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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&); |
| 32 InternalAPMConfig& operator=(InternalAPMConfig&&) = delete; |
| 33 |
| 34 bool operator==(const InternalAPMConfig& other); |
| 35 |
| 36 bool aec_enabled = false; |
| 37 bool aec_delay_agnostic_enabled = false; |
| 38 bool aec_drift_compensation_enabled = false; |
| 39 bool aec_extended_filter_enabled = false; |
| 40 int aec_suppression_level = 0; |
| 41 bool aecm_enabled = false; |
| 42 bool aecm_comfort_noise_enabled = false; |
| 43 int aecm_routing_mode = 0; |
| 44 bool agc_enabled = false; |
| 45 int agc_mode = 0; |
| 46 bool agc_limiter_enabled = false; |
| 47 bool hpf_enabled = false; |
| 48 bool ns_enabled = false; |
| 49 int ns_level = 0; |
| 50 bool transient_suppression_enabled = false; |
| 51 bool intelligibility_enhancer_enabled = false; |
| 52 bool noise_robust_agc_enabled = false; |
| 53 std::string experiments_description = ""; |
| 54 }; |
| 55 |
| 56 struct InternalAPMStreamsConfig { |
| 57 int input_sample_rate = 0; |
| 58 int output_sample_rate = 0; |
| 59 int render_input_sample_rate = 0; |
| 60 int render_output_sample_rate = 0; |
| 61 |
| 62 size_t input_num_channels = 0; |
| 63 size_t output_num_channels = 0; |
| 64 size_t render_input_num_channels = 0; |
| 65 size_t render_output_num_channels = 0; |
| 66 }; |
| 67 |
| 68 // Class to pass audio data in float** format. This is to avoid |
| 69 // dependence on AudioBuffer, and avoid problems associated with |
| 70 // rtc::ArrayView<rtc::ArrayView>. |
| 71 class FloatAudioFrame { |
| 72 public: |
| 73 // |num_channels| and |channel_size| describe the float** |
| 74 // |audio_samples|. |audio_samples| is assumed to point to a |
| 75 // two-dimensional |num_channels * channel_size| array of floats. |
| 76 FloatAudioFrame(const float* const* audio_samples, |
| 77 size_t num_channels, |
| 78 size_t channel_size) |
| 79 : audio_samples_(audio_samples), |
| 80 num_channels_(num_channels), |
| 81 channel_size_(channel_size) {} |
| 82 |
| 83 FloatAudioFrame() = delete; |
| 84 |
| 85 size_t num_channels() const { return num_channels_; } |
| 86 |
| 87 rtc::ArrayView<const float> channel(size_t idx) const { |
| 88 RTC_DCHECK_LE(0, idx); |
| 89 RTC_DCHECK_LE(idx, num_channels_); |
| 90 return rtc::ArrayView<const float>(audio_samples_[idx], channel_size_); |
| 91 } |
| 92 |
| 93 private: |
| 94 const float* const* audio_samples_; |
| 95 size_t num_channels_; |
| 96 size_t channel_size_; |
| 97 }; |
| 98 |
| 99 // An interface for recording configuration and input/output streams |
| 100 // of the Audio Processing Module. The recordings are called |
| 101 // 'aec-dumps' and are stored in a protobuf format defined in |
| 102 // debug.proto. |
| 103 // The Write* methods are always safe to call concurrently or |
| 104 // otherwise for all implementing subclasses. The intended mode of |
| 105 // operation is to create a protobuf object from the input, and send |
| 106 // it away to be written to file asynchronously. |
| 107 class AecDump { |
| 108 public: |
| 109 struct AudioProcessingState { |
| 110 int delay; |
| 111 int drift; |
| 112 int level; |
| 113 bool keypress; |
| 114 }; |
| 115 |
| 116 virtual ~AecDump() = default; |
| 117 |
| 118 // Logs Event::Type INIT message. |
| 119 virtual void WriteInitMessage( |
| 120 const InternalAPMStreamsConfig& streams_config) = 0; |
| 121 |
| 122 // Logs Event::Type STREAM message. To log an input/output pair, |
| 123 // call the AddCapture* and AddAudioProcessingState methods followed |
| 124 // by a WriteCaptureStreamMessage call. |
| 125 virtual void AddCaptureStreamInput(const FloatAudioFrame& src) = 0; |
| 126 virtual void AddCaptureStreamOutput(const FloatAudioFrame& src) = 0; |
| 127 virtual void AddCaptureStreamInput(const AudioFrame& frame) = 0; |
| 128 virtual void AddCaptureStreamOutput(const AudioFrame& frame) = 0; |
| 129 virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0; |
| 130 virtual void WriteCaptureStreamMessage() = 0; |
| 131 |
| 132 // Logs Event::Type REVERSE_STREAM message. |
| 133 virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0; |
| 134 virtual void WriteRenderStreamMessage(const FloatAudioFrame& src) = 0; |
| 135 |
| 136 // Logs Event::Type CONFIG message. |
| 137 virtual void WriteConfig(const InternalAPMConfig& config) = 0; |
| 138 }; |
| 139 } // namespace webrtc |
| 140 |
| 141 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_ |
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