Remove ResetStatistics from RTP feedback.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1213603002
Cr-Commit-Position: refs/heads/master@{#9548}
diff --git a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
index f7de91f..6bd5cd8 100644
--- a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
+++ b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
@@ -35,9 +35,6 @@
virtual uint32_t BitrateReceived() const = 0;
- // Resets all statistics.
- virtual void ResetStatistics() = 0;
-
// Returns true if the packet with RTP header |header| is likely to be a
// retransmitted packet, false otherwise.
virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 4fe10c8..98f7c26 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -415,13 +415,6 @@
virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
/*
- * Reset RTP data counters for the sending side
- *
- * return -1 on failure else 0
- */
- virtual int32_t ResetSendDataCountersRTP() = 0;
-
- /*
* Statistics of the amount of data sent
*
* return -1 on failure else 0
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index 1587762..ed7dfe0 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -252,8 +252,6 @@
virtual void OnIncomingCSRCChanged( const int32_t id,
const uint32_t CSRC,
const bool added) = 0;
-
- virtual void ResetStatistics(uint32_t ssrc) = 0;
};
class RtpAudioFeedback {
@@ -323,8 +321,6 @@
void OnIncomingCSRCChanged(const int32_t id,
const uint32_t CSRC,
const bool added) override {}
-
- void ResetStatistics(uint32_t ssrc) override {}
};
// Null object version of RtpData.
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index fce5f1b..99e5b1c 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -161,8 +161,6 @@
int32_t(const uint64_t pictureID));
MOCK_METHOD1(SendRTCPSliceLossIndication,
int32_t(const uint8_t pictureID));
- MOCK_METHOD0(ResetSendDataCountersRTP,
- int32_t());
MOCK_CONST_METHOD2(DataCountersRTP,
int32_t(size_t *bytesSent, uint32_t *packetsSent));
MOCK_CONST_METHOD2(GetSendStreamDataCounters,
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 3846558..01116d6 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -51,22 +51,6 @@
rtcp_callback_(rtcp_callback),
rtp_callback_(rtp_callback) {}
-void StreamStatisticianImpl::ResetStatistics() {
- CriticalSectionScoped cs(stream_lock_.get());
- last_report_inorder_packets_ = 0;
- last_report_old_packets_ = 0;
- last_report_seq_max_ = 0;
- last_reported_statistics_ = RtcpStatistics();
- jitter_q4_ = 0;
- cumulative_loss_ = 0;
- jitter_q4_transmission_time_offset_ = 0;
- received_seq_wraps_ = 0;
- received_seq_max_ = 0;
- received_seq_first_ = 0;
- stored_sum_receive_counters_.Add(receive_counters_);
- receive_counters_ = StreamDataCounters();
-}
-
void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
size_t packet_length,
bool retransmitted) {
@@ -320,7 +304,6 @@
StreamDataCounters* data_counters) const {
CriticalSectionScoped cs(stream_lock_.get());
*data_counters = receive_counters_;
- data_counters->Add(stored_sum_receive_counters_);
}
uint32_t StreamStatisticianImpl::BitrateReceived() const {
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index cd41744..e949229 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -36,7 +36,6 @@
void GetReceiveStreamDataCounters(
StreamDataCounters* data_counters) const override;
uint32_t BitrateReceived() const override;
- void ResetStatistics() override;
bool IsRetransmitOfOldPacket(const RTPHeader& header,
int64_t min_rtt) const override;
bool IsPacketInOrder(uint16_t sequence_number) const override;
@@ -85,9 +84,6 @@
size_t received_packet_overhead_;
StreamDataCounters receive_counters_;
- // Stored counter values. Includes sum of reset counter values for the stream.
- StreamDataCounters stored_sum_receive_counters_;
-
// Counter values when we sent the last report.
uint32_t last_report_inorder_packets_;
uint32_t last_report_old_packets_;
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 8b25bcf..fe9f714 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -143,12 +143,6 @@
EXPECT_GT(counters.first_packet_time_ms, -1);
EXPECT_EQ(1u, counters.transmitted.packets);
- statistician->ResetStatistics();
- // GetReceiveStreamDataCounters includes reset counter values.
- statistician->GetReceiveStreamDataCounters(&counters);
- EXPECT_GT(counters.first_packet_time_ms, -1);
- EXPECT_EQ(1u, counters.transmitted.packets);
-
receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
statistician->GetReceiveStreamDataCounters(&counters);
EXPECT_GT(counters.first_packet_time_ms, -1);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index e19378a..c9a1adf 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -228,10 +228,8 @@
void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
PayloadUnion* specific_payload,
- bool* should_reset_statistics,
bool* should_discard_changes) {
*should_discard_changes = false;
- *should_reset_statistics = false;
if (TelephoneEventPayloadType(payload_type)) {
// Don't do callbacks for DTMF packets.
@@ -244,8 +242,6 @@
&specific_payload->Audio.frequency,
&cng_payload_type_has_changed);
- *should_reset_statistics = cng_payload_type_has_changed;
-
if (is_cng_payload_type) {
// Don't do callbacks for DTMF packets.
*should_discard_changes = true;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index cd146b9..a7efcbb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -93,7 +93,6 @@
// statistics. In addition we sometimes need to tweak the frequency.
void CheckPayloadChanged(int8_t payload_type,
PayloadUnion* specific_payload,
- bool* should_reset_statistics,
bool* should_discard_changes) override;
int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 3640933..6be0c5a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -171,13 +171,9 @@
int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
bool is_red = false;
- bool should_reset_statistics = false;
- if (CheckPayloadChanged(rtp_header,
- first_payload_byte,
- is_red,
- &payload_specific,
- &should_reset_statistics) == -1) {
+ if (CheckPayloadChanged(rtp_header, first_payload_byte, is_red,
+ &payload_specific) == -1) {
if (payload_length == 0) {
// OK, keep-alive packet.
return true;
@@ -186,10 +182,6 @@
return false;
}
- if (should_reset_statistics) {
- cb_rtp_feedback_->ResetStatistics(ssrc_);
- }
-
WebRtcRTPHeader webrtc_rtp_header;
memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
webrtc_rtp_header.header = rtp_header;
@@ -276,8 +268,6 @@
// We need the payload_type_ to make the call if the remote SSRC is 0.
new_ssrc = true;
- cb_rtp_feedback_->ResetStatistics(ssrc_);
-
last_received_timestamp_ = 0;
last_received_sequence_number_ = 0;
last_received_frame_time_ms_ = -1;
@@ -330,12 +320,10 @@
// this code path moves we can get rid of some of the rtp_receiver ->
// media_specific interface (such as CheckPayloadChange, possibly get/set
// last known payload).
-int32_t RtpReceiverImpl::CheckPayloadChanged(
- const RTPHeader& rtp_header,
- const int8_t first_payload_byte,
- bool& is_red,
- PayloadUnion* specific_payload,
- bool* should_reset_statistics) {
+int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
+ const int8_t first_payload_byte,
+ bool& is_red,
+ PayloadUnion* specific_payload) {
bool re_initialize_decoder = false;
char payload_name[RTP_PAYLOAD_NAME_SIZE];
@@ -367,11 +355,10 @@
return 0;
}
}
- *should_reset_statistics = false;
bool should_discard_changes = false;
rtp_media_receiver_->CheckPayloadChanged(
- payload_type, specific_payload, should_reset_statistics,
+ payload_type, specific_payload,
&should_discard_changes);
if (should_discard_changes) {
@@ -403,9 +390,6 @@
re_initialize_decoder = false;
}
}
- if (re_initialize_decoder) {
- *should_reset_statistics = true;
- }
} else {
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
is_red = false;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 539c226..c904e1f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -73,8 +73,7 @@
int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
const int8_t first_payload_byte,
bool& is_red,
- PayloadUnion* payload,
- bool* should_reset_statistics);
+ PayloadUnion* payload);
Clock* clock_;
RTPPayloadRegistry* rtp_payload_registry_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
index 56dd081..5fae3db 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
@@ -36,11 +36,9 @@
void RTPReceiverStrategy::CheckPayloadChanged(int8_t payload_type,
PayloadUnion* specific_payload,
- bool* should_reset_statistics,
bool* should_discard_changes) {
- // Default: Keep changes and don't reset statistics.
+ // Default: Keep changes.
*should_discard_changes = false;
- *should_reset_statistics = false;
}
int RTPReceiverStrategy::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index b34ad38..9c09f8e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -79,7 +79,6 @@
// reset statistics and/or discard this packet.
virtual void CheckPayloadChanged(int8_t payload_type,
PayloadUnion* specific_payload,
- bool* should_reset_statistics,
bool* should_discard_changes);
virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 50382ea..d2e224d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -540,12 +540,6 @@
return ret;
}
-// Reset RTP data counters for the sending side.
-int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
- rtp_sender_.ResetDataCounters();
- return 0; // TODO(pwestin): change to void.
-}
-
// Force a send of an RTCP packet.
// Normal SR and RR are triggered via the process function.
int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 96eee96..9cd7e70 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -162,8 +162,6 @@
int32_t SendCompoundRTCP(
const std::set<RTCPPacketType>& rtcpPacketTypes) override;
- int32_t ResetSendDataCountersRTP() override;
-
// Statistics of the amount of data sent and received.
int32_t DataCountersRTP(size_t* bytes_sent,
uint32_t* packets_sent) const override;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 29b2897..0456688 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1101,26 +1101,6 @@
return first_allocated_sequence_number;
}
-void RTPSender::ResetDataCounters() {
- uint32_t ssrc;
- uint32_t ssrc_rtx;
- bool report_rtx;
- {
- CriticalSectionScoped ssrc_lock(send_critsect_.get());
- ssrc = ssrc_;
- ssrc_rtx = ssrc_rtx_;
- report_rtx = rtx_ != kRtxOff;
- }
- CriticalSectionScoped lock(statistics_crit_.get());
- rtp_stats_ = StreamDataCounters();
- rtx_rtp_stats_ = StreamDataCounters();
- if (rtp_stats_callback_) {
- rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc);
- if (report_rtx)
- rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx);
- }
-}
-
void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const {
CriticalSectionScoped lock(statistics_crit_.get());
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 8a624c7..61a1fb5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -131,8 +131,6 @@
void GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const;
- void ResetDataCounters();
-
uint32_t StartTimestamp() const;
void SetStartTimestamp(uint32_t timestamp, bool force);
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 10f5055..83ce92b 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -245,7 +245,6 @@
void GetReceiveStreamDataCounters(
StreamDataCounters* data_counters) const override {}
uint32_t BitrateReceived() const override { return 0; }
- void ResetStatistics() override {}
bool IsRetransmitOfOldPacket(const RTPHeader& header,
int64_t min_rtt) const override {
return false;
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 2eb7a5c..f4d28eb 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -229,20 +229,61 @@
int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
- rtcp_counter.nack_packets * 60 / elapsed_sec);
+ rtcp_counter.nack_packets * 60 / elapsed_sec);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
- rtcp_counter.fir_packets * 60 / elapsed_sec);
+ rtcp_counter.fir_packets * 60 / elapsed_sec);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
- rtcp_counter.pli_packets * 60 / elapsed_sec);
+ rtcp_counter.pli_packets * 60 / elapsed_sec);
if (rtcp_counter.nack_requests > 0) {
RTC_HISTOGRAM_PERCENTAGE(
"WebRTC.Video.UniqueNackRequestsReceivedInPercent",
- rtcp_counter.UniqueNackRequestsInPercent());
+ rtcp_counter.UniqueNackRequestsInPercent());
}
int fraction_lost = report_block_stats_sender_->FractionLostInPercent();
if (fraction_lost != -1) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent",
- fraction_lost);
+ fraction_lost);
+ }
+ }
+
+ StreamDataCounters rtp;
+ StreamDataCounters rtx;
+ GetSendStreamDataCounters(&rtp, &rtx);
+ StreamDataCounters rtp_rtx = rtp;
+ rtp_rtx.Add(rtx);
+ elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
+ Clock::GetRealTimeClock()->TimeInMilliseconds()) /
+ 1000;
+ if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+ RTC_HISTOGRAM_COUNTS_100000(
+ "WebRTC.Video.BitrateSentInKbps",
+ static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+ 1000));
+ RTC_HISTOGRAM_COUNTS_10000(
+ "WebRTC.Video.MediaBitrateSentInKbps",
+ static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+ RTC_HISTOGRAM_COUNTS_10000(
+ "WebRTC.Video.PaddingBitrateSentInKbps",
+ static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
+ 1000));
+ RTC_HISTOGRAM_COUNTS_10000(
+ "WebRTC.Video.RetransmittedBitrateSentInKbps",
+ static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
+ elapsed_sec / 1000));
+ if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
+ RTC_HISTOGRAM_COUNTS_10000(
+ "WebRTC.Video.RtxBitrateSentInKbps",
+ static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+ 1000));
+ }
+ bool fec_enabled = false;
+ uint8_t pltype_red;
+ uint8_t pltype_fec;
+ rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
+ if (fec_enabled) {
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
+ static_cast<int>(rtp_rtx.fec.TotalBytes() *
+ 8 / elapsed_sec / 1000));
}
}
} else if (vie_receiver_.GetRemoteSsrc() > 0) {
@@ -302,49 +343,6 @@
}
}
-void ViEChannel::UpdateHistogramsAtStopSend() {
- StreamDataCounters rtp;
- StreamDataCounters rtx;
- GetSendStreamDataCounters(&rtp, &rtx);
- StreamDataCounters rtp_rtx = rtp;
- rtp_rtx.Add(rtx);
-
- int64_t elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
- Clock::GetRealTimeClock()->TimeInMilliseconds()) / 1000;
- if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
- return;
- }
- RTC_HISTOGRAM_COUNTS_100000(
- "WebRTC.Video.BitrateSentInKbps",
- static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
- 1000));
- RTC_HISTOGRAM_COUNTS_10000(
- "WebRTC.Video.MediaBitrateSentInKbps",
- static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
- RTC_HISTOGRAM_COUNTS_10000(
- "WebRTC.Video.PaddingBitrateSentInKbps",
- static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
- 1000));
- RTC_HISTOGRAM_COUNTS_10000(
- "WebRTC.Video.RetransmittedBitrateSentInKbps",
- static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
- 1000));
- if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateSentInKbps",
- static_cast<int>(rtx.transmitted.TotalBytes() *
- 8 / elapsed_sec / 1000));
- }
- bool fec_enabled = false;
- uint8_t pltype_red;
- uint8_t pltype_fec;
- rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
- if (fec_enabled) {
- RTC_HISTOGRAM_COUNTS_10000(
- "WebRTC.Video.FecBitrateSentInKbps",
- static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000));
- }
-}
-
int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
bool new_stream) {
DCHECK(sender_);
@@ -1336,7 +1334,6 @@
}
int32_t ViEChannel::StopSend() {
- UpdateHistogramsAtStopSend();
send_payload_router_->set_active(false);
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
rtp_rtcp_->SetSendingMediaStatus(false);
@@ -1350,8 +1347,6 @@
return -1;
}
- // Reset.
- rtp_rtcp_->ResetSendDataCountersRTP();
if (rtp_rtcp_->SetSendingStatus(false) != 0) {
return -1;
}
@@ -1359,7 +1354,6 @@
it != simulcast_rtp_rtcp_.end();
it++) {
RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->ResetSendDataCountersRTP();
rtp_rtcp->SetSendingStatus(false);
}
return 0;
@@ -1698,13 +1692,6 @@
CriticalSectionScoped cs(callback_cs_.get());
}
-void ViEChannel::ResetStatistics(uint32_t ssrc) {
- StreamStatistician* statistician =
- vie_receiver_.GetReceiveStatistics()->GetStatistician(ssrc);
- if (statistician)
- statistician->ResetStatistics();
-}
-
void ViEChannel::RegisterSendFrameCountObserver(
FrameCountObserver* observer) {
send_frame_count_observer_.Set(observer);
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 12f23fb..426cc3a 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -258,7 +258,6 @@
virtual void OnIncomingCSRCChanged(const int32_t id,
const uint32_t CSRC,
const bool added);
- virtual void ResetStatistics(uint32_t);
int32_t SetRemoteSSRCType(const StreamType usage, const uint32_t SSRC);
@@ -375,7 +374,6 @@
void SetRtxSendStatus(bool enable);
void UpdateHistograms();
- void UpdateHistogramsAtStopSend();
// ViEChannel exposes methods that allow to modify observers and callbacks
// to be modified. Such an API-style is cumbersome to implement and maintain
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 6dd64c7..8992425 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -74,11 +74,6 @@
void CNameChanged(const char* cname, uint32_t ssrc) override {}
- void ResetStatistics() {
- CriticalSectionScoped cs(stats_lock_.get());
- stats_ = ChannelStatistics();
- }
-
ChannelStatistics GetStats() {
CriticalSectionScoped cs(stats_lock_.get());
return stats_;
@@ -340,15 +335,6 @@
id, CSRC, added);
}
-void Channel::ResetStatistics(uint32_t ssrc) {
- StreamStatistician* statistician =
- rtp_receive_statistics_->GetStatistician(ssrc);
- if (statistician) {
- statistician->ResetStatistics();
- }
- statistics_proxy_->ResetStatistics();
-}
-
int32_t
Channel::OnInitializeDecoder(
int32_t id,
@@ -1206,8 +1192,7 @@
// Reset sending SSRC and sequence number and triggers direct transmission
// of RTCP BYE
- if (_rtpRtcpModule->SetSendingStatus(false) == -1 ||
- _rtpRtcpModule->ResetSendDataCountersRTP() == -1)
+ if (_rtpRtcpModule->SetSendingStatus(false) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index bb9ded9..398087e 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -368,7 +368,6 @@
uint32_t rate) override;
void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override;
void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override;
- void ResetStatistics(uint32_t ssrc) override;
// From RtpAudioFeedback in the RTP/RTCP module
void OnPlayTelephoneEvent(int32_t id,
