blob: fdedf6cd9d3ee332bee31ebcef77c341c82a1253 [file] [log] [blame]
/*
* 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/base/checks.h"
#include "webrtc/base/ignore_wundef.h"
#include "webrtc/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;
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
}
std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {
return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle));
}
} // namespace webrtc