 Chromium Code Reviews
 Chromium Code Reviews Issue 2559953002:
  Log audio network adapter decisions in event log.  (Closed)
    
  
    Issue 2559953002:
  Log audio network adapter decisions in event log.  (Closed) 
  | 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( | 
| + const AudioNetworkAdaptor::EncoderRuntimeConfig& config) { | 
| + if (last_runtime_config_) { | 
| 
the sun
2016/12/13 16:31:18
I don't understand the mechanics here. It looks li
 
michaelt
2016/12/14 14:41:37
This is partly a bug and party a feature. 
It shou
 | 
| + 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 |