diff options
Diffstat (limited to '')
14 files changed, 116 insertions, 79 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/BUILD.gn index 6aca7dee46..817a3515b0 100644 --- a/third_party/libwebrtc/modules/audio_processing/BUILD.gn +++ b/third_party/libwebrtc/modules/audio_processing/BUILD.gn @@ -34,6 +34,7 @@ rtc_library("api") { "../../api/audio:aec3_config", "../../api/audio:audio_frame_api", "../../api/audio:echo_control", + "../../api/task_queue", "../../rtc_base:macromagic", "../../rtc_base:refcount", "../../rtc_base:stringutils", @@ -43,6 +44,7 @@ rtc_library("api") { "agc:gain_control_interface", ] absl_deps = [ + "//third_party/abseil-cpp/absl/base:nullability", "//third_party/abseil-cpp/absl/strings", "//third_party/abseil-cpp/absl/types:optional", ] @@ -185,6 +187,7 @@ rtc_library("audio_processing") { "../../api/audio:aec3_config", "../../api/audio:audio_frame_api", "../../api/audio:echo_control", + "../../api/task_queue", "../../audio/utility:audio_frame_operations", "../../common_audio:common_audio_c", "../../common_audio/third_party/ooura:fft_size_256", @@ -217,6 +220,7 @@ rtc_library("audio_processing") { "vad", ] absl_deps = [ + "//third_party/abseil-cpp/absl/base:nullability", "//third_party/abseil-cpp/absl/strings", "//third_party/abseil-cpp/absl/types:optional", ] @@ -337,9 +341,13 @@ if (rtc_include_tests) { ":audio_buffer", ":audio_processing", ":audio_processing_statistics", + "../../api/task_queue", "../../test:test_support", ] - absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] + absl_deps = [ + "//third_party/abseil-cpp/absl/base:nullability", + "//third_party/abseil-cpp/absl/strings", + ] } if (!build_with_chromium) { diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn index 78bae56835..5193e28dff 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn @@ -14,10 +14,14 @@ rtc_source_set("aec_dump") { deps = [ "..:aec_dump_interface", + "../../../api/task_queue", "../../../rtc_base/system:file_wrapper", "../../../rtc_base/system:rtc_export", ] - absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] + absl_deps = [ + "//third_party/abseil-cpp/absl/base:nullability", + "//third_party/abseil-cpp/absl/strings", + ] } if (rtc_include_tests) { @@ -71,11 +75,13 @@ if (rtc_enable_protobuf) { "../../../rtc_base:protobuf_utils", "../../../rtc_base:race_checker", "../../../rtc_base:rtc_event", - "../../../rtc_base:rtc_task_queue", "../../../rtc_base/system:file_wrapper", "../../../system_wrappers", ] - absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] + absl_deps = [ + "//third_party/abseil-cpp/absl/base:nullability", + "//third_party/abseil-cpp/absl/strings", + ] deps += [ "../:audioproc_debug_proto" ] } @@ -106,6 +112,10 @@ rtc_library("null_aec_dump_factory") { deps = [ ":aec_dump", "..:aec_dump_interface", + "../../../api/task_queue", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/base:nullability", + "//third_party/abseil-cpp/absl/strings", ] - absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] } diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h index 20718c3d7f..0d258a9ebc 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h @@ -13,34 +13,34 @@ #include <memory> +#include "absl/base/nullability.h" #include "absl/strings/string_view.h" +#include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/include/aec_dump.h" #include "rtc_base/system/file_wrapper.h" #include "rtc_base/system/rtc_export.h" -namespace rtc { -class TaskQueue; -} // namespace rtc - namespace webrtc { class RTC_EXPORT AecDumpFactory { public: - // The `worker_queue` may not be null and must outlive the created - // AecDump instance. `max_log_size_bytes == -1` means the log size - // will be unlimited. `handle` may not be null. The AecDump takes - // responsibility for `handle` and closes it in the destructor. A - // non-null return value indicates that the file has been + // The `worker_queue` must outlive the created AecDump instance. + // `max_log_size_bytes == -1` means the log size will be unlimited. + // The AecDump takes responsibility for `handle` and closes it in the + // destructor. A non-null return value indicates that the file has been // sucessfully opened. - static std::unique_ptr<AecDump> Create(webrtc::FileWrapper file, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue); - static std::unique_ptr<AecDump> Create(absl::string_view file_name, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue); - static std::unique_ptr<AecDump> Create(FILE* handle, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue); + static absl::Nullable<std::unique_ptr<AecDump>> Create( + FileWrapper file, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue); + static absl::Nullable<std::unique_ptr<AecDump>> Create( + absl::string_view file_name, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue); + static absl::Nullable<std::unique_ptr<AecDump>> Create( + absl::Nonnull<FILE*> handle, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue); }; } // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc index 94c24048e0..8484fcc6e1 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc @@ -13,11 +13,12 @@ #include <memory> #include <utility> +#include "absl/base/nullability.h" #include "absl/strings/string_view.h" +#include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" #include "rtc_base/checks.h" #include "rtc_base/event.h" -#include "rtc_base/task_queue.h" namespace webrtc { @@ -59,7 +60,7 @@ void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config, AecDumpImpl::AecDumpImpl(FileWrapper debug_file, int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) + absl::Nonnull<TaskQueueBase*> worker_queue) : debug_file_(std::move(debug_file)), num_bytes_left_for_log_(max_log_size_bytes), worker_queue_(worker_queue) {} @@ -254,9 +255,10 @@ void AecDumpImpl::PostWriteToFileTask(std::unique_ptr<audioproc::Event> event) { }); } -std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) { +absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create( + FileWrapper file, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) { RTC_DCHECK(worker_queue); if (!file.is_open()) return nullptr; @@ -265,16 +267,18 @@ std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file, worker_queue); } -std::unique_ptr<AecDump> AecDumpFactory::Create(absl::string_view file_name, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) { +absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create( + absl::string_view file_name, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) { return Create(FileWrapper::OpenWriteOnly(file_name), max_log_size_bytes, worker_queue); } -std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) { +absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create( + absl::Nonnull<FILE*> handle, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) { return Create(FileWrapper(handle), max_log_size_bytes, worker_queue); } diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h index 429808f9af..d5af31b01e 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h @@ -15,11 +15,11 @@ #include <string> #include <vector> +#include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/aec_dump/capture_stream_info.h" #include "modules/audio_processing/include/aec_dump.h" #include "rtc_base/race_checker.h" #include "rtc_base/system/file_wrapper.h" -#include "rtc_base/task_queue.h" #include "rtc_base/thread_annotations.h" // Files generated at build-time by the protobuf compiler. @@ -39,7 +39,7 @@ class AecDumpImpl : public AecDump { // `max_log_size_bytes == -1` means the log size will be unlimited. AecDumpImpl(FileWrapper debug_file, int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue); + absl::Nonnull<TaskQueueBase*> worker_queue); AecDumpImpl(const AecDumpImpl&) = delete; AecDumpImpl& operator=(const AecDumpImpl&) = delete; ~AecDumpImpl() override; @@ -74,7 +74,7 @@ class AecDumpImpl : public AecDump { FileWrapper debug_file_; int64_t num_bytes_left_for_log_ = 0; rtc::RaceChecker race_checker_; - rtc::TaskQueue* worker_queue_; + absl::Nonnull<TaskQueueBase*> worker_queue_; CaptureStreamInfo capture_stream_info_; }; } // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc index 62f896fe14..2a8110c4fc 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc @@ -28,7 +28,7 @@ TEST(AecDumper, APICallsDoNotCrash) { { std::unique_ptr<webrtc::AecDump> aec_dump = - webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue); + webrtc::AecDumpFactory::Create(filename, -1, file_writer_queue.Get()); constexpr int kNumChannels = 1; constexpr int kNumSamplesPerChannel = 160; @@ -63,7 +63,7 @@ TEST(AecDumper, WriteToFile) { { std::unique_ptr<webrtc::AecDump> aec_dump = - webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue); + webrtc::AecDumpFactory::Create(filename, -1, file_writer_queue.Get()); constexpr int kNumChannels = 1; constexpr int kNumSamplesPerChannel = 160; diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc index 9bd9745069..63929afac4 100644 --- a/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc @@ -8,27 +8,32 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include "absl/base/nullability.h" #include "absl/strings/string_view.h" +#include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" #include "modules/audio_processing/include/aec_dump.h" namespace webrtc { -std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) { +absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create( + FileWrapper file, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) { return nullptr; } -std::unique_ptr<AecDump> AecDumpFactory::Create(absl::string_view file_name, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) { +absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create( + absl::string_view file_name, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) { return nullptr; } -std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) { +absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create( + absl::Nonnull<FILE*> handle, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) { return nullptr; } } // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc index c304453388..4ac074526c 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc @@ -18,11 +18,13 @@ #include <type_traits> #include <utility> +#include "absl/base/nullability.h" #include "absl/strings/match.h" #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/array_view.h" #include "api/audio/audio_frame.h" +#include "api/task_queue/task_queue_base.h" #include "common_audio/audio_converter.h" #include "common_audio/include/audio_util.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" @@ -2083,9 +2085,10 @@ void AudioProcessingImpl::UpdateRecommendedInputVolumeLocked() { capture_.recommended_input_volume = capture_.applied_input_volume; } -bool AudioProcessingImpl::CreateAndAttachAecDump(absl::string_view file_name, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) { +bool AudioProcessingImpl::CreateAndAttachAecDump( + absl::string_view file_name, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) { std::unique_ptr<AecDump> aec_dump = AecDumpFactory::Create(file_name, max_log_size_bytes, worker_queue); if (!aec_dump) { @@ -2096,9 +2099,10 @@ bool AudioProcessingImpl::CreateAndAttachAecDump(absl::string_view file_name, return true; } -bool AudioProcessingImpl::CreateAndAttachAecDump(FILE* handle, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) { +bool AudioProcessingImpl::CreateAndAttachAecDump( + FILE* handle, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) { std::unique_ptr<AecDump> aec_dump = AecDumpFactory::Create(handle, max_log_size_bytes, worker_queue); if (!aec_dump) { diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h index 1e058b5a32..2c0ab198db 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h @@ -19,10 +19,12 @@ #include <string> #include <vector> +#include "absl/base/nullability.h" #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/array_view.h" #include "api/function_view.h" +#include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/aec3/echo_canceller3.h" #include "modules/audio_processing/agc/agc_manager_direct.h" #include "modules/audio_processing/agc/gain_control.h" @@ -71,12 +73,14 @@ class AudioProcessingImpl : public AudioProcessing { int Initialize() override; int Initialize(const ProcessingConfig& processing_config) override; void ApplyConfig(const AudioProcessing::Config& config) override; - bool CreateAndAttachAecDump(absl::string_view file_name, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) override; - bool CreateAndAttachAecDump(FILE* handle, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) override; + bool CreateAndAttachAecDump( + absl::string_view file_name, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) override; + bool CreateAndAttachAecDump( + FILE* handle, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) override; // TODO(webrtc:5298) Deprecated variant. void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override; void DetachAecDump() override; diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc index c2bedb2da4..2d3684e9b5 100644 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc @@ -1485,8 +1485,8 @@ void ApmTest::ProcessDebugDump(absl::string_view in_filename, if (first_init) { // AttachAecDump() writes an additional init message. Don't start // recording until after the first init to avoid the extra message. - auto aec_dump = - AecDumpFactory::Create(out_filename, max_size_bytes, &worker_queue); + auto aec_dump = AecDumpFactory::Create(out_filename, max_size_bytes, + worker_queue.Get()); EXPECT_TRUE(aec_dump); apm_->AttachAecDump(std::move(aec_dump)); first_init = false; @@ -1632,7 +1632,7 @@ TEST_F(ApmTest, DebugDump) { const std::string filename = test::TempFilename(test::OutputPath(), "debug_aec"); { - auto aec_dump = AecDumpFactory::Create("", -1, &worker_queue); + auto aec_dump = AecDumpFactory::Create("", -1, worker_queue.Get()); EXPECT_FALSE(aec_dump); } @@ -1640,7 +1640,7 @@ TEST_F(ApmTest, DebugDump) { // Stopping without having started should be OK. apm_->DetachAecDump(); - auto aec_dump = AecDumpFactory::Create(filename, -1, &worker_queue); + auto aec_dump = AecDumpFactory::Create(filename, -1, worker_queue.Get()); EXPECT_TRUE(aec_dump); apm_->AttachAecDump(std::move(aec_dump)); EXPECT_EQ(apm_->kNoError, @@ -1683,7 +1683,7 @@ TEST_F(ApmTest, DebugDumpFromFileHandle) { // Stopping without having started should be OK. apm_->DetachAecDump(); - auto aec_dump = AecDumpFactory::Create(std::move(f), -1, &worker_queue); + auto aec_dump = AecDumpFactory::Create(std::move(f), -1, worker_queue.Get()); EXPECT_TRUE(aec_dump); apm_->AttachAecDump(std::move(aec_dump)); EXPECT_EQ(apm_->kNoError, diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h index e3223513af..dd484be4f1 100644 --- a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h +++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h @@ -23,6 +23,7 @@ #include <vector> +#include "absl/base/nullability.h" #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/array_view.h" @@ -30,15 +31,12 @@ #include "api/audio/echo_control.h" #include "api/ref_count.h" #include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/include/audio_processing_statistics.h" #include "rtc_base/arraysize.h" #include "rtc_base/system/file_wrapper.h" #include "rtc_base/system/rtc_export.h" -namespace rtc { -class TaskQueue; -} // namespace rtc - namespace webrtc { class AecDump; @@ -632,12 +630,14 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface { // return value of true indicates that the file has been // sucessfully opened, while a value of false indicates that // opening the file failed. - virtual bool CreateAndAttachAecDump(absl::string_view file_name, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) = 0; - virtual bool CreateAndAttachAecDump(FILE* handle, - int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue) = 0; + virtual bool CreateAndAttachAecDump( + absl::string_view file_name, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) = 0; + virtual bool CreateAndAttachAecDump( + absl::Nonnull<FILE*> handle, + int64_t max_log_size_bytes, + absl::Nonnull<TaskQueueBase*> worker_queue) = 0; // TODO(webrtc:5298) Deprecated variant. // Attaches provided webrtc::AecDump for recording debugging diff --git a/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h b/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h index 2ea1a865c3..dfe7d84e07 100644 --- a/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h +++ b/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h @@ -13,7 +13,9 @@ #include <memory> +#include "absl/base/nullability.h" #include "absl/strings/string_view.h" +#include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/include/aec_dump.h" #include "modules/audio_processing/include/audio_processing.h" #include "modules/audio_processing/include/audio_processing_statistics.h" @@ -155,13 +157,13 @@ class MockAudioProcessing : public AudioProcessing { CreateAndAttachAecDump, (absl::string_view file_name, int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue), + absl::Nonnull<TaskQueueBase*> worker_queue), (override)); MOCK_METHOD(bool, CreateAndAttachAecDump, (FILE * handle, int64_t max_log_size_bytes, - rtc::TaskQueue* worker_queue), + absl::Nonnull<TaskQueueBase*> worker_queue), (override)); MOCK_METHOD(void, AttachAecDump, (std::unique_ptr<AecDump>), (override)); MOCK_METHOD(void, DetachAecDump, (), (override)); diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc index 7bd6da0133..500005f26a 100644 --- a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc +++ b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc @@ -622,7 +622,7 @@ void AudioProcessingSimulator::ConfigureAudioProcessor() { if (settings_.aec_dump_output_filename) { ap_->AttachAecDump(AecDumpFactory::Create( - *settings_.aec_dump_output_filename, -1, &worker_queue_)); + *settings_.aec_dump_output_filename, -1, worker_queue_.Get())); } } diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc index cded5de217..0d3eefa94a 100644 --- a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc +++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc @@ -197,7 +197,7 @@ void DebugDumpGenerator::SetOutputChannels(int channels) { void DebugDumpGenerator::StartRecording() { apm_->AttachAecDump( - AecDumpFactory::Create(dump_file_name_.c_str(), -1, &worker_queue_)); + AecDumpFactory::Create(dump_file_name_.c_str(), -1, worker_queue_.Get())); } void DebugDumpGenerator::Process(size_t num_blocks) { |