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| 1 /* | |
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_ | |
| 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_ | |
| 13 | |
| 14 #include <memory> | |
| 15 #include <string> | |
| 16 #include <vector> | |
| 17 | |
| 18 #include "webrtc/base/array_view.h" | |
| 19 | |
| 20 namespace webrtc { | |
| 21 | |
| 22 class AudioFrame; | |
| 23 | |
| 24 // Struct for passing current config from APM without having to | |
| 25 // include protobuf headers. | |
| 26 struct InternalAPMConfig { | |
| 27 InternalAPMConfig(); | |
| 28 InternalAPMConfig(const InternalAPMConfig&); | |
| 29 InternalAPMConfig(InternalAPMConfig&&); | |
| 30 | |
| 31 InternalAPMConfig& operator=(const InternalAPMConfig&) = delete; | |
| 32 InternalAPMConfig& operator=(const InternalAPMConfig&&) = delete; | |
| 33 | |
| 34 bool aec_enabled = false; | |
| 35 bool aec_delay_agnostic_enabled = false; | |
| 36 bool aec_drift_compensation_enabled = false; | |
| 37 bool aec_extended_filter_enabled = false; | |
| 38 int aec_suppression_level = 0; | |
| 39 bool aecm_enabled = false; | |
| 40 bool aecm_comfort_noise_enabled = false; | |
| 41 int aecm_routing_mode = 0; | |
| 42 bool agc_enabled = false; | |
| 43 int agc_mode = 0; | |
| 44 bool agc_limiter_enabled = false; | |
| 45 bool hpf_enabled = false; | |
| 46 bool ns_enabled = false; | |
| 47 int ns_level = 0; | |
| 48 bool transient_suppression_enabled = false; | |
| 49 bool intelligibility_enhancer_enabled = false; | |
| 50 bool noise_robust_agc_enabled = false; | |
| 51 std::string experiments_description = ""; | |
| 52 }; | |
| 53 | |
| 54 struct InternalAPMStreamsConfig { | |
| 55 int input_sample_rate = 0; | |
| 56 int output_sample_rate = 0; | |
| 57 int render_input_sample_rate = 0; | |
| 58 int render_output_sample_rate = 0; | |
| 59 | |
| 60 size_t input_num_channels = 0; | |
| 61 size_t output_num_channels = 0; | |
| 62 size_t render_input_num_channels = 0; | |
| 63 size_t render_output_num_channels = 0; | |
| 64 }; | |
| 65 | |
| 66 // Class to pass audio data in float** format. This is to avoid | |
| 67 // dependence on AudioBuffer, and avoid problems associated with | |
| 68 // rtc::ArrayView<rtc::ArrayView>. | |
| 69 class FloatAudioFrame { | |
| 70 public: | |
| 71 FloatAudioFrame(const float* const* audio_samples, | |
| 72 size_t num_channels, | |
| 73 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.
| |
| 74 : audio_samples_(audio_samples), | |
| 75 num_channels_(num_channels), | |
| 76 channel_size_(channel_size) {} | |
| 77 | |
| 78 size_t num_channels() const { return num_channels_; } | |
| 79 | |
| 80 rtc::ArrayView<const float> channel(size_t idx) const { | |
| 81 RTC_DCHECK_LE(0, idx); | |
| 82 RTC_DCHECK_LE(idx, num_channels_); | |
| 83 return rtc::ArrayView<const float>(audio_samples_[idx], channel_size_); | |
| 84 } | |
| 85 | |
| 86 private: | |
| 87 const float* const* audio_samples_; | |
| 88 size_t num_channels_; | |
| 89 size_t channel_size_; | |
| 90 }; | |
|
kwiberg-webrtc
2017/05/03 18:57:57
This is exactly what I had in mind when we spoke:
| |
| 91 | |
| 92 // An interface for recording configuration and input/output streams | |
| 93 // of the Audio Processing Module. The recordings are called | |
| 94 // 'aec-dumps' and are stored in a protobuf format defined in | |
| 95 // debug.proto. | |
| 96 class AecDump { | |
| 97 public: | |
| 98 // A capture stream frame is logged before and after processing in | |
| 99 // the same protobuf message. To facilitate that, a CaptureStreamInfo | |
| 100 // instance is first filled with Input, then Output. | |
| 101 // | |
| 102 // To log an input/output pair, first call | |
| 103 // AecDump::GetCaptureStreamInfo. Add the input and output to | |
| 104 // it. Then call AecDump::WriteCaptureStreamMessage. | |
| 105 class CaptureStreamInfo { | |
| 106 public: | |
| 107 virtual ~CaptureStreamInfo() = default; | |
| 108 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
| |
| 109 virtual void AddOutput(FloatAudioFrame src) = 0; | |
| 110 | |
| 111 virtual void AddInput(const AudioFrame& frame) = 0; | |
| 112 virtual void AddOutput(const AudioFrame& frame) = 0; | |
| 113 | |
| 114 virtual void set_delay(int delay) = 0; | |
| 115 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
| |
| 116 virtual void set_level(int level) = 0; | |
| 117 virtual void set_keypress(bool keypress) = 0; | |
| 118 }; | |
| 119 | |
| 120 virtual ~AecDump() = default; | |
| 121 | |
| 122 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'
| |
| 123 | |
| 124 // The Write* methods are always safe to call concurrently or | |
| 125 // otherwise for all implementing subclasses. The intended mode of | |
| 126 // operation is to create a protobuf object from the input, and send | |
| 127 // it away to be written to file asynchronously. | |
| 128 virtual void WriteInitMessage( | |
| 129 const InternalAPMStreamsConfig& streams_config) = 0; | |
| 130 | |
| 131 virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0; | |
| 132 | |
| 133 virtual void WriteRenderStreamMessage(FloatAudioFrame src) = 0; | |
| 134 | |
| 135 virtual void WriteCaptureStreamMessage( | |
| 136 std::unique_ptr<CaptureStreamInfo> stream_info) = 0; | |
| 137 | |
| 138 // If not |forced|, only writes the current config if it is | |
| 139 // different from the last saved one; if |forced|, writes the config | |
| 140 // regardless of the last saved. | |
| 141 virtual void WriteConfig(const InternalAPMConfig& config, bool forced) = 0; | |
| 142 }; | |
| 143 } // namespace webrtc | |
| 144 | |
| 145 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_ | |
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