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Unified Diff: webrtc/video/vie_receiver.cc

Issue 1739893005: Remove add/removal of ViEReceiver RTP modules. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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Index: webrtc/video/vie_receiver.cc
diff --git a/webrtc/video/vie_receiver.cc b/webrtc/video/vie_receiver.cc
index 2f0cf7a8eb493a83a84e5bea5b311a440c98e1c9..0ab72dcc2fedff766781a10fe7aaa32d4142c74a 100644
--- a/webrtc/video/vie_receiver.cc
+++ b/webrtc/video/vie_receiver.cc
@@ -17,10 +17,8 @@
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/include/video_coding.h"
@@ -37,20 +35,17 @@ ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm,
RemoteBitrateEstimator* remote_bitrate_estimator,
RtpFeedback* rtp_feedback)
: clock_(Clock::GetRealTimeClock()),
+ vcm_(module_vcm),
+ remote_bitrate_estimator_(remote_bitrate_estimator),
+ ntp_estimator_(clock_),
+ rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
rtp_header_parser_(RtpHeaderParser::Create()),
- rtp_payload_registry_(
- new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
- rtp_receiver_(
- RtpReceiver::CreateVideoReceiver(clock_,
- this,
- rtp_feedback,
- rtp_payload_registry_.get())),
+ rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
+ this,
+ rtp_feedback,
+ &rtp_payload_registry_)),
rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
fec_receiver_(FecReceiver::Create(this)),
- rtp_rtcp_(NULL),
- vcm_(module_vcm),
- remote_bitrate_estimator_(remote_bitrate_estimator),
- ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
receiving_(false),
restored_packet_in_use_(false),
last_packet_log_ms_(-1) {}
@@ -75,23 +70,17 @@ void ViEReceiver::UpdateHistograms() {
bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
int8_t old_pltype = -1;
- if (rtp_payload_registry_->ReceivePayloadType(video_codec.plName,
+ if (rtp_payload_registry_.ReceivePayloadType(video_codec.plName,
kVideoPayloadTypeFrequency,
0,
video_codec.maxBitrate,
&old_pltype) != -1) {
- rtp_payload_registry_->DeRegisterReceivePayload(old_pltype);
+ rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
}
- return RegisterPayload(video_codec);
-}
-
-bool ViEReceiver::RegisterPayload(const VideoCodec& video_codec) {
- return rtp_receiver_->RegisterReceivePayload(video_codec.plName,
- video_codec.plType,
- kVideoPayloadTypeFrequency,
- 0,
- video_codec.maxBitrate) == 0;
+ return rtp_receiver_->RegisterReceivePayload(
+ video_codec.plName, video_codec.plType, kVideoPayloadTypeFrequency,
+ 0, 0) == 0;
}
void ViEReceiver::SetNackStatus(bool enable,
@@ -108,24 +97,24 @@ void ViEReceiver::SetNackStatus(bool enable,
void ViEReceiver::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
- rtp_payload_registry_->SetRtxPayloadType(payload_type,
+ rtp_payload_registry_.SetRtxPayloadType(payload_type,
associated_payload_type);
}
void ViEReceiver::SetUseRtxPayloadMappingOnRestore(bool val) {
- rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(val);
+ rtp_payload_registry_.set_use_rtx_payload_mapping_on_restore(val);
}
void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
- rtp_payload_registry_->SetRtxSsrc(ssrc);
+ rtp_payload_registry_.SetRtxSsrc(ssrc);
}
bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const {
- return rtp_payload_registry_->GetRtxSsrc(ssrc);
+ return rtp_payload_registry_.GetRtxSsrc(ssrc);
}
bool ViEReceiver::IsFecEnabled() const {
- return rtp_payload_registry_->ulpfec_payload_type() > -1;
+ return rtp_payload_registry_.ulpfec_payload_type() > -1;
}
uint32_t ViEReceiver::GetRemoteSsrc() const {
@@ -136,24 +125,14 @@ int ViEReceiver::GetCsrcs(uint32_t* csrcs) const {
return rtp_receiver_->CSRCs(csrcs);
}
-void ViEReceiver::SetRtpRtcpModule(RtpRtcp* module) {
- rtp_rtcp_ = module;
+void ViEReceiver::Init(const std::vector<RtpRtcp*>& modules) {
+ rtp_rtcp_ = modules;
}
RtpReceiver* ViEReceiver::GetRtpReceiver() const {
return rtp_receiver_.get();
}
-void ViEReceiver::RegisterRtpRtcpModules(
- const std::vector<RtpRtcp*>& rtp_modules) {
- rtc::CritScope lock(&receive_cs_);
- // Only change the "simulcast" modules, the base module can be accessed
- // without a lock whereas the simulcast modules require locking as they can be
- // changed in runtime.
- rtp_rtcp_simulcast_ =
- std::vector<RtpRtcp*>(rtp_modules.begin() + 1, rtp_modules.end());
-}
-
void ViEReceiver::EnableReceiveRtpHeaderExtension(const std::string& extension,
int id) {
RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
@@ -167,7 +146,7 @@ int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data,
RTC_DCHECK(vcm_);
WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
rtp_header_with_ntp.ntp_time_ms =
- ntp_estimator_->Estimate(rtp_header->header.timestamp);
+ ntp_estimator_.Estimate(rtp_header->header.timestamp);
if (vcm_->IncomingPacket(payload_data,
payload_size,
rtp_header_with_ntp) != 0) {
@@ -235,7 +214,7 @@ bool ViEReceiver::DeliverRtp(const uint8_t* rtp_packet,
header.payload_type_frequency = kVideoPayloadTypeFrequency;
bool in_order = IsPacketInOrder(header);
- rtp_payload_registry_->SetIncomingPayloadType(header);
+ rtp_payload_registry_.SetIncomingPayloadType(header);
bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
// Update receive statistics after ReceivePacket.
// Receive statistics will be reset if the payload type changes (make sure
@@ -249,14 +228,14 @@ bool ViEReceiver::ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header,
bool in_order) {
- if (rtp_payload_registry_->IsEncapsulated(header)) {
+ if (rtp_payload_registry_.IsEncapsulated(header)) {
return ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
}
const uint8_t* payload = packet + header.headerLength;
assert(packet_length >= header.headerLength);
size_t payload_length = packet_length - header.headerLength;
PayloadUnion payload_specific;
- if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
+ if (!rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
&payload_specific)) {
return false;
}
@@ -267,8 +246,8 @@ bool ViEReceiver::ReceivePacket(const uint8_t* packet,
bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header) {
- if (rtp_payload_registry_->IsRed(header)) {
- int8_t ulpfec_pt = rtp_payload_registry_->ulpfec_payload_type();
+ if (rtp_payload_registry_.IsRed(header)) {
+ int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
if (packet[header.headerLength] == ulpfec_pt) {
rtp_receive_statistics_->FecPacketReceived(header, packet_length);
// Notify vcm about received FEC packets to avoid NACKing these packets.
@@ -279,7 +258,7 @@ bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
return false;
}
return fec_receiver_->ProcessReceivedFec() == 0;
- } else if (rtp_payload_registry_->IsRtx(header)) {
+ } else if (rtp_payload_registry_.IsRtx(header)) {
if (header.headerLength + header.paddingLength == packet_length) {
// This is an empty packet and should be silently dropped before trying to
// parse the RTX header.
@@ -295,7 +274,7 @@ bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
return false;
}
- if (!rtp_payload_registry_->RestoreOriginalPacket(
+ if (!rtp_payload_registry_.RestoreOriginalPacket(
restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
header)) {
LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
@@ -313,7 +292,7 @@ bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
int8_t last_media_payload_type =
- rtp_payload_registry_->last_received_media_payload_type();
+ rtp_payload_registry_.last_received_media_payload_type();
if (last_media_payload_type < 0) {
LOG(LS_WARNING) << "Failed to get last media payload type.";
return;
@@ -324,7 +303,7 @@ void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
rtp_header.header.payloadType = last_media_payload_type;
rtp_header.header.paddingLength = 0;
PayloadUnion payload_specific;
- if (!rtp_payload_registry_->GetPayloadSpecifics(last_media_payload_type,
+ if (!rtp_payload_registry_.GetPayloadSpecifics(last_media_payload_type,
&payload_specific)) {
LOG(LS_WARNING) << "Failed to get payload specifics.";
return;
@@ -340,23 +319,23 @@ void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
bool ViEReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
size_t rtcp_packet_length) {
stefan-webrtc 2016/02/29 09:15:04 DCHECK or CHECK
pbos-webrtc 2016/02/29 09:58:57 Done.
+ assert(!rtp_rtcp_.empty()); // Should be set by owner at construction time.
{
rtc::CritScope lock(&receive_cs_);
if (!receiving_) {
return false;
}
-
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_simulcast_)
- rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
}
- assert(rtp_rtcp_); // Should be set by owner at construction time.
- int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
- if (ret != 0) {
+
+ // Deliver to simulcast modules first, then base one.
stefan-webrtc 2016/02/29 09:15:04 Do you know why? Seems like it would be cleaner to
pbos-webrtc 2016/02/29 09:58:57 Done.
+ for (size_t i = 1; i < rtp_rtcp_.size(); ++i)
+ rtp_rtcp_[i]->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
+ // TODO(pbos): Is it important to check the return code here?
+ if (rtp_rtcp_[0]->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length) != 0)
return false;
- }
int64_t rtt = 0;
- rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
+ rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
if (rtt == 0) {
// Waiting for valid rtt.
return true;
@@ -364,12 +343,12 @@ bool ViEReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
uint32_t ntp_secs = 0;
uint32_t ntp_frac = 0;
uint32_t rtp_timestamp = 0;
- if (0 != rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
- &rtp_timestamp)) {
+ if (rtp_rtcp_[0]->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
+ &rtp_timestamp) != 0) {
// Waiting for RTCP.
return true;
}
- ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
+ ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
return true;
}
@@ -399,7 +378,7 @@ bool ViEReceiver::IsPacketInOrder(const RTPHeader& header) const {
bool ViEReceiver::IsPacketRetransmitted(const RTPHeader& header,
bool in_order) const {
// Retransmissions are handled separately if RTX is enabled.
- if (rtp_payload_registry_->RtxEnabled())
+ if (rtp_payload_registry_.RtxEnabled())
return false;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(header.ssrc);
@@ -407,7 +386,7 @@ bool ViEReceiver::IsPacketRetransmitted(const RTPHeader& header,
return false;
// Check if this is a retransmission.
int64_t min_rtt = 0;
- rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
+ rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
return !in_order &&
statistician->IsRetransmitOfOldPacket(header, min_rtt);
}
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