Rename Call(Send|Receive)Statistics to Channel...Statistics
Follow-up from
https://webrtc-review.googlesource.com/c/src/+/403188
since "Call" is generally thought to refer to the APIs from call/
nowadays. Also fix variable naming style and partially move from
integer to TimeDelta.
Bug: None
Change-Id: I35e4028173c55e3fc3a81a4066a04eba920bf7f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/403481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45319}
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 0496aa8..3a4ae17 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -278,29 +278,33 @@
stats.codec_payload_type = receive_codec->first;
}
- CallReceiveStatistics call_stats = channel_receive_->GetRTCPStatistics();
- stats.payload_bytes_received = call_stats.payload_bytes_received;
+ ChannelReceiveStatistics channel_stats =
+ channel_receive_->GetRTCPStatistics();
+ stats.payload_bytes_received = channel_stats.payload_bytes_received;
stats.header_and_padding_bytes_received =
- call_stats.header_and_padding_bytes_received;
- stats.packets_received = call_stats.packets_received;
- stats.packets_received_with_ect1 = call_stats.packets_received_with_ect1;
- stats.packets_received_with_ce = call_stats.packets_received_with_ce;
- stats.packets_lost = call_stats.packets_lost;
- stats.jitter_ms = call_stats.jitter_ms;
- stats.nacks_sent = call_stats.nacks_sent;
- stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms;
- stats.last_packet_received = call_stats.last_packet_received;
- stats.last_sender_report_timestamp = call_stats.last_sender_report_timestamp;
+ channel_stats.header_and_padding_bytes_received;
+ stats.packets_received = channel_stats.packets_received;
+ stats.packets_received_with_ect1 = channel_stats.packets_received_with_ect1;
+ stats.packets_received_with_ce = channel_stats.packets_received_with_ce;
+ stats.packets_lost = channel_stats.packets_lost;
+ stats.jitter_ms = channel_stats.jitter_ms;
+ stats.nacks_sent = channel_stats.nacks_sent;
+ stats.capture_start_ntp_time_ms = channel_stats.capture_start_ntp_time_ms;
+ stats.last_packet_received = channel_stats.last_packet_received;
+ stats.last_sender_report_timestamp =
+ channel_stats.last_sender_report_timestamp;
stats.last_sender_report_utc_timestamp =
- call_stats.last_sender_report_utc_timestamp;
+ channel_stats.last_sender_report_utc_timestamp;
stats.last_sender_report_remote_utc_timestamp =
- call_stats.last_sender_report_remote_utc_timestamp;
- stats.sender_reports_packets_sent = call_stats.sender_reports_packets_sent;
- stats.sender_reports_bytes_sent = call_stats.sender_reports_bytes_sent;
- stats.sender_reports_reports_count = call_stats.sender_reports_reports_count;
- stats.round_trip_time = call_stats.round_trip_time;
- stats.round_trip_time_measurements = call_stats.round_trip_time_measurements;
- stats.total_round_trip_time = call_stats.total_round_trip_time;
+ channel_stats.last_sender_report_remote_utc_timestamp;
+ stats.sender_reports_packets_sent = channel_stats.sender_reports_packets_sent;
+ stats.sender_reports_bytes_sent = channel_stats.sender_reports_bytes_sent;
+ stats.sender_reports_reports_count =
+ channel_stats.sender_reports_reports_count;
+ stats.round_trip_time = channel_stats.round_trip_time;
+ stats.round_trip_time_measurements =
+ channel_stats.round_trip_time_measurements;
+ stats.total_round_trip_time = channel_stats.total_round_trip_time;
stats.delay_estimate_ms = channel_receive_->GetDelayEstimate();
stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange();
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index c3459b7..98c728e 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -74,7 +74,8 @@
constexpr double kTotalOutputDuration = 0.5;
constexpr int64_t kPlayoutNtpTimestampMs = 5678;
-const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 78, 890, 123};
+const ChannelReceiveStatistics kChannelStats = {678, 234, -12, 567,
+ 78, 890, 123};
const std::pair<int, SdpAudioFormat> kReceiveCodec = {
123,
{"codec_name_recv", 96000, 0}};
@@ -170,7 +171,7 @@
ASSERT_TRUE(channel_receive_);
EXPECT_CALL(*channel_receive_, GetRTCPStatistics())
- .WillOnce(Return(kCallStats));
+ .WillOnce(Return(kChannelStats));
EXPECT_CALL(*channel_receive_, GetDelayEstimate())
.WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
EXPECT_CALL(*channel_receive_, GetSpeechOutputLevelFullRange())
@@ -256,14 +257,15 @@
AudioReceiveStreamInterface::Stats stats =
recv_stream->GetStats(/*get_and_clear_legacy_stats=*/true);
EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
- EXPECT_EQ(kCallStats.payload_bytes_received, stats.payload_bytes_received);
- EXPECT_EQ(kCallStats.header_and_padding_bytes_received,
+ EXPECT_EQ(kChannelStats.payload_bytes_received,
+ stats.payload_bytes_received);
+ EXPECT_EQ(kChannelStats.header_and_padding_bytes_received,
stats.header_and_padding_bytes_received);
- EXPECT_EQ(static_cast<uint32_t>(kCallStats.packets_received),
+ EXPECT_EQ(static_cast<uint32_t>(kChannelStats.packets_received),
stats.packets_received);
- EXPECT_EQ(kCallStats.packets_lost, stats.packets_lost);
+ EXPECT_EQ(kChannelStats.packets_lost, stats.packets_lost);
EXPECT_EQ(kReceiveCodec.second.name, stats.codec_name);
- EXPECT_EQ(kCallStats.jitter_ms, stats.jitter_ms);
+ EXPECT_EQ(kChannelStats.jitter_ms, stats.jitter_ms);
EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms);
EXPECT_EQ(kNetworkStats.preferredBufferSize,
stats.jitter_buffer_preferred_ms);
@@ -327,7 +329,7 @@
EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
EXPECT_EQ(kAudioDecodeStats.decoded_muted_output,
stats.decoding_muted_output);
- EXPECT_EQ(kCallStats.capture_start_ntp_time_ms,
+ EXPECT_EQ(kChannelStats.capture_start_ntp_time_ms,
stats.capture_start_ntp_time_ms);
EXPECT_EQ(kPlayoutNtpTimestampMs, stats.estimated_playout_ntp_timestamp_ms);
recv_stream->UnregisterFromTransport();
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index ac91dd5..024d617 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -434,19 +434,20 @@
stats.local_ssrc = config_.rtp.ssrc;
stats.target_bitrate_bps = channel_send_->GetTargetBitrate();
- webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
- stats.payload_bytes_sent = call_stats.payload_bytes_sent;
+ webrtc::ChannelSendStatistics channel_stats =
+ channel_send_->GetRTCPStatistics();
+ stats.payload_bytes_sent = channel_stats.payload_bytes_sent;
stats.header_and_padding_bytes_sent =
- call_stats.header_and_padding_bytes_sent;
- stats.retransmitted_bytes_sent = call_stats.retransmitted_bytes_sent;
- stats.packets_sent = call_stats.packetsSent;
- stats.packets_sent_with_ect1 = call_stats.packets_sent_with_ect1;
- stats.total_packet_send_delay = call_stats.total_packet_send_delay;
- stats.retransmitted_packets_sent = call_stats.retransmitted_packets_sent;
+ channel_stats.header_and_padding_bytes_sent;
+ stats.retransmitted_bytes_sent = channel_stats.retransmitted_bytes_sent;
+ stats.packets_sent = channel_stats.packets_sent;
+ stats.packets_sent_with_ect1 = channel_stats.packets_sent_with_ect1;
+ stats.total_packet_send_delay = channel_stats.total_packet_send_delay;
+ stats.retransmitted_packets_sent = channel_stats.retransmitted_packets_sent;
// RTT isn't known until a RTCP report is received. Until then, VoiceEngine
// returns 0 to indicate an error value.
- if (call_stats.rttMs > 0) {
- stats.rtt_ms = call_stats.rttMs;
+ if (channel_stats.round_trip_time.ms() > 0) {
+ stats.rtt_ms = channel_stats.round_trip_time.ms();
}
if (config_.send_codec_spec) {
const auto& spec = *config_.send_codec_spec;
@@ -482,9 +483,9 @@
stats.apm_statistics = ap->GetStatistics(has_remote_tracks);
}
- stats.report_block_datas = std::move(call_stats.report_block_datas);
+ stats.report_block_datas = std::move(channel_stats.report_block_datas);
- stats.nacks_received = call_stats.nacks_received;
+ stats.nacks_received = channel_stats.nacks_received;
return stats;
}
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 4383b17..cfd5380 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -90,7 +90,8 @@
constexpr double kEchoReturnLossEnhancement = 101;
constexpr double kResidualEchoLikelihood = -1.0f;
constexpr double kResidualEchoLikelihoodMax = 23.0f;
-constexpr CallSendStatistics kCallStats = {112, 12, 13456, 17890};
+constexpr ChannelSendStatistics kChannelStats = {TimeDelta::Millis(112), 12,
+ 13456, 17890};
constexpr int kFractionLost = 123;
constexpr int kCumulativeLost = 567;
constexpr uint32_t kInterarrivalJitter = 132;
@@ -308,7 +309,7 @@
EXPECT_TRUE(channel_send_);
EXPECT_CALL(*channel_send_, GetRTCPStatistics())
- .WillRepeatedly(Return(kCallStats));
+ .WillRepeatedly(Return(kChannelStats));
EXPECT_CALL(*channel_send_, GetRemoteRTCPReportBlocks())
.WillRepeatedly(Return(report_blocks));
EXPECT_CALL(*channel_send_, GetANAStatistics())
@@ -463,17 +464,17 @@
helper.SetupMockForGetStats(use_null_audio_processing);
AudioSendStream::Stats stats = send_stream->GetStats(true);
EXPECT_EQ(kSsrc, stats.local_ssrc);
- EXPECT_EQ(kCallStats.payload_bytes_sent, stats.payload_bytes_sent);
- EXPECT_EQ(kCallStats.header_and_padding_bytes_sent,
+ EXPECT_EQ(kChannelStats.payload_bytes_sent, stats.payload_bytes_sent);
+ EXPECT_EQ(kChannelStats.header_and_padding_bytes_sent,
stats.header_and_padding_bytes_sent);
- EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
+ EXPECT_EQ(kChannelStats.packets_sent, stats.packets_sent);
EXPECT_EQ(stats.packets_lost, kCumulativeLost);
EXPECT_FLOAT_EQ(stats.fraction_lost, Q8ToFloat(kFractionLost));
EXPECT_EQ(kIsacFormat.name, stats.codec_name);
EXPECT_EQ(stats.jitter_ms,
static_cast<int32_t>(kInterarrivalJitter /
(kIsacFormat.clockrate_hz / 1000)));
- EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
+ EXPECT_EQ(kChannelStats.round_trip_time.ms(), stats.rtt_ms);
EXPECT_EQ(0, stats.audio_level);
EXPECT_EQ(0, stats.total_input_energy);
EXPECT_EQ(0, stats.total_input_duration);
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index b98ac8e..3e119fc 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -186,7 +186,7 @@
PacketRouter* packet_router) override;
void ResetReceiverCongestionControlObjects() override;
- CallReceiveStatistics GetRTCPStatistics() const override;
+ ChannelReceiveStatistics GetRTCPStatistics() const override;
void SetNACKStatus(bool enable, int max_packets) override;
void SetRtcpMode(RtcpMode mode) override;
void SetNonSenderRttMeasurement(bool enabled) override;
@@ -824,9 +824,9 @@
packet_router_ = nullptr;
}
-CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
+ChannelReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- CallReceiveStatistics stats;
+ ChannelReceiveStatistics stats;
// The jitter statistics is updated for each received RTP packet and is based
// on received packets.
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 7eb8639..da8129a 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -50,7 +50,7 @@
class RtpPacketReceived;
class RtpRtcp;
-struct CallReceiveStatistics {
+struct ChannelReceiveStatistics {
int packets_lost = 0;
uint32_t jitter_ms = 0;
int64_t payload_bytes_received = 0;
@@ -145,7 +145,7 @@
PacketRouter* packet_router) = 0;
virtual void ResetReceiverCongestionControlObjects() = 0;
- virtual CallReceiveStatistics GetRTCPStatistics() const = 0;
+ virtual ChannelReceiveStatistics GetRTCPStatistics() const = 0;
virtual void SetNACKStatus(bool enable, int max_packets) = 0;
virtual void SetRtcpMode(webrtc::RtcpMode mode) = 0;
virtual void SetNonSenderRttMeasurement(bool enabled) = 0;
diff --git a/audio/channel_receive_unittest.cc b/audio/channel_receive_unittest.cc
index e228b42..56161d3 100644
--- a/audio/channel_receive_unittest.cc
+++ b/audio/channel_receive_unittest.cc
@@ -163,7 +163,7 @@
AudioFrame audio_frame;
channel.OnRtpPacket(CreateRtpPacket());
channel.GetAudioFrameWithInfo(kSampleRateHz, &audio_frame);
- CallReceiveStatistics stats = channel.GetRTCPStatistics();
+ ChannelReceiveStatistics stats = channel.GetRTCPStatistics();
return stats.capture_start_ntp_time_ms;
}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index b075940..791c597 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -187,7 +187,7 @@
void ResetSenderCongestionControlObjects() override;
void SetRTCP_CNAME(absl::string_view c_name) override;
std::vector<ReportBlockData> GetRemoteRTCPReportBlocks() const override;
- CallSendStatistics GetRTCPStatistics() const override;
+ ChannelSendStatistics GetRTCPStatistics() const override;
// ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
// which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
@@ -791,10 +791,10 @@
return rtp_rtcp_->GetLatestReportBlockData();
}
-CallSendStatistics ChannelSend::GetRTCPStatistics() const {
+ChannelSendStatistics ChannelSend::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- CallSendStatistics stats = {0};
- stats.rttMs = rtp_rtcp_->LastRtt().value_or(TimeDelta::Zero()).ms();
+ ChannelSendStatistics stats = {
+ .round_trip_time = rtp_rtcp_->LastRtt().value_or(TimeDelta::Zero())};
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
@@ -808,7 +808,7 @@
// TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
// separate outbound-rtp stream objects.
stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
- stats.packetsSent =
+ stats.packets_sent =
rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
stats.packets_sent_with_ect1 = rtp_stats.transmitted.packets_with_ect1 +
rtx_stats.transmitted.packets_with_ect1;
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 1706406..185f3e9 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -39,13 +39,13 @@
class FrameEncryptorInterface;
class RtpTransportControllerSendInterface;
-struct CallSendStatistics {
- int64_t rttMs;
+struct ChannelSendStatistics {
+ TimeDelta round_trip_time;
int64_t payload_bytes_sent;
int64_t header_and_padding_bytes_sent;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
uint64_t retransmitted_bytes_sent;
- int packetsSent;
+ int packets_sent;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-packetssentwithect1
int packets_sent_with_ect1;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
@@ -68,7 +68,7 @@
virtual void ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
- virtual CallSendStatistics GetRTCPStatistics() const = 0;
+ virtual ChannelSendStatistics GetRTCPStatistics() const = 0;
virtual void SetEncoder(int payload_type,
const SdpAudioFormat& encoder_format,
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index 7818a36..bedfa5e 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -60,7 +60,10 @@
(PacketRouter*),
(override));
MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override));
- MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override));
+ MOCK_METHOD(ChannelReceiveStatistics,
+ GetRTCPStatistics,
+ (),
+ (const, override));
MOCK_METHOD(NetworkStatistics,
GetNetworkStatistics,
(bool),
@@ -155,7 +158,7 @@
(RtpTransportControllerSendInterface*),
(override));
MOCK_METHOD(void, ResetSenderCongestionControlObjects, (), (override));
- MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const, override));
+ MOCK_METHOD(ChannelSendStatistics, GetRTCPStatistics, (), (const, override));
MOCK_METHOD(std::vector<ReportBlockData>,
GetRemoteRTCPReportBlocks,
(),
