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Remove ResetStatistics from RTP feedback.

Browse Source 
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1213603002

Cr-Commit-Position: refs/heads/master@{#9548}
This commit is contained in:
pbos 2015-07-07 08:32:48 -07:00 committed by Commit bot
parent 19492f1c4c
commit d436298332
22 changed files with 56 additions and 186 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/modules/rtp_rtcp/interface/receive_statistics.h
View File

@ -35,9 +35,6 @@ class StreamStatistician {
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,

7
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
View File

@ -414,13 +414,6 @@ class RtpRtcp : public Module {
*/
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
*

4
webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
View File

@ -252,8 +252,6 @@ public:
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 @@ class NullRtpFeedback : public RtpFeedback {
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.

2
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -161,8 +161,6 @@ class MockRtpRtcp : public RtpRtcp {
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,

17
webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
View File

@ -51,22 +51,6 @@ StreamStatisticianImpl::StreamStatisticianImpl(
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 @@ void StreamStatisticianImpl::GetReceiveStreamDataCounters(
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 {

4
webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
View File

@ -36,7 +36,6 @@ class StreamStatisticianImpl : public StreamStatistician {
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 @@ class StreamStatisticianImpl : public StreamStatistician {
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_;

6
webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
View File

@ -143,12 +143,6 @@ TEST_F(ReceiveStatisticsTest, GetReceiveStreamDataCounters) {
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);

4
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
View File

@ -228,10 +228,8 @@ RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
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 @@ void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
&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;

1
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
View File

@ -93,7 +93,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
// 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;

30
webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
View File

@ -171,13 +171,9 @@ bool RtpReceiverImpl::IncomingRtpPacket(
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 @@ bool RtpReceiverImpl::IncomingRtpPacket(
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 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
// 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 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
// 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 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(
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 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(
re_initialize_decoder = false;
}
}
if (re_initialize_decoder) {
*should_reset_statistics = true;
}
} else {
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
is_red = false;

3
webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
View File

@ -73,8 +73,7 @@ class RtpReceiverImpl : public RtpReceiver {
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_;

4
webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
View File

@ -36,11 +36,9 @@ void RTPReceiverStrategy::SetLastMediaSpecificPayload(
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 {

1
webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
View File

@ -79,7 +79,6 @@ class RTPReceiverStrategy {
// 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;

6
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -540,12 +540,6 @@ int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc,
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) {

2
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -162,8 +162,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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;

20
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -1101,26 +1101,6 @@ uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
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());

2
webrtc/modules/rtp_rtcp/source/rtp_sender.h
View File

@ -131,8 +131,6 @@ class RTPSender : public RTPSenderInterface {
void GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const;
void ResetDataCounters();
uint32_t StartTimestamp() const;
void SetStartTimestamp(uint32_t timestamp, bool force);

1
webrtc/video/video_send_stream_tests.cc
View File

@ -245,7 +245,6 @@ class FakeReceiveStatistics : public NullReceiveStatistics {
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;

105
webrtc/video_engine/vie_channel.cc
View File

@ -229,20 +229,61 @@ void ViEChannel::UpdateHistograms() {
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::UpdateHistograms() {
}
}
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::StartSend() {
}
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 @@ int32_t ViEChannel::StopSend() {
return -1;
}
// Reset.
rtp_rtcp_->ResetSendDataCountersRTP();
if (rtp_rtcp_->SetSendingStatus(false) != 0) {
return -1;
}
@ -1359,7 +1354,6 @@ int32_t ViEChannel::StopSend() {
it != simulcast_rtp_rtcp_.end();
it++) {
RtpRtcp* rtp_rtcp = *it;
rtp_rtcp->ResetSendDataCountersRTP();
rtp_rtcp->SetSendingStatus(false);
}
return 0;
@ -1698,13 +1692,6 @@ void ViEChannel::OnIncomingCSRCChanged(const int32_t id,
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);

2
webrtc/video_engine/vie_channel.h
View File

@ -258,7 +258,6 @@ class ViEChannel : public VCMFrameTypeCallback,
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 @@ class ViEChannel : public VCMFrameTypeCallback,
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

17
webrtc/voice_engine/channel.cc
View File

@ -74,11 +74,6 @@ class StatisticsProxy : public RtcpStatisticsCallback {
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 @@ void Channel::OnIncomingCSRCChanged(int32_t id,
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 @@ Channel::StopSend()
// 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,

1
webrtc/voice_engine/channel.h
View File

@ -368,7 +368,6 @@ public:
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,