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Change RtpSender::OnReceiveNACK name and signature

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Name changed to follow style.
list replaced with vector to decrease number of included headers.

R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2276833003 .

Cr-Commit-Position: refs/heads/master@{#13938}
This commit is contained in:
Danil Chapovalov
2016-08-26 18:48:46 +02:00
parent 1bcfce5ff2
commit 2800d74fcf
7 changed files with 13 additions and 16 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
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@ -1263,7 +1263,7 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) { if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) {
LOG(LS_VERBOSE) << "Incoming NACK length: " LOG(LS_VERBOSE) << "Incoming NACK length: "
<< rtcpPacketInformation.nackSequenceNumbers.size(); << rtcpPacketInformation.nackSequenceNumbers.size();
_rtpRtcp.OnReceivedNACK(rtcpPacketInformation.nackSequenceNumbers); _rtpRtcp.OnReceivedNack(rtcpPacketInformation.nackSequenceNumbers);
} }
} }
{ {

3
webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
#include <list>
#include <map> #include <map>
#include <memory> #include <memory>
#include <vector> #include <vector>
@ -61,7 +60,7 @@ class RTCPPacketInformation {
uint32_t rtcpPacketTypeFlags; // RTCPPacketTypeFlags bit field uint32_t rtcpPacketTypeFlags; // RTCPPacketTypeFlags bit field
uint32_t remoteSSRC; uint32_t remoteSSRC;
std::list<uint16_t> nackSequenceNumbers; std::vector<uint16_t> nackSequenceNumbers;
uint8_t applicationSubType; uint8_t applicationSubType;
uint32_t applicationName; uint32_t applicationName;

6
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -870,8 +870,8 @@ int32_t ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection(
GetFeedbackState(), kRtcpRpsi, 0, 0, false, picture_id); GetFeedbackState(), kRtcpRpsi, 0, 0, false, picture_id);
} }
void ModuleRtpRtcpImpl::OnReceivedNACK( void ModuleRtpRtcpImpl::OnReceivedNack(
const std::list<uint16_t>& nack_sequence_numbers) { const std::vector<uint16_t>& nack_sequence_numbers) {
for (uint16_t nack_sequence_number : nack_sequence_numbers) { for (uint16_t nack_sequence_number : nack_sequence_numbers) {
send_loss_stats_.AddLostPacket(nack_sequence_number); send_loss_stats_.AddLostPacket(nack_sequence_number);
} }
@ -884,7 +884,7 @@ void ModuleRtpRtcpImpl::OnReceivedNACK(
if (rtt == 0) { if (rtt == 0) {
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL); rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
} }
rtp_sender_.OnReceivedNACK(nack_sequence_numbers, rtt); rtp_sender_.OnReceivedNack(nack_sequence_numbers, rtt);
} }
void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks( void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(

3
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
#include <list>
#include <set> #include <set>
#include <utility> #include <utility>
#include <vector> #include <vector>
@ -312,7 +311,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback() StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
const override; const override;
void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers); void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers);
void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks); void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
void OnRequestSendReport(); void OnRequestSendReport();

5
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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@ -721,8 +721,9 @@ int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
return 0; return 0;
} }
void RTPSender::OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers, void RTPSender::OnReceivedNack(
int64_t avg_rtt) { const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt) {
TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"RTPSender::OnReceivedNACK", "num_seqnum", "RTPSender::OnReceivedNACK", "num_seqnum",
nack_sequence_numbers.size(), "avg_rtt", avg_rtt); nack_sequence_numbers.size(), "avg_rtt", avg_rtt);

3
webrtc/modules/rtp_rtcp/source/rtp_sender.h
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@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
#include <list>
#include <map> #include <map>
#include <memory> #include <memory>
#include <utility> #include <utility>
@ -183,7 +182,7 @@ class RTPSender {
// NACK. // NACK.
int SelectiveRetransmissions() const; int SelectiveRetransmissions() const;
int SetSelectiveRetransmissions(uint8_t settings); int SetSelectiveRetransmissions(uint8_t settings);
void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers, void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt); int64_t avg_rtt);
void SetStorePacketsStatus(bool enable, uint16_t number_to_store); void SetStorePacketsStatus(bool enable, uint16_t number_to_store);

7
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <list>
#include <memory> #include <memory>
#include <vector> #include <vector>
@ -1637,7 +1636,7 @@ TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
rtp_sender_->SetStorePacketsStatus(true, kNumPackets); rtp_sender_->SetStorePacketsStatus(true, kNumPackets);
const uint16_t kStartSequenceNumber = rtp_sender_->SequenceNumber(); const uint16_t kStartSequenceNumber = rtp_sender_->SequenceNumber();
std::list<uint16_t> sequence_numbers; std::vector<uint16_t> sequence_numbers;
for (int32_t i = 0; i < kNumPackets; ++i) { for (int32_t i = 0; i < kNumPackets; ++i) {
sequence_numbers.push_back(kStartSequenceNumber + i); sequence_numbers.push_back(kStartSequenceNumber + i);
fake_clock_.AdvanceTimeMilliseconds(1); fake_clock_.AdvanceTimeMilliseconds(1);
@ -1650,14 +1649,14 @@ TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
// Resending should work - brings the bandwidth up to the limit. // Resending should work - brings the bandwidth up to the limit.
// NACK bitrate is capped to the same bitrate as the encoder, since the max // NACK bitrate is capped to the same bitrate as the encoder, since the max
// protection overhead is 50% (see MediaOptimization::SetTargetRates). // protection overhead is 50% (see MediaOptimization::SetTargetRates).
rtp_sender_->OnReceivedNACK(sequence_numbers, 0); rtp_sender_->OnReceivedNack(sequence_numbers, 0);
EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_); EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
// Must be at least 5ms in between retransmission attempts. // Must be at least 5ms in between retransmission attempts.
fake_clock_.AdvanceTimeMilliseconds(5); fake_clock_.AdvanceTimeMilliseconds(5);
// Resending should not work, bandwidth exceeded. // Resending should not work, bandwidth exceeded.
rtp_sender_->OnReceivedNACK(sequence_numbers, 0); rtp_sender_->OnReceivedNack(sequence_numbers, 0);
EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_); EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
} }