 audioproc::ReverseStream* msg = 954 audioproc::ReverseStream* msg =
952 debug_dump_.render.event_msg->mutable_reverse_stream(); 955 debug_dump_.render.event_msg->mutable_reverse_stream();
953 const size_t data_size = 956 const size_t data_size =
954 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; 957 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
955 msg->set_data(frame->data_, data_size); 958 msg->set_data(frame->data_, data_size);
956 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 959 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
960 &debug_dump_.num_bytes_left_for_log_,
957 &crit_debug_, &debug_dump_.render)); 961 &crit_debug_, &debug_dump_.render));
958 } 962 }
959 #endif 963 #endif
960 render_.render_audio->DeinterleaveFrom(frame); 964 render_.render_audio->DeinterleaveFrom(frame);
961 return ProcessReverseStreamLocked(); 965 return ProcessReverseStreamLocked();
962 } 966 }
963 967
964 int AudioProcessingImpl::ProcessReverseStreamLocked() { 968 int AudioProcessingImpl::ProcessReverseStreamLocked() {
965 AudioBuffer* ra = render_.render_audio.get(); // For brevity. 969 AudioBuffer* ra = render_.render_audio.get(); // For brevity.
966 if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz) { 970 if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz) {
(...skipping 65 matching lines...) Expand 10 before | Expand all | Expand 10 after
1032 rtc::CritScope cs(&crit_capture_); 1036 rtc::CritScope cs(&crit_capture_);
1033 capture_.delay_offset_ms = offset; 1037 capture_.delay_offset_ms = offset;
1034 } 1038 }
1035 1039
1036 int AudioProcessingImpl::delay_offset_ms() const { 1040 int AudioProcessingImpl::delay_offset_ms() const {
1037 rtc::CritScope cs(&crit_capture_); 1041 rtc::CritScope cs(&crit_capture_);
1038 return capture_.delay_offset_ms; 1042 return capture_.delay_offset_ms;
1039 } 1043 }
1040 1044
1041 int AudioProcessingImpl::StartDebugRecording( 1045 int AudioProcessingImpl::StartDebugRecording(
1042 const char filename[AudioProcessing::kMaxFilenameSize]) { 1046 const char filename[AudioProcessing::kMaxFilenameSize],
1047 int64_t max_log_size_bytes) {
1043 // Run in a single-threaded manner. 1048 // Run in a single-threaded manner.
1044 rtc::CritScope cs_render(&crit_render_); 1049 rtc::CritScope cs_render(&crit_render_);
1045 rtc::CritScope cs_capture(&crit_capture_); 1050 rtc::CritScope cs_capture(&crit_capture_);
1046 static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, ""); 1051 static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, "");
1047 1052
1048 if (filename == nullptr) { 1053 if (filename == nullptr) {
1049 return kNullPointerError; 1054 return kNullPointerError;
1050 } 1055 }
1051 1056
1052 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 1057 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
1058 debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
1053 // Stop any ongoing recording. 1059 // Stop any ongoing recording.
1054 if (debug_dump_.debug_file->Open()) { 1060 if (debug_dump_.debug_file->Open()) {
1055 if (debug_dump_.debug_file->CloseFile() == -1) { 1061 if (debug_dump_.debug_file->CloseFile() == -1) {
1056 return kFileError; 1062 return kFileError;
1057 } 1063 }
1058 } 1064 }
1059 1065
1060 if (debug_dump_.debug_file->OpenFile(filename, false) == -1) { 1066 if (debug_dump_.debug_file->OpenFile(filename, false) == -1) {
1061 debug_dump_.debug_file->CloseFile(); 1067 debug_dump_.debug_file->CloseFile();
1062 return kFileError; 1068 return kFileError;
1063 } 1069 }
1064 1070
1065 RETURN_ON_ERR(WriteConfigMessage(true)); 1071 RETURN_ON_ERR(WriteConfigMessage(true));
1066 RETURN_ON_ERR(WriteInitMessage()); 1072 RETURN_ON_ERR(WriteInitMessage());
1067 return kNoError; 1073 return kNoError;
1068 #else 1074 #else
1069 return kUnsupportedFunctionError; 1075 return kUnsupportedFunctionError;
1070 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP 1076 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
1071 } 1077 }
1072 1078
1073 int AudioProcessingImpl::StartDebugRecording(FILE* handle) { 1079 int AudioProcessingImpl::StartDebugRecording(FILE* handle,
1080 int64_t max_log_size_bytes) {
1074 // Run in a single-threaded manner. 1081 // Run in a single-threaded manner.
1075 rtc::CritScope cs_render(&crit_render_); 1082 rtc::CritScope cs_render(&crit_render_);
1076 rtc::CritScope cs_capture(&crit_capture_); 1083 rtc::CritScope cs_capture(&crit_capture_);
1077 1084
1078 if (handle == nullptr) { 1085 if (handle == nullptr) {
1079 return kNullPointerError; 1086 return kNullPointerError;
1080 } 1087 }
1081 1088
1082 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 1089 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
1090 debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
1091
1083 // Stop any ongoing recording. 1092 // Stop any ongoing recording.
1084 if (debug_dump_.debug_file->Open()) { 1093 if (debug_dump_.debug_file->Open()) {
1085 if (debug_dump_.debug_file->CloseFile() == -1) { 1094 if (debug_dump_.debug_file->CloseFile() == -1) {
1086 return kFileError; 1095 return kFileError;
1087 } 1096 }
1088 } 1097 }
1089 1098
1090 if (debug_dump_.debug_file->OpenFromFileHandle(handle, true, false) == -1) { 1099 if (debug_dump_.debug_file->OpenFromFileHandle(handle, true, false) == -1) {
1091 return kFileError; 1100 return kFileError;
1092 } 1101 }
1093 1102
1094 RETURN_ON_ERR(WriteConfigMessage(true)); 1103 RETURN_ON_ERR(WriteConfigMessage(true));
1095 RETURN_ON_ERR(WriteInitMessage()); 1104 RETURN_ON_ERR(WriteInitMessage());
1096 return kNoError; 1105 return kNoError;
1097 #else 1106 #else
1098 return kUnsupportedFunctionError; 1107 return kUnsupportedFunctionError;
1099 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP 1108 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
1100 } 1109 }
1101 1110
1102 int AudioProcessingImpl::StartDebugRecordingForPlatformFile( 1111 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
1103 rtc::PlatformFile handle) { 1112 rtc::PlatformFile handle) {
1104 // Run in a single-threaded manner. 1113 // Run in a single-threaded manner.
1105 rtc::CritScope cs_render(&crit_render_); 1114 rtc::CritScope cs_render(&crit_render_);
1106 rtc::CritScope cs_capture(&crit_capture_); 1115 rtc::CritScope cs_capture(&crit_capture_);
1107 FILE* stream = rtc::FdopenPlatformFileForWriting(handle); 1116 FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
1108 return StartDebugRecording(stream); 1117 return StartDebugRecording(stream, -1);
1109 } 1118 }
1110 1119
1111 int AudioProcessingImpl::StopDebugRecording() { 1120 int AudioProcessingImpl::StopDebugRecording() {
1112 // Run in a single-threaded manner. 1121 // Run in a single-threaded manner.
1113 rtc::CritScope cs_render(&crit_render_); 1122 rtc::CritScope cs_render(&crit_render_);
1114 rtc::CritScope cs_capture(&crit_capture_); 1123 rtc::CritScope cs_capture(&crit_capture_);
1115 1124
1116 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 1125 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
1117 // We just return if recording hasn't started. 1126 // We just return if recording hasn't started.
1118 if (debug_dump_.debug_file->Open()) { 1127 if (debug_dump_.debug_file->Open()) {
(...skipping 274 matching lines...) Expand 10 before | Expand all | Expand 10 after
1393 RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps", 1402 RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
1394 capture_.aec_system_delay_jumps, 51); 1403 capture_.aec_system_delay_jumps, 51);
1395 } 1404 }
1396 capture_.aec_system_delay_jumps = -1; 1405 capture_.aec_system_delay_jumps = -1;
1397 capture_.last_aec_system_delay_ms = 0; 1406 capture_.last_aec_system_delay_ms = 0;
1398 } 1407 }
1399 1408
1400 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 1409 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
1401 int AudioProcessingImpl::WriteMessageToDebugFile( 1410 int AudioProcessingImpl::WriteMessageToDebugFile(
1402 FileWrapper* debug_file, 1411 FileWrapper* debug_file,
1412 int64_t* filesize_limit_bytes,
1403 rtc::CriticalSection* crit_debug, 1413 rtc::CriticalSection* crit_debug,
1404 ApmDebugDumpThreadState* debug_state) { 1414 ApmDebugDumpThreadState* debug_state) {
1405 int32_t size = debug_state->event_msg->ByteSize(); 1415 int32_t size = debug_state->event_msg->ByteSize();
1406 if (size <= 0) { 1416 if (size <= 0) {
1407 return kUnspecifiedError; 1417 return kUnspecifiedError;
1408 } 1418 }
1409 #if defined(WEBRTC_ARCH_BIG_ENDIAN) 1419 #if defined(WEBRTC_ARCH_BIG_ENDIAN)
1410 // TODO(ajm): Use little-endian "on the wire". For the moment, we can be 1420 // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
1411 // pretty safe in assuming little-endian. 1421 // pretty safe in assuming little-endian.
1412 #endif 1422 #endif
1413 1423
1414 if (!debug_state->event_msg->SerializeToString(&debug_state->event_str)) { 1424 if (!debug_state->event_msg->SerializeToString(&debug_state->event_str)) {
1415 return kUnspecifiedError; 1425 return kUnspecifiedError;
1416 } 1426 }
1417 1427
1418 { 1428 {
1419 // Ensure atomic writes of the message. 1429 // Ensure atomic writes of the message.
1420 rtc::CritScope cs_capture(crit_debug); 1430 rtc::CritScope cs_debug(crit_debug);
1431
1432 RTC_DCHECK(debug_file->Open());
1433 // Update the byte counter.
1434 if (*filesize_limit_bytes >= 0) {
1435 *filesize_limit_bytes -=
1436 (sizeof(int32_t) + debug_state->event_str.length());
1437 if (*filesize_limit_bytes < 0) {
1438 // Not enough bytes are left to write this message, so stop logging.
1439 debug_file->CloseFile();
1440 return kNoError;
1441 }
1442 }
1421 // Write message preceded by its size. 1443 // Write message preceded by its size.
1422 if (!debug_file->Write(&size, sizeof(int32_t))) { 1444 if (!debug_file->Write(&size, sizeof(int32_t))) {
1423 return kFileError; 1445 return kFileError;
1424 } 1446 }
1425 if (!debug_file->Write(debug_state->event_str.data(), 1447 if (!debug_file->Write(debug_state->event_str.data(),
1426 debug_state->event_str.length())) { 1448 debug_state->event_str.length())) {
1427 return kFileError; 1449 return kFileError;
1428 } 1450 }
1429 } 1451 }
1430 1452
(...skipping 14 matching lines...) Expand all
1445 msg->set_num_reverse_channels( 1467 msg->set_num_reverse_channels(
1446 formats_.api_format.reverse_input_stream().num_channels()); 1468 formats_.api_format.reverse_input_stream().num_channels());
1447 msg->set_reverse_sample_rate( 1469 msg->set_reverse_sample_rate(
1448 formats_.api_format.reverse_input_stream().sample_rate_hz()); 1470 formats_.api_format.reverse_input_stream().sample_rate_hz());
1449 msg->set_output_sample_rate( 1471 msg->set_output_sample_rate(
1450 formats_.api_format.output_stream().sample_rate_hz()); 1472 formats_.api_format.output_stream().sample_rate_hz());
1451 // TODO(ekmeyerson): Add reverse output fields to 1473 // TODO(ekmeyerson): Add reverse output fields to
1452 // debug_dump_.capture.event_msg. 1474 // debug_dump_.capture.event_msg.
1453 1475
1454 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 1476 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
1477 &debug_dump_.num_bytes_left_for_log_,
1455 &crit_debug_, &debug_dump_.capture)); 1478 &crit_debug_, &debug_dump_.capture));
1456 return kNoError; 1479 return kNoError;
1457 } 1480 }
1458 1481
1459 int AudioProcessingImpl::WriteConfigMessage(bool forced) { 1482 int AudioProcessingImpl::WriteConfigMessage(bool forced) {
1460 audioproc::Config config; 1483 audioproc::Config config;
1461 1484
1462 config.set_aec_enabled(public_submodules_->echo_cancellation->is_enabled()); 1485 config.set_aec_enabled(public_submodules_->echo_cancellation->is_enabled());
1463 config.set_aec_delay_agnostic_enabled( 1486 config.set_aec_delay_agnostic_enabled(
1464 public_submodules_->echo_cancellation->is_delay_agnostic_enabled()); 1487 public_submodules_->echo_cancellation->is_delay_agnostic_enabled());
(...skipping 32 matching lines...) Expand 10 before | Expand all | Expand 10 after
1497 debug_dump_.capture.last_serialized_config == serialized_config) { 1520 debug_dump_.capture.last_serialized_config == serialized_config) {
1498 return kNoError; 1521 return kNoError;
1499 } 1522 }
1500 1523
1501 debug_dump_.capture.last_serialized_config = serialized_config; 1524 debug_dump_.capture.last_serialized_config = serialized_config;
1502 1525
1503 debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG); 1526 debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
1504 debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config); 1527 debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
1505 1528
1506 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 1529 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
1530 &debug_dump_.num_bytes_left_for_log_,
1507 &crit_debug_, &debug_dump_.capture)); 1531 &crit_debug_, &debug_dump_.capture));
1508 return kNoError; 1532 return kNoError;
1509 } 1533 }
1510 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP 1534 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
1511 1535
1512 } // namespace webrtc 1536 } // namespace webrtc
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