| /* |
| * Copyright (c) 2016 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_TEST_DEBUG_DUMP_REPLAYER_H_ |
| #define MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_ |
| |
| #include <memory> |
| |
| #include "absl/strings/string_view.h" |
| #include "common_audio/channel_buffer.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| |
| // Generated at build-time by the protobuf compiler. |
| #include "modules/audio_processing/debug.pb.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| class DebugDumpReplayer { |
| public: |
| DebugDumpReplayer(); |
| ~DebugDumpReplayer(); |
| |
| // Set dump file |
| bool SetDumpFile(absl::string_view filename); |
| |
| // Return next event. |
| absl::optional<audioproc::Event> GetNextEvent() const; |
| |
| // Run the next event. Returns true if succeeded. |
| bool RunNextEvent(); |
| |
| const ChannelBuffer<float>* GetOutput() const; |
| StreamConfig GetOutputConfig() const; |
| |
| private: |
| // Following functions are facilities for replaying debug dumps. |
| void OnInitEvent(const audioproc::Init& msg); |
| void OnStreamEvent(const audioproc::Stream& msg); |
| void OnReverseStreamEvent(const audioproc::ReverseStream& msg); |
| void OnConfigEvent(const audioproc::Config& msg); |
| void OnRuntimeSettingEvent(const audioproc::RuntimeSetting& msg); |
| |
| void MaybeRecreateApm(const audioproc::Config& msg); |
| void ConfigureApm(const audioproc::Config& msg); |
| |
| void LoadNextMessage(); |
| |
| // Buffer for APM input/output. |
| std::unique_ptr<ChannelBuffer<float>> input_; |
| std::unique_ptr<ChannelBuffer<float>> reverse_; |
| std::unique_ptr<ChannelBuffer<float>> output_; |
| |
| rtc::scoped_refptr<AudioProcessing> apm_; |
| |
| FILE* debug_file_; |
| |
| StreamConfig input_config_; |
| StreamConfig reverse_config_; |
| StreamConfig output_config_; |
| |
| bool has_next_event_; |
| audioproc::Event next_event_; |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_ |