| /* |
| * 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/debug_dump_writer.h" |
| |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/ignore_wundef.h" |
| #include "webrtc/rtc_base/protobuf_utils.h" |
| |
| #if WEBRTC_ENABLE_PROTOBUF |
| RTC_PUSH_IGNORING_WUNDEF() |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" |
| #else |
| #include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" |
| #endif |
| RTC_POP_IGNORING_WUNDEF() |
| #endif |
| |
| namespace webrtc { |
| |
| #if WEBRTC_ENABLE_PROTOBUF |
| namespace { |
| |
| using audio_network_adaptor::debug_dump::Event; |
| using audio_network_adaptor::debug_dump::NetworkMetrics; |
| using audio_network_adaptor::debug_dump::EncoderRuntimeConfig; |
| |
| void DumpEventToFile(const Event& event, FileWrapper* dump_file) { |
| RTC_CHECK(dump_file->is_open()); |
| ProtoString dump_data; |
| event.SerializeToString(&dump_data); |
| int32_t size = event.ByteSize(); |
| dump_file->Write(&size, sizeof(size)); |
| dump_file->Write(dump_data.data(), dump_data.length()); |
| } |
| |
| } // namespace |
| #endif // WEBRTC_ENABLE_PROTOBUF |
| |
| class DebugDumpWriterImpl final : public DebugDumpWriter { |
| public: |
| explicit DebugDumpWriterImpl(FILE* file_handle); |
| ~DebugDumpWriterImpl() override = default; |
| |
| void DumpEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& config, |
| int64_t timestamp) override; |
| |
| void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics, |
| int64_t timestamp) override; |
| |
| #if WEBRTC_ENABLE_PROTOBUF |
| void DumpControllerManagerConfig( |
| const audio_network_adaptor::config::ControllerManager& |
| controller_manager_config, |
| int64_t timestamp) override; |
| #endif |
| |
| private: |
| std::unique_ptr<FileWrapper> dump_file_; |
| }; |
| |
| DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle) |
| : dump_file_(FileWrapper::Create()) { |
| #if WEBRTC_ENABLE_PROTOBUF |
| dump_file_->OpenFromFileHandle(file_handle); |
| RTC_CHECK(dump_file_->is_open()); |
| #else |
| RTC_NOTREACHED(); |
| #endif |
| } |
| |
| void DebugDumpWriterImpl::DumpNetworkMetrics( |
| const Controller::NetworkMetrics& metrics, |
| int64_t timestamp) { |
| #if WEBRTC_ENABLE_PROTOBUF |
| Event event; |
| event.set_timestamp(timestamp); |
| event.set_type(Event::NETWORK_METRICS); |
| auto dump_metrics = event.mutable_network_metrics(); |
| |
| if (metrics.uplink_bandwidth_bps) |
| dump_metrics->set_uplink_bandwidth_bps(*metrics.uplink_bandwidth_bps); |
| |
| if (metrics.uplink_packet_loss_fraction) { |
| dump_metrics->set_uplink_packet_loss_fraction( |
| *metrics.uplink_packet_loss_fraction); |
| } |
| |
| if (metrics.target_audio_bitrate_bps) { |
| dump_metrics->set_target_audio_bitrate_bps( |
| *metrics.target_audio_bitrate_bps); |
| } |
| |
| if (metrics.rtt_ms) |
| dump_metrics->set_rtt_ms(*metrics.rtt_ms); |
| |
| if (metrics.uplink_recoverable_packet_loss_fraction) { |
| dump_metrics->set_uplink_recoverable_packet_loss_fraction( |
| *metrics.uplink_recoverable_packet_loss_fraction); |
| } |
| |
| DumpEventToFile(event, dump_file_.get()); |
| #endif // WEBRTC_ENABLE_PROTOBUF |
| } |
| |
| void DebugDumpWriterImpl::DumpEncoderRuntimeConfig( |
| const AudioEncoderRuntimeConfig& config, |
| int64_t timestamp) { |
| #if WEBRTC_ENABLE_PROTOBUF |
| Event event; |
| event.set_timestamp(timestamp); |
| event.set_type(Event::ENCODER_RUNTIME_CONFIG); |
| auto dump_config = event.mutable_encoder_runtime_config(); |
| |
| if (config.bitrate_bps) |
| dump_config->set_bitrate_bps(*config.bitrate_bps); |
| |
| if (config.frame_length_ms) |
| dump_config->set_frame_length_ms(*config.frame_length_ms); |
| |
| if (config.uplink_packet_loss_fraction) { |
| dump_config->set_uplink_packet_loss_fraction( |
| *config.uplink_packet_loss_fraction); |
| } |
| |
| if (config.enable_fec) |
| dump_config->set_enable_fec(*config.enable_fec); |
| |
| if (config.enable_dtx) |
| dump_config->set_enable_dtx(*config.enable_dtx); |
| |
| if (config.num_channels) |
| dump_config->set_num_channels(*config.num_channels); |
| |
| DumpEventToFile(event, dump_file_.get()); |
| #endif // WEBRTC_ENABLE_PROTOBUF |
| } |
| |
| #if WEBRTC_ENABLE_PROTOBUF |
| void DebugDumpWriterImpl::DumpControllerManagerConfig( |
| const audio_network_adaptor::config::ControllerManager& |
| controller_manager_config, |
| int64_t timestamp) { |
| Event event; |
| event.set_timestamp(timestamp); |
| event.set_type(Event::CONTROLLER_MANAGER_CONFIG); |
| event.mutable_controller_manager_config()->CopyFrom( |
| controller_manager_config); |
| DumpEventToFile(event, dump_file_.get()); |
| } |
| #endif // WEBRTC_ENABLE_PROTOBUF |
| |
| std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) { |
| return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle)); |
| } |
| |
| } // namespace webrtc |