Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )
Reason for revert:
Unexpected perf regressions.
Original issue's description:
> Parse FlexFEC RTP headers in Call and add integration with BWE.
>
> BUG=webrtc:5654
>
> Review-Url: https://codereview.webrtc.org/2553863003
> Cr-Commit-Position: refs/heads/master@{#15709}
> Committed: https://chromium.googlesource.com/external/webrtc/+/ab2ffa3b28b55ef359232723049fb88b2dcd807a
TBR=philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2589393002
Cr-Commit-Position: refs/heads/master@{#15727}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index c7998b2..1c4b7b2 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -24,7 +24,6 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/logging.h"
-#include "webrtc/base/optional.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
@@ -40,8 +39,6 @@
#include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
@@ -110,8 +107,6 @@
// Implements RecoveredPacketReceiver.
bool OnRecoveredPacket(const uint8_t* packet, size_t length) override;
- void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet);
-
void SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
@@ -159,11 +154,6 @@
return nullptr;
}
- rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
- size_t length,
- const PacketTime& packet_time)
- SHARED_LOCKS_REQUIRED(receive_crit_);
-
void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
void UpdateReceiveHistograms();
void UpdateHistograms();
@@ -202,14 +192,6 @@
std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
GUARDED_BY(receive_crit_);
- // Registered RTP header extensions for each stream.
- // Note that RTP header extensions are negotiated per track ("m= line") in the
- // SDP, but we have no notion of tracks at the Call level. We therefore store
- // the RTP header extensions per SSRC instead, which leads to some storage
- // overhead.
- std::map<uint32_t, RtpHeaderExtensionMap> received_rtp_header_extensions_
- GUARDED_BY(receive_crit_);
-
std::unique_ptr<RWLockWrapper> send_crit_;
// Audio and Video send streams are owned by the client that creates them.
std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
@@ -363,29 +345,6 @@
Trace::ReturnTrace();
}
-rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
- const uint8_t* packet,
- size_t length,
- const PacketTime& packet_time) {
- RtpPacketReceived parsed_packet;
- if (!parsed_packet.Parse(packet, length))
- return rtc::Optional<RtpPacketReceived>();
-
- auto it = received_rtp_header_extensions_.find(parsed_packet.Ssrc());
- if (it != received_rtp_header_extensions_.end())
- parsed_packet.IdentifyExtensions(it->second);
-
- int64_t arrival_time_ms;
- if (packet_time.timestamp != -1) {
- arrival_time_ms = (packet_time.timestamp + 500) / 1000;
- } else {
- arrival_time_ms = clock_->TimeInMilliseconds();
- }
- parsed_packet.set_arrival_time_ms(arrival_time_ms);
-
- return rtc::Optional<RtpPacketReceived>(std::move(parsed_packet));
-}
-
void Call::UpdateHistograms() {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.LifetimeInSeconds",
@@ -700,40 +659,25 @@
const FlexfecReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-
- RecoveredPacketReceiver* recovered_packet_receiver = this;
FlexfecReceiveStreamImpl* receive_stream =
- new FlexfecReceiveStreamImpl(config, recovered_packet_receiver);
+ new FlexfecReceiveStreamImpl(config, this);
{
WriteLockScoped write_lock(*receive_crit_);
-
- RTC_DCHECK(flexfec_receive_streams_.find(receive_stream) ==
- flexfec_receive_streams_.end());
- flexfec_receive_streams_.insert(receive_stream);
-
for (auto ssrc : config.protected_media_ssrcs)
flexfec_receive_ssrcs_media_.insert(std::make_pair(ssrc, receive_stream));
-
RTC_DCHECK(flexfec_receive_ssrcs_protection_.find(config.remote_ssrc) ==
flexfec_receive_ssrcs_protection_.end());
flexfec_receive_ssrcs_protection_[config.remote_ssrc] = receive_stream;
-
- RTC_DCHECK(received_rtp_header_extensions_.find(config.remote_ssrc) ==
- received_rtp_header_extensions_.end());
- RtpHeaderExtensionMap rtp_header_extensions(config.rtp_header_extensions);
- received_rtp_header_extensions_[config.remote_ssrc] = rtp_header_extensions;
+ flexfec_receive_streams_.insert(receive_stream);
}
-
// TODO(brandtr): Store config in RtcEventLog here.
-
return receive_stream;
}
void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-
RTC_DCHECK(receive_stream != nullptr);
// There exist no other derived classes of FlexfecReceiveStream,
// so this downcast is safe.
@@ -741,19 +685,8 @@
static_cast<FlexfecReceiveStreamImpl*>(receive_stream);
{
WriteLockScoped write_lock(*receive_crit_);
-
- uint32_t ssrc = receive_stream_impl->GetConfig().remote_ssrc;
- received_rtp_header_extensions_.erase(ssrc);
-
// Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
// destroyed.
- auto prot_it = flexfec_receive_ssrcs_protection_.begin();
- while (prot_it != flexfec_receive_ssrcs_protection_.end()) {
- if (prot_it->second == receive_stream_impl)
- prot_it = flexfec_receive_ssrcs_protection_.erase(prot_it);
- else
- ++prot_it;
- }
auto media_it = flexfec_receive_ssrcs_media_.begin();
while (media_it != flexfec_receive_ssrcs_media_.end()) {
if (media_it->second == receive_stream_impl)
@@ -761,10 +694,15 @@
else
++media_it;
}
-
+ auto prot_it = flexfec_receive_ssrcs_protection_.begin();
+ while (prot_it != flexfec_receive_ssrcs_protection_.end()) {
+ if (prot_it->second == receive_stream_impl)
+ prot_it = flexfec_receive_ssrcs_protection_.erase(prot_it);
+ else
+ ++prot_it;
+ }
flexfec_receive_streams_.erase(receive_stream_impl);
}
-
delete receive_stream_impl;
}
@@ -1138,21 +1076,13 @@
if (it != video_receive_ssrcs_.end()) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
- // TODO(brandtr): Notify the BWE of received media packets here.
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
- // Deliver media packets to FlexFEC subsystem. RTP header extensions need
- // not be parsed, as FlexFEC is oblivious to the semantic meaning of the
- // packet contents beyond the 12 byte RTP base header. The BWE is fed
- // information about these media packets from the regular media pipeline.
- rtc::Optional<RtpPacketReceived> parsed_packet =
- ParseRtpPacket(packet, length, packet_time);
- if (parsed_packet) {
- auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc);
- for (auto it = it_bounds.first; it != it_bounds.second; ++it)
- it->second->AddAndProcessReceivedPacket(*parsed_packet);
- }
+ // Deliver media packets to FlexFEC subsystem.
+ auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc);
+ for (auto it = it_bounds.first; it != it_bounds.second; ++it)
+ it->second->AddAndProcessReceivedPacket(packet, length);
if (status == DELIVERY_OK)
event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
return status;
@@ -1161,18 +1091,12 @@
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
auto it = flexfec_receive_ssrcs_protection_.find(ssrc);
if (it != flexfec_receive_ssrcs_protection_.end()) {
- rtc::Optional<RtpPacketReceived> parsed_packet =
- ParseRtpPacket(packet, length, packet_time);
- if (parsed_packet) {
- NotifyBweOfReceivedPacket(*parsed_packet);
- auto status =
- it->second->AddAndProcessReceivedPacket(std::move(*parsed_packet))
- ? DELIVERY_OK
- : DELIVERY_PACKET_ERROR;
- if (status == DELIVERY_OK)
- event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
- return status;
- }
+ auto status = it->second->AddAndProcessReceivedPacket(packet, length)
+ ? DELIVERY_OK
+ : DELIVERY_PACKET_ERROR;
+ if (status == DELIVERY_OK)
+ event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
+ return status;
}
}
return DELIVERY_UNKNOWN_SSRC;
@@ -1204,12 +1128,5 @@
return it->second->OnRecoveredPacket(packet, length);
}
-void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet) {
- RTPHeader header;
- packet.GetHeader(&header);
- congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(),
- packet.payload_size(), header);
-}
-
} // namespace internal
} // namespace webrtc
