Remove Call::ParseRtpPacket
Bug: None
Change-Id: I14e589b1570a81c505336a8972c0e10214e2b289
Reviewed-on: https://webrtc-review.googlesource.com/2002
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20226}diff --git a/call/call.cc b/call/call.cc
index 959fd4b..215c103 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -248,11 +248,6 @@
MediaType media_type)
RTC_SHARED_LOCKS_REQUIRED(receive_crit_);
- rtc::Optional<RtpPacketReceived> ParseRtpPacket(
- const uint8_t* packet,
- size_t length,
- const PacketTime* packet_time) const;
-
void UpdateSendHistograms(int64_t first_sent_packet_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
void UpdateReceiveHistograms();
@@ -504,25 +499,6 @@
UpdateHistograms();
}
-rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
- const uint8_t* packet,
- size_t length,
- const PacketTime* packet_time) const {
- RtpPacketReceived parsed_packet;
- if (!parsed_packet.Parse(packet, length))
- return rtc::Optional<RtpPacketReceived>();
-
- int64_t arrival_time_ms;
- if (packet_time && 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",
@@ -1329,28 +1305,29 @@
const PacketTime& packet_time) {
TRACE_EVENT0("webrtc", "Call::DeliverRtp");
- // TODO(nisse): We should parse the RTP header only here, and pass
- // on parsed_packet to the receive streams.
- rtc::Optional<RtpPacketReceived> parsed_packet =
- ParseRtpPacket(packet, length, &packet_time);
+ RtpPacketReceived parsed_packet;
+ if (!parsed_packet.Parse(packet, length))
+ return DELIVERY_PACKET_ERROR;
+
+ if (packet_time.timestamp != -1) {
+ parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000);
+ } else {
+ parsed_packet.set_arrival_time_ms(clock_->TimeInMilliseconds());
+ }
// We might get RTP keep-alive packets in accordance with RFC6263 section 4.6.
// These are empty (zero length payload) RTP packets with an unsignaled
// payload type.
- const bool is_keep_alive_packet =
- parsed_packet && parsed_packet->payload_size() == 0;
+ const bool is_keep_alive_packet = parsed_packet.payload_size() == 0;
RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO ||
is_keep_alive_packet);
- if (!parsed_packet)
- return DELIVERY_PACKET_ERROR;
-
ReadLockScoped read_lock(*receive_crit_);
- auto it = receive_rtp_config_.find(parsed_packet->Ssrc());
+ auto it = receive_rtp_config_.find(parsed_packet.Ssrc());
if (it == receive_rtp_config_.end()) {
LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc "
- << parsed_packet->Ssrc();
+ << parsed_packet.Ssrc();
// Destruction of the receive stream, including deregistering from the
// RtpDemuxer, is not protected by the |receive_crit_| lock. But
// deregistering in the |receive_rtp_config_| map is protected by that lock.
@@ -1359,17 +1336,17 @@
// which is being torned down.
return DELIVERY_UNKNOWN_SSRC;
}
- parsed_packet->IdentifyExtensions(it->second.extensions);
+ parsed_packet.IdentifyExtensions(it->second.extensions);
- NotifyBweOfReceivedPacket(*parsed_packet, media_type);
+ NotifyBweOfReceivedPacket(parsed_packet, media_type);
if (media_type == MediaType::AUDIO) {
- if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) {
+ if (audio_receiver_controller_.OnRtpPacket(parsed_packet)) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
event_log_->Log(
- rtc::MakeUnique<RtcEventRtpPacketIncoming>(*parsed_packet));
- const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
+ rtc::MakeUnique<RtcEventRtpPacketIncoming>(parsed_packet));
+ const int64_t arrival_time_ms = parsed_packet.arrival_time_ms();
if (!first_received_rtp_audio_ms_) {
first_received_rtp_audio_ms_.emplace(arrival_time_ms);
}
@@ -1377,12 +1354,12 @@
return DELIVERY_OK;
}
} else if (media_type == MediaType::VIDEO) {
- if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) {
+ if (video_receiver_controller_.OnRtpPacket(parsed_packet)) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
event_log_->Log(
- rtc::MakeUnique<RtcEventRtpPacketIncoming>(*parsed_packet));
- const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
+ rtc::MakeUnique<RtcEventRtpPacketIncoming>(parsed_packet));
+ const int64_t arrival_time_ms = parsed_packet.arrival_time_ms();
if (!first_received_rtp_video_ms_) {
first_received_rtp_video_ms_.emplace(arrival_time_ms);
}
@@ -1406,18 +1383,17 @@
}
void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
- rtc::Optional<RtpPacketReceived> parsed_packet =
- ParseRtpPacket(packet, length, nullptr);
- if (!parsed_packet)
+ RtpPacketReceived parsed_packet;
+ if (!parsed_packet.Parse(packet, length))
return;
- parsed_packet->set_recovered(true);
+ parsed_packet.set_recovered(true);
ReadLockScoped read_lock(*receive_crit_);
- auto it = receive_rtp_config_.find(parsed_packet->Ssrc());
+ auto it = receive_rtp_config_.find(parsed_packet.Ssrc());
if (it == receive_rtp_config_.end()) {
LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc "
- << parsed_packet->Ssrc();
+ << parsed_packet.Ssrc();
// Destruction of the receive stream, including deregistering from the
// RtpDemuxer, is not protected by the |receive_crit_| lock. But
// deregistering in the |receive_rtp_config_| map is protected by that lock.
@@ -1426,10 +1402,10 @@
// which is being torned down.
return;
}
- parsed_packet->IdentifyExtensions(it->second.extensions);
+ parsed_packet.IdentifyExtensions(it->second.extensions);
// TODO(brandtr): Update here when we support protecting audio packets too.
- video_receiver_controller_.OnRtpPacket(*parsed_packet);
+ video_receiver_controller_.OnRtpPacket(parsed_packet);
}
void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,