Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index c7998b2cd8a79026aa532a3a3d91265c8bbd3956..1c4b7b2a714b69df89d4b94695cf63570bd2f824 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; |
@@ -158,11 +153,6 @@ |
else |
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(); |
@@ -200,14 +190,6 @@ |
std::set<FlexfecReceiveStreamImpl*> flexfec_receive_streams_ |
GUARDED_BY(receive_crit_); |
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_; |
@@ -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. |
@@ -741,12 +685,15 @@ |
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 media_it = flexfec_receive_ssrcs_media_.begin(); |
+ while (media_it != flexfec_receive_ssrcs_media_.end()) { |
+ if (media_it->second == receive_stream_impl) |
+ media_it = flexfec_receive_ssrcs_media_.erase(media_it); |
+ 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) |
@@ -754,17 +701,8 @@ |
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) |
- media_it = flexfec_receive_ssrcs_media_.erase(media_it); |
- else |
- ++media_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 |