Chromium Code Reviews| Index: webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc |
| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..6e9173019b650138a30f76fd8e3309c89e5dde8e |
| --- /dev/null |
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc |
| @@ -0,0 +1,79 @@ |
| +/* |
| + * 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 <cmath> |
| + |
| +#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h" |
| +#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| + |
| +namespace webrtc { |
| + |
| +namespace { |
| +constexpr int kMinBitreateChangeBps = 5000; |
| +constexpr float kMinPacketLossChangeFraction = 0.5; |
| +} |
| + |
| +EventLogWriter::EventLogWriter(RtcEventLog* event_log) |
| + : EventLogWriter(event_log, |
| + kMinBitreateChangeBps, |
| + kMinPacketLossChangeFraction) {} |
| + |
| +EventLogWriter::EventLogWriter(RtcEventLog* event_log, |
| + int min_bitreate_change_bps, |
| + float min_packet_loss_change_fraction) |
| + : event_log_(event_log), |
| + min_bitreate_change_bps_(min_bitreate_change_bps), |
| + min_packet_loss_change_fraction_(min_packet_loss_change_fraction) { |
| + RTC_CHECK(event_log_); |
| +} |
| + |
| +EventLogWriter::~EventLogWriter() = default; |
| + |
| +void EventLogWriter::MayLogEncoderRuntimeConfig( |
|
minyue-webrtc
2016/12/12 10:33:12
Do we want to log only the changes or the whole ru
michaelt
2016/12/12 10:50:08
We could save some space if we would just log chan
|
| + const AudioNetworkAdaptor::EncoderRuntimeConfig& config) { |
| + if (last_runtime_config_) { |
| + if (last_runtime_config_->num_channels != config.num_channels || |
| + last_runtime_config_->enable_dtx != config.enable_dtx || |
| + last_runtime_config_->enable_fec != config.enable_fec || |
| + last_runtime_config_->frame_length_ms != config.frame_length_ms) |
| + event_log_->LogAnaDecisionEvent( |
| + config.bitrate_bps, config.frame_length_ms, |
| + config.uplink_packet_loss_fraction, config.enable_fec, |
| + config.enable_dtx, config.num_channels); |
| + |
| + if (last_runtime_config_->bitrate_bps && config.bitrate_bps && |
| + std::abs(*last_runtime_config_->bitrate_bps - *config.bitrate_bps) >= |
| + min_bitreate_change_bps_) |
| + event_log_->LogAnaDecisionEvent( |
| + config.bitrate_bps, config.frame_length_ms, |
| + config.uplink_packet_loss_fraction, config.enable_fec, |
| + config.enable_dtx, config.num_channels); |
| + |
| + if (last_runtime_config_->uplink_packet_loss_fraction && |
| + config.uplink_packet_loss_fraction && |
| + std::abs(*last_runtime_config_->uplink_packet_loss_fraction - |
| + *config.uplink_packet_loss_fraction) >= |
| + min_packet_loss_change_fraction_ * |
| + *last_runtime_config_->uplink_packet_loss_fraction) |
| + event_log_->LogAnaDecisionEvent( |
| + config.bitrate_bps, config.frame_length_ms, |
| + config.uplink_packet_loss_fraction, config.enable_fec, |
| + config.enable_dtx, config.num_channels); |
| + |
| + } else { |
| + event_log_->LogAnaDecisionEvent(config.bitrate_bps, config.frame_length_ms, |
| + config.uplink_packet_loss_fraction, |
| + config.enable_fec, config.enable_dtx, |
| + config.num_channels); |
| + } |
| + last_runtime_config_ = |
| + rtc::Optional<AudioNetworkAdaptor::EncoderRuntimeConfig>(config); |
| +} |
| +} // namespace webrtc |