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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h | 85 |
1 files changed, 85 insertions, 0 deletions
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 new file mode 100644 index 0000000000..fac3712b7a --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h @@ -0,0 +1,85 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ +#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ + +#include <memory> +#include <string> +#include <vector> + +#include "modules/audio_processing/aec_dump/capture_stream_info.h" +#include "modules/audio_processing/include/aec_dump.h" +#include "rtc_base/ignore_wundef.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. +RTC_PUSH_IGNORING_WUNDEF() +#ifdef WEBRTC_ANDROID_PLATFORM_BUILD +#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" +#else +#include "modules/audio_processing/debug.pb.h" +#endif +RTC_POP_IGNORING_WUNDEF() + +namespace webrtc { + +// Task-queue based implementation of AecDump. It is thread safe by +// relying on locks in TaskQueue. +class AecDumpImpl : public AecDump { + public: + // `max_log_size_bytes` - maximum number of bytes to write to the debug file, + // `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); + AecDumpImpl(const AecDumpImpl&) = delete; + AecDumpImpl& operator=(const AecDumpImpl&) = delete; + ~AecDumpImpl() override; + + void WriteInitMessage(const ProcessingConfig& api_format, + int64_t time_now_ms) override; + void AddCaptureStreamInput(const AudioFrameView<const float>& src) override; + void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override; + void AddCaptureStreamInput(const int16_t* const data, + int num_channels, + int samples_per_channel) override; + void AddCaptureStreamOutput(const int16_t* const data, + int num_channels, + int samples_per_channel) override; + void AddAudioProcessingState(const AudioProcessingState& state) override; + void WriteCaptureStreamMessage() override; + + void WriteRenderStreamMessage(const int16_t* const data, + int num_channels, + int samples_per_channel) override; + void WriteRenderStreamMessage( + const AudioFrameView<const float>& src) override; + + void WriteConfig(const InternalAPMConfig& config) override; + + void WriteRuntimeSetting( + const AudioProcessing::RuntimeSetting& runtime_setting) override; + + private: + void PostWriteToFileTask(std::unique_ptr<audioproc::Event> event); + + FileWrapper debug_file_; + int64_t num_bytes_left_for_log_ = 0; + rtc::RaceChecker race_checker_; + rtc::TaskQueue* worker_queue_; + CaptureStreamInfo capture_stream_info_; +}; +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ |