| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" |
| |
| #include <utility> |
| |
| namespace webrtc { |
| |
| AudioNetworkAdaptorImpl::Config::Config() = default; |
| |
| AudioNetworkAdaptorImpl::Config::~Config() = default; |
| |
| AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl( |
| const Config& config, |
| std::unique_ptr<ControllerManager> controller_manager, |
| std::unique_ptr<DebugDumpWriter> debug_dump_writer) |
| : config_(config), |
| controller_manager_(std::move(controller_manager)), |
| debug_dump_writer_(std::move(debug_dump_writer)) { |
| RTC_DCHECK(controller_manager_); |
| } |
| |
| AudioNetworkAdaptorImpl::~AudioNetworkAdaptorImpl() = default; |
| |
| void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) { |
| last_metrics_.uplink_bandwidth_bps = rtc::Optional<int>(uplink_bandwidth_bps); |
| DumpNetworkMetrics(); |
| } |
| |
| void AudioNetworkAdaptorImpl::SetUplinkPacketLossFraction( |
| float uplink_packet_loss_fraction) { |
| last_metrics_.uplink_packet_loss_fraction = |
| rtc::Optional<float>(uplink_packet_loss_fraction); |
| DumpNetworkMetrics(); |
| } |
| |
| void AudioNetworkAdaptorImpl::SetTargetAudioBitrate( |
| int target_audio_bitrate_bps) { |
| last_metrics_.target_audio_bitrate_bps = |
| rtc::Optional<int>(target_audio_bitrate_bps); |
| DumpNetworkMetrics(); |
| } |
| |
| void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) { |
| last_metrics_.rtt_ms = rtc::Optional<int>(rtt_ms); |
| DumpNetworkMetrics(); |
| } |
| |
| void AudioNetworkAdaptorImpl::SetReceiverFrameLengthRange( |
| int min_frame_length_ms, |
| int max_frame_length_ms) { |
| Controller::Constraints constraints; |
| constraints.receiver_frame_length_range = |
| rtc::Optional<Controller::Constraints::FrameLengthRange>( |
| Controller::Constraints::FrameLengthRange(min_frame_length_ms, |
| max_frame_length_ms)); |
| auto controllers = controller_manager_->GetControllers(); |
| for (auto& controller : controllers) |
| controller->SetConstraints(constraints); |
| } |
| |
| AudioNetworkAdaptor::EncoderRuntimeConfig |
| AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() { |
| EncoderRuntimeConfig config; |
| for (auto& controller : |
| controller_manager_->GetSortedControllers(last_metrics_)) |
| controller->MakeDecision(last_metrics_, &config); |
| |
| // TODO(minyue): Add debug dumping. |
| if (debug_dump_writer_) |
| debug_dump_writer_->DumpEncoderRuntimeConfig( |
| config, config_.clock->TimeInMilliseconds()); |
| |
| return config; |
| } |
| |
| void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) { |
| debug_dump_writer_ = DebugDumpWriter::Create(file_handle); |
| } |
| |
| void AudioNetworkAdaptorImpl::StopDebugDump() { |
| debug_dump_writer_.reset(nullptr); |
| } |
| |
| void AudioNetworkAdaptorImpl::DumpNetworkMetrics() { |
| if (debug_dump_writer_) |
| debug_dump_writer_->DumpNetworkMetrics(last_metrics_, |
| config_.clock->TimeInMilliseconds()); |
| } |
| |
| } // namespace webrtc |