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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}
61 lines
2.2 KiB
C++
61 lines
2.2 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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#include "modules/audio_processing/aec_dump/capture_stream_info.h"
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namespace webrtc {
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void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
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auto* stream = event_->mutable_stream();
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for (int i = 0; i < src.num_channels(); ++i) {
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const auto& channel_view = src.channel(i);
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stream->add_input_channel(channel_view.begin(),
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sizeof(float) * channel_view.size());
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}
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}
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void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
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auto* stream = event_->mutable_stream();
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for (int i = 0; i < src.num_channels(); ++i) {
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const auto& channel_view = src.channel(i);
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stream->add_output_channel(channel_view.begin(),
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sizeof(float) * channel_view.size());
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}
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}
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void CaptureStreamInfo::AddInput(const int16_t* const data,
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int num_channels,
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int samples_per_channel) {
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auto* stream = event_->mutable_stream();
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const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
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stream->set_input_data(data, data_size);
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}
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void CaptureStreamInfo::AddOutput(const int16_t* const data,
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int num_channels,
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int samples_per_channel) {
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auto* stream = event_->mutable_stream();
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const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
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stream->set_output_data(data, data_size);
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}
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void CaptureStreamInfo::AddAudioProcessingState(
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const AecDump::AudioProcessingState& state) {
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auto* stream = event_->mutable_stream();
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stream->set_delay(state.delay);
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stream->set_drift(state.drift);
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if (state.applied_input_volume.has_value()) {
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stream->set_applied_input_volume(*state.applied_input_volume);
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}
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stream->set_keypress(state.keypress);
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}
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} // namespace webrtc
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