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The `recommended_stream_analog_level()` getter is used to retrieve both the applied and the recommended input volume. This behavior is error-prone since the caller must know what is returned based on the point in the code (namely, before/after the AGC has changed the last applied input volume into a recommended level). This CL is a first step to make clarity on which input volume is handled in different parts of APM. Next in the pipeline: make `recommended_stream_analog_level()` a trivial getter that always returns the recommended level. Main changes: - When `recommended_stream_analog_level()` is called but `set_stream_analog_level()` is not called, APM logs an error and returns a fall-back volume (which should not be applied since, when `set_stream_analog_level()` is not called, no external input volume is expected to be present - When APM is used without calling the `*_stream_analog_level()` methods (e.g., when the caller does not provide any input volume), the recorded AEC dumps won't store `Stream::applied_input_level` Other changes: - Removed `AudioProcessingImpl::capture_::prev_analog_mic_level` - Removed redundant code in `GainController2` around detecting input volume changes (already done by APM) - Adapted the `audioproc_f` and `unpack_aecdump` tools - Data dumps clean-up: the applied and the recommended input volumes are now recorded in an AGC implementation agnostic way Bug: webrtc:7494, b/241923537 Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38054}
116 lines
4.2 KiB
C++
116 lines
4.2 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
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#define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
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#include <stdint.h>
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#include <string>
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#include "absl/base/attributes.h"
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#include "absl/types/optional.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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// Struct for passing current config from APM without having to
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// include protobuf headers.
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struct InternalAPMConfig {
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InternalAPMConfig();
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InternalAPMConfig(const InternalAPMConfig&);
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InternalAPMConfig(InternalAPMConfig&&);
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InternalAPMConfig& operator=(const InternalAPMConfig&);
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InternalAPMConfig& operator=(InternalAPMConfig&&) = delete;
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bool operator==(const InternalAPMConfig& other) const;
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bool aec_enabled = false;
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bool aec_delay_agnostic_enabled = false;
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bool aec_drift_compensation_enabled = false;
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bool aec_extended_filter_enabled = false;
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int aec_suppression_level = 0;
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bool aecm_enabled = false;
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bool aecm_comfort_noise_enabled = false;
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int aecm_routing_mode = 0;
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bool agc_enabled = false;
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int agc_mode = 0;
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bool agc_limiter_enabled = false;
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bool hpf_enabled = false;
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bool ns_enabled = false;
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int ns_level = 0;
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bool transient_suppression_enabled = false;
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bool noise_robust_agc_enabled = false;
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bool pre_amplifier_enabled = false;
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float pre_amplifier_fixed_gain_factor = 1.f;
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std::string experiments_description = "";
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};
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// An interface for recording configuration and input/output streams
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// of the Audio Processing Module. The recordings are called
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// 'aec-dumps' and are stored in a protobuf format defined in
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// debug.proto.
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// The Write* methods are always safe to call concurrently or
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// otherwise for all implementing subclasses. The intended mode of
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// operation is to create a protobuf object from the input, and send
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// it away to be written to file asynchronously.
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class AecDump {
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public:
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struct AudioProcessingState {
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int delay;
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int drift;
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absl::optional<int> applied_input_volume;
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bool keypress;
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};
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virtual ~AecDump() = default;
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// Logs Event::Type INIT message.
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virtual void WriteInitMessage(const ProcessingConfig& api_format,
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int64_t time_now_ms) = 0;
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ABSL_DEPRECATED("")
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void WriteInitMessage(const ProcessingConfig& api_format) {
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WriteInitMessage(api_format, 0);
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}
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// Logs Event::Type STREAM message. To log an input/output pair,
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// call the AddCapture* and AddAudioProcessingState methods followed
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// by a WriteCaptureStreamMessage call.
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virtual void AddCaptureStreamInput(
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const AudioFrameView<const float>& src) = 0;
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virtual void AddCaptureStreamOutput(
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const AudioFrameView<const float>& src) = 0;
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virtual void AddCaptureStreamInput(const int16_t* const data,
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int num_channels,
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int samples_per_channel) = 0;
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virtual void AddCaptureStreamOutput(const int16_t* const data,
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int num_channels,
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int samples_per_channel) = 0;
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virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
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virtual void WriteCaptureStreamMessage() = 0;
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// Logs Event::Type REVERSE_STREAM message.
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virtual void WriteRenderStreamMessage(const int16_t* const data,
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int num_channels,
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int samples_per_channel) = 0;
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virtual void WriteRenderStreamMessage(
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const AudioFrameView<const float>& src) = 0;
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virtual void WriteRuntimeSetting(
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const AudioProcessing::RuntimeSetting& runtime_setting) = 0;
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// Logs Event::Type CONFIG message.
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virtual void WriteConfig(const InternalAPMConfig& config) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
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