chromium/third_party/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc

/*
 *  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/numerics/safe_conversions.h"
#include "rtc_base/system/file_wrapper.h"

#if WEBRTC_ENABLE_PROTOBUF
#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
#endif

namespace webrtc {

#if WEBRTC_ENABLE_PROTOBUF
namespace {

EncoderRuntimeConfig;
Event;
NetworkMetrics;

void DumpEventToFile(const Event& event, FileWrapper* dump_file) { … }

}  // namespace
#endif  // WEBRTC_ENABLE_PROTOBUF

class DebugDumpWriterImpl final : public DebugDumpWriter { … };

DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle) { … }

void DebugDumpWriterImpl::DumpNetworkMetrics(
    const Controller::NetworkMetrics& metrics,
    int64_t timestamp) { … }

void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
    const AudioEncoderRuntimeConfig& config,
    int64_t timestamp) { … }

#if WEBRTC_ENABLE_PROTOBUF
void DebugDumpWriterImpl::DumpControllerManagerConfig(
    const audio_network_adaptor::config::ControllerManager&
        controller_manager_config,
    int64_t timestamp) { … }
#endif  // WEBRTC_ENABLE_PROTOBUF

std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) { … }

}  // namespace webrtc