Chromium Code Reviews| Index: webrtc/modules/audio_processing/aec_dumper/aec_dumper.h |
| diff --git a/webrtc/modules/audio_processing/aec_dumper/aec_dumper.h b/webrtc/modules/audio_processing/aec_dumper/aec_dumper.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..fa645e8ad448e1fff09993808d81cb3c4daca63f |
| --- /dev/null |
| +++ b/webrtc/modules/audio_processing/aec_dumper/aec_dumper.h |
| @@ -0,0 +1,133 @@ |
| +/* |
| + * 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMPER_AEC_DUMPER_H_ |
| +#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMPER_AEC_DUMPER_H_ |
| + |
| +#include <memory> |
| +#include <string> |
| +#include <vector> |
| + |
| +#include "webrtc/base/array_view.h" |
| +#include "webrtc/modules/audio_processing/include/audio_processing.h" |
| + |
| +namespace audioproc { |
| +class Event; |
| +} // namespace audioproc |
| + |
| +namespace rtc { |
| +class TaskQueue; |
| +} // namespace rtc |
| + |
| +namespace webrtc { |
| + |
| +class AudioFrame; |
| + |
| +// Struct for passing current config from APM without having to |
| +// include protobuf headers. |
| +struct InternalAPMConfig { |
| + InternalAPMConfig(); |
| + |
| + bool aec_enabled = false; |
| + bool aec_delay_agnostic_enabled = false; |
| + bool aec_drift_compensation_enabled = false; |
| + bool aec_extended_filter_enabled = false; |
| + int aec_suppression_level = 0; |
| + bool aecm_enabled = false; |
| + bool aecm_comfort_noise_enabled = false; |
| + int aecm_routing_mode = 0; |
| + bool agc_enabled = false; |
| + int agc_mode = 0; |
| + bool agc_limiter_enabled = false; |
| + bool hpf_enabled = false; |
| + bool ns_enabled = false; |
| + int ns_level = 0; |
| + bool transient_suppression_enabled = false; |
| + bool intelligibility_enhancer_enabled = false; |
| + bool noise_robust_agc_enabled = false; |
| + std::string experiments_description = ""; |
| + |
| + private: |
| + RTC_DISALLOW_COPY_AND_ASSIGN(InternalAPMConfig); |
|
kwiberg-webrtc
2017/03/29 08:57:11
You don't have to use this macro anymore. Just do
|
| +}; |
| + |
| +class AecDumper { |
| + public: |
| + // A capture stream frame is logged before and after processing in |
| + // the same protobuf message. To facilitate that, a |
| + // CaptureStreamInfo instance is first filled with Input, then |
| + // Output. |
| + // |
| + // To log an input/output pair, first call |
| + // AecDumper::GetCaptureStreamInfo. Add the input and output to |
| + // it. Then call AecDumper::WriteCaptureStreamMessage. |
| + class CaptureStreamInfo { |
| + public: |
| + virtual ~CaptureStreamInfo() = default; |
| + virtual void AddInput(std::vector<rtc::ArrayView<const float>> src) = 0; |
| + virtual void AddOutput(std::vector<rtc::ArrayView<const float>> src) = 0; |
| + |
| + virtual void AddInput(const AudioFrame& frame) = 0; |
| + virtual void AddOutput(const AudioFrame& frame) = 0; |
| + |
| + virtual void set_delay(int delay) = 0; |
| + virtual void set_drift(int drift) = 0; |
| + virtual void set_level(int level) = 0; |
| + virtual void set_keypress(bool keypress) = 0; |
| + }; |
| + |
| + AecDumper() = default; |
|
kwiberg-webrtc
2017/03/29 08:57:11
You forgot to remove this one.
|
| + |
| + virtual ~AecDumper() = default; |
| + |
| + // TODO(aleloi): update comments to new creation scheme. |
| + // If called when a recording is active, that file is closed, and a |
| + // new file is opened. Messages waiting to be written asynchronously |
| + // to the old file may be lost. Returns true iff opening file for |
| + // writing succeeded. |
| + |
| + // Closes associated file. Messages waiting to be written to file |
| + // asynchronously may be lost. This method is safe to call when no |
| + // recording is active. A recording does not have to be closed |
| + // manually with this method; instead the AecDumper instance may be |
| + // destroyed. |
| + static std::unique_ptr<AecDumper> Create(std::string file_name, |
| + int64_t max_log_size_bytes, |
| + rtc::TaskQueue* worker_queue); |
| + static std::unique_ptr<AecDumper> Create(FILE* handle, |
| + int64_t max_log_size_bytes, |
| + rtc::TaskQueue* worker_queue); |
| + |
| + static std::unique_ptr<AecDumper> CreateNullDumper(); |
| + |
| + virtual std::unique_ptr<CaptureStreamInfo> GetCaptureStreamInfo() = 0; |
| + |
| + // The Write* methods are always safe to call. If no recording is in |
| + // progress, calls will have no effect. Messages are written to file |
| + // in a 'best effort' manner. If the AecDumper can't keep up with |
| + // the flow of messages, some will be silently dropped. |
| + virtual void WriteInitMessage(const ProcessingConfig& api_format) = 0; |
| + |
| + virtual void WriteReverseStreamMessage(const AudioFrame& frame) = 0; |
| + |
| + virtual void WriteReverseStreamMessage( |
| + std::vector<rtc::ArrayView<const float>> src) = 0; |
| + |
| + virtual void WriteCaptureStreamMessage( |
| + std::unique_ptr<CaptureStreamInfo> stream_info) = 0; |
| + |
| + // If not |forced|, only writes the current config if it is |
| + // different from the last saved one; if |forced|, writes the config |
| + // regardless of the last saved. |
| + virtual void WriteConfig(const InternalAPMConfig& config, bool forced) = 0; |
| +}; |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMPER_AEC_DUMPER_H_ |