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/*
* 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 "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
#include <string>
#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/system/file_wrapper.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 "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::EncoderRuntimeConfig;
using audio_network_adaptor::debug_dump::Event;
using audio_network_adaptor::debug_dump::NetworkMetrics;
void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
RTC_CHECK(dump_file->is_open());
std::string dump_data;
event.SerializeToString(&dump_data);
int32_t size = rtc::checked_cast<int32_t>(event.ByteSizeLong());
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:
FileWrapper dump_file_;
};
DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle) {
#if WEBRTC_ENABLE_PROTOBUF
dump_file_ = FileWrapper(file_handle);
RTC_CHECK(dump_file_.is_open());
#else
RTC_DCHECK_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);
DumpEventToFile(event, &dump_file_);
#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_);
#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_);
}
#endif // WEBRTC_ENABLE_PROTOBUF
std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {
return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle));
}
} // namespace webrtc
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