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f0d1c03c31
The pointer-to-submodule interfaces are being removed. This CL: 1) introduces AudioProcessing::Config::GainController1 with most config, 2) adds functions to APM for setting and getting analog gain, 3) creates a temporary GainControlConfigProxy to support the transition to the new config. 4) Moves the lock references in GainControlForExperimentalAgc and GainControlImpl into the GainControlConfigProxy, as it becomes the sole AGC object with functionality exposed to the client. Bug: webrtc:9947, webrtc:9878 Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27316}
117 lines
3.5 KiB
C++
117 lines
3.5 KiB
C++
/*
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* Copyright (c) 2016 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/gain_control_for_experimental_agc.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/atomic_ops.h"
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#include "rtc_base/critical_section.h"
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namespace webrtc {
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int GainControlForExperimentalAgc::instance_counter_ = 0;
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GainControlForExperimentalAgc::GainControlForExperimentalAgc(
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GainControl* gain_control)
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: data_dumper_(
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new ApmDataDumper(rtc::AtomicOps::Increment(&instance_counter_))),
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real_gain_control_(gain_control),
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volume_(0) {}
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GainControlForExperimentalAgc::~GainControlForExperimentalAgc() = default;
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int GainControlForExperimentalAgc::Enable(bool enable) {
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return real_gain_control_->Enable(enable);
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}
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bool GainControlForExperimentalAgc::is_enabled() const {
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return real_gain_control_->is_enabled();
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}
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int GainControlForExperimentalAgc::set_stream_analog_level(int level) {
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data_dumper_->DumpRaw("experimental_gain_control_set_stream_analog_level", 1,
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&level);
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do_log_level_ = true;
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volume_ = level;
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return AudioProcessing::kNoError;
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}
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int GainControlForExperimentalAgc::stream_analog_level() const {
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if (do_log_level_) {
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data_dumper_->DumpRaw("experimental_gain_control_stream_analog_level", 1,
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&volume_);
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do_log_level_ = false;
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}
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return volume_;
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}
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int GainControlForExperimentalAgc::set_mode(Mode mode) {
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return AudioProcessing::kNoError;
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}
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GainControl::Mode GainControlForExperimentalAgc::mode() const {
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return GainControl::kAdaptiveAnalog;
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}
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int GainControlForExperimentalAgc::set_target_level_dbfs(int level) {
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return AudioProcessing::kNoError;
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}
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int GainControlForExperimentalAgc::target_level_dbfs() const {
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return real_gain_control_->target_level_dbfs();
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}
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int GainControlForExperimentalAgc::set_compression_gain_db(int gain) {
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return AudioProcessing::kNoError;
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}
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int GainControlForExperimentalAgc::compression_gain_db() const {
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return real_gain_control_->compression_gain_db();
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}
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int GainControlForExperimentalAgc::enable_limiter(bool enable) {
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return AudioProcessing::kNoError;
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}
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bool GainControlForExperimentalAgc::is_limiter_enabled() const {
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return real_gain_control_->is_limiter_enabled();
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}
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int GainControlForExperimentalAgc::set_analog_level_limits(int minimum,
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int maximum) {
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return AudioProcessing::kNoError;
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}
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int GainControlForExperimentalAgc::analog_level_minimum() const {
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return real_gain_control_->analog_level_minimum();
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}
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int GainControlForExperimentalAgc::analog_level_maximum() const {
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return real_gain_control_->analog_level_maximum();
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}
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bool GainControlForExperimentalAgc::stream_is_saturated() const {
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return real_gain_control_->stream_is_saturated();
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}
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void GainControlForExperimentalAgc::SetMicVolume(int volume) {
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volume_ = volume;
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}
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int GainControlForExperimentalAgc::GetMicVolume() {
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return volume_;
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}
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void GainControlForExperimentalAgc::Initialize() {
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data_dumper_->InitiateNewSetOfRecordings();
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}
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} // namespace webrtc
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