Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index 95f1e6d09882c32c81356254dc63c874fc091929..e23ce21c5dba5c57438060d67eca0879569485c1 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -20,6 +20,25 @@ |
#include "webrtc/audio/utility/audio_frame_operations.h" |
#include "webrtc/call/rtp_transport_controller_send_interface.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
+// TODO(eladalon): Remove events/* after removing the deprecated functions. |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_logging_started.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_logging_stopped.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h" |
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" |
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" |
#include "webrtc/modules/audio_device/include/audio_device.h" |
#include "webrtc/modules/audio_processing/include/audio_processing.h" |
@@ -36,6 +55,7 @@ |
#include "webrtc/rtc_base/format_macros.h" |
#include "webrtc/rtc_base/location.h" |
#include "webrtc/rtc_base/logging.h" |
+#include "webrtc/rtc_base/ptr_util.h" |
#include "webrtc/rtc_base/rate_limiter.h" |
#include "webrtc/rtc_base/task_queue.h" |
#include "webrtc/rtc_base/thread_checker.h" |
@@ -78,6 +98,13 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
void StopLogging() override { RTC_NOTREACHED(); } |
+ void Log(std::unique_ptr<RtcEvent> event) override { |
+ rtc::CritScope lock(&crit_); |
+ if (event_log_) { |
+ event_log_->Log(std::move(event)); |
+ } |
+ } |
+ |
void LogVideoReceiveStreamConfig( |
const webrtc::rtclog::StreamConfig&) override { |
RTC_NOTREACHED(); |
@@ -91,7 +118,8 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
const webrtc::rtclog::StreamConfig& config) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogAudioReceiveStreamConfig(config); |
+ event_log_->Log(rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>( |
+ rtc::MakeUnique<webrtc::rtclog::StreamConfig>(config))); |
} |
} |
@@ -99,14 +127,15 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
const webrtc::rtclog::StreamConfig& config) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogAudioSendStreamConfig(config); |
+ event_log_->Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>( |
+ rtc::MakeUnique<webrtc::rtclog::StreamConfig>(config))); |
} |
} |
void LogIncomingRtpHeader(const RtpPacketReceived& packet) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogIncomingRtpHeader(packet); |
+ event_log_->Log(rtc::MakeUnique<RtcEventRtpPacketIncoming>(packet)); |
} |
} |
@@ -114,28 +143,29 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
int probe_cluster_id) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogOutgoingRtpHeader(packet, probe_cluster_id); |
+ event_log_->Log( |
+ rtc::MakeUnique<RtcEventRtpPacketOutgoing>(packet, probe_cluster_id)); |
} |
} |
void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogIncomingRtcpPacket(packet); |
+ event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>(packet)); |
} |
} |
void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogOutgoingRtcpPacket(packet); |
+ event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(packet)); |
} |
} |
void LogAudioPlayout(uint32_t ssrc) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogAudioPlayout(ssrc); |
+ event_log_->Log(rtc::MakeUnique<RtcEventAudioPlayout>(ssrc)); |
} |
} |
@@ -144,8 +174,8 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
int32_t total_packets) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogLossBasedBweUpdate(bitrate_bps, fraction_loss, |
- total_packets); |
+ event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateLossBased>( |
+ bitrate_bps, fraction_loss, total_packets)); |
} |
} |
@@ -153,7 +183,8 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
BandwidthUsage detector_state) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogDelayBasedBweUpdate(bitrate_bps, detector_state); |
+ event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateDelayBased>( |
+ bitrate_bps, detector_state)); |
} |
} |
@@ -161,7 +192,8 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
const AudioEncoderRuntimeConfig& config) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogAudioNetworkAdaptation(config); |
+ event_log_->Log(rtc::MakeUnique<RtcEventAudioNetworkAdaptation>( |
+ rtc::MakeUnique<AudioEncoderRuntimeConfig>(config))); |
} |
} |
@@ -171,15 +203,16 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
int min_bytes) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogProbeClusterCreated(id, bitrate_bps, min_probes, |
- min_bytes); |
+ event_log_->Log(rtc::MakeUnique<RtcEventProbeClusterCreated>( |
+ id, bitrate_bps, min_probes, min_bytes)); |
} |
}; |
void LogProbeResultSuccess(int id, int bitrate_bps) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogProbeResultSuccess(id, bitrate_bps); |
+ event_log_->Log( |
+ rtc::MakeUnique<RtcEventProbeResultSuccess>(id, bitrate_bps)); |
} |
}; |
@@ -187,7 +220,8 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { |
ProbeFailureReason failure_reason) override { |
rtc::CritScope lock(&crit_); |
if (event_log_) { |
- event_log_->LogProbeResultFailure(id, failure_reason); |
+ event_log_->Log( |
+ rtc::MakeUnique<RtcEventProbeResultFailure>(id, failure_reason)); |
} |
}; |