Chromium Code Reviews| Index: webrtc/modules/audio_processing/include/aec_dump.h |
| diff --git a/webrtc/modules/audio_processing/include/aec_dump.h b/webrtc/modules/audio_processing/include/aec_dump.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..298ee04df68268b435e401d06073186aebecb39d |
| --- /dev/null |
| +++ b/webrtc/modules/audio_processing/include/aec_dump.h |
| @@ -0,0 +1,145 @@ |
| +/* |
| + * 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_INCLUDE_AEC_DUMP_H_ |
| +#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_ |
| + |
| +#include <memory> |
| +#include <string> |
| +#include <vector> |
| + |
| +#include "webrtc/base/array_view.h" |
| + |
| +namespace webrtc { |
| + |
| +class AudioFrame; |
| + |
| +// Struct for passing current config from APM without having to |
| +// include protobuf headers. |
| +struct InternalAPMConfig { |
| + InternalAPMConfig(); |
| + InternalAPMConfig(const InternalAPMConfig&); |
| + InternalAPMConfig(InternalAPMConfig&&); |
| + |
| + InternalAPMConfig& operator=(const InternalAPMConfig&) = delete; |
| + InternalAPMConfig& operator=(const InternalAPMConfig&&) = delete; |
| + |
| + 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 = ""; |
| +}; |
| + |
| +struct InternalAPMStreamsConfig { |
| + int input_sample_rate = 0; |
| + int output_sample_rate = 0; |
| + int render_input_sample_rate = 0; |
| + int render_output_sample_rate = 0; |
| + |
| + size_t input_num_channels = 0; |
| + size_t output_num_channels = 0; |
| + size_t render_input_num_channels = 0; |
| + size_t render_output_num_channels = 0; |
| +}; |
| + |
| +// Class to pass audio data in float** format. This is to avoid |
| +// dependence on AudioBuffer, and avoid problems associated with |
| +// rtc::ArrayView<rtc::ArrayView>. |
| +class FloatAudioFrame { |
| + public: |
| + FloatAudioFrame(const float* const* audio_samples, |
| + size_t num_channels, |
| + size_t channel_size) |
|
kwiberg-webrtc
2017/05/03 18:57:57
These are nontrivial arguments. Document them?
aleloi
2017/05/04 08:56:40
Done.
|
| + : audio_samples_(audio_samples), |
| + num_channels_(num_channels), |
| + channel_size_(channel_size) {} |
| + |
| + size_t num_channels() const { return num_channels_; } |
| + |
| + rtc::ArrayView<const float> channel(size_t idx) const { |
| + RTC_DCHECK_LE(0, idx); |
| + RTC_DCHECK_LE(idx, num_channels_); |
| + return rtc::ArrayView<const float>(audio_samples_[idx], channel_size_); |
| + } |
| + |
| + private: |
| + const float* const* audio_samples_; |
| + size_t num_channels_; |
| + size_t channel_size_; |
| +}; |
|
kwiberg-webrtc
2017/05/03 18:57:57
This is exactly what I had in mind when we spoke:
|
| + |
| +// An interface for recording configuration and input/output streams |
| +// of the Audio Processing Module. The recordings are called |
| +// 'aec-dumps' and are stored in a protobuf format defined in |
| +// debug.proto. |
| +class AecDump { |
| + 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 |
| + // AecDump::GetCaptureStreamInfo. Add the input and output to |
| + // it. Then call AecDump::WriteCaptureStreamMessage. |
| + class CaptureStreamInfo { |
| + public: |
| + virtual ~CaptureStreamInfo() = default; |
| + virtual void AddInput(FloatAudioFrame src) = 0; |
|
peah-webrtc
2017/05/04 06:19:08
These calls are made by copying the FloatAudioFram
aleloi
2017/05/04 08:56:40
I think no copies are made here because of return
|
| + virtual void AddOutput(FloatAudioFrame 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; |
|
peah-webrtc
2017/05/04 06:19:08
I think it would make sense to combine these 4 set
aleloi
2017/05/04 08:56:40
I think a set_state(int, int, int, bool) method wo
peah-webrtc
2017/05/04 10:57:51
I don't think the style guide comment really appli
|
| + virtual void set_level(int level) = 0; |
| + virtual void set_keypress(bool keypress) = 0; |
| + }; |
| + |
| + virtual ~AecDump() = default; |
| + |
| + virtual std::unique_ptr<CaptureStreamInfo> GetCaptureStreamInfo() const = 0; |
|
peah-webrtc
2017/05/04 06:19:08
Does the GetCaptureStreamInfo() method create Capt
aleloi
2017/05/04 08:56:40
Yes, it should be called 'CreateCaptureStreamInfo'
|
| + |
| + // The Write* methods are always safe to call concurrently or |
| + // otherwise for all implementing subclasses. The intended mode of |
| + // operation is to create a protobuf object from the input, and send |
| + // it away to be written to file asynchronously. |
| + virtual void WriteInitMessage( |
| + const InternalAPMStreamsConfig& streams_config) = 0; |
| + |
| + virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0; |
| + |
| + virtual void WriteRenderStreamMessage(FloatAudioFrame 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_INCLUDE_AEC_DUMP_H_ |