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e36e8bbf6d
The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
69 lines
2.4 KiB
C++
69 lines
2.4 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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CaptureStreamInfo::CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task)
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: task_(std::move(task)) {
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RTC_DCHECK(task_);
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task_->GetEvent()->set_type(audioproc::Event::STREAM);
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}
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CaptureStreamInfo::~CaptureStreamInfo() = default;
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void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
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RTC_DCHECK(task_);
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auto* stream = task_->GetEvent()->mutable_stream();
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for (size_t 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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RTC_DCHECK(task_);
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auto* stream = task_->GetEvent()->mutable_stream();
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for (size_t 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 AudioFrame& frame) {
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RTC_DCHECK(task_);
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auto* stream = task_->GetEvent()->mutable_stream();
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const size_t data_size =
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sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
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stream->set_input_data(frame.data(), data_size);
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}
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void CaptureStreamInfo::AddOutput(const AudioFrame& frame) {
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RTC_DCHECK(task_);
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auto* stream = task_->GetEvent()->mutable_stream();
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const size_t data_size =
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sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
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stream->set_output_data(frame.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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RTC_DCHECK(task_);
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auto* stream = task_->GetEvent()->mutable_stream();
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stream->set_delay(state.delay);
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stream->set_drift(state.drift);
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stream->set_level(state.level);
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stream->set_keypress(state.keypress);
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}
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} // namespace webrtc
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