Replace all RTPSender::SendToNetwork with EnqueuePackets
Bug: None
Change-Id: I1bcfbd9c16b329f3aa3f95d8ed61b82131e0da1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316922
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40591}
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index bb65b98..d899b4f 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -461,22 +461,6 @@
return padding_packets;
}
-bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet) {
- RTC_DCHECK(packet);
- auto packet_type = packet->packet_type();
- RTC_CHECK(packet_type) << "Packet type must be set before sending.";
-
- if (packet->capture_time() <= Timestamp::Zero()) {
- packet->set_capture_time(clock_->CurrentTime());
- }
-
- std::vector<std::unique_ptr<RtpPacketToSend>> packets;
- packets.emplace_back(std::move(packet));
- paced_sender_->EnqueuePackets(std::move(packets));
-
- return true;
-}
-
void RTPSender::EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
RTC_DCHECK(!packets.empty());
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index b6a04a7..a398f16 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -144,11 +144,6 @@
return flexfec_ssrc_;
}
- // Sends packet to `transport_` or to the pacer, depending on configuration.
- // TODO(bugs.webrtc.org/XXX): Remove in favor of EnqueuePackets().
- bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet)
- RTC_LOCKS_EXCLUDED(send_mutex_);
-
// Pass a set of packets to RtpPacketSender instance, for paced or immediate
// sending to the network.
void EnqueuePackets(std::vector<std::unique_ptr<RtpPacketToSend>> packets)
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index a0f1af5..0fac440 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -14,6 +14,7 @@
#include <memory>
#include <utility>
+#include <vector>
#include "absl/strings/match.h"
#include "absl/types/optional.h"
@@ -300,11 +301,13 @@
packet->SequenceNumber());
packet->set_packet_type(RtpPacketMediaType::kAudio);
packet->set_allow_retransmission(true);
- bool send_result = rtp_sender_->SendToNetwork(std::move(packet));
+ std::vector<std::unique_ptr<RtpPacketToSend>> packets(1);
+ packets[0] = std::move(packet);
+ rtp_sender_->EnqueuePackets(std::move(packets));
if (first_packet_sent_()) {
RTC_LOG(LS_INFO) << "First audio RTP packet sent to pacer";
}
- return send_result;
+ return true;
}
// Audio level magnitude and voice activity flag are set for each RTP packet
@@ -340,19 +343,16 @@
uint32_t dtmf_timestamp,
uint16_t duration,
bool marker_bit) {
- uint8_t send_count = 1;
- bool result = true;
+ size_t send_count = ended ? 3 : 1;
- if (ended) {
- // resend last packet in an event 3 times
- send_count = 3;
- }
- do {
+ std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+ packets.reserve(send_count);
+ for (size_t i = 0; i < send_count; ++i) {
// Send DTMF data.
constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
constexpr size_t kDtmfSize = 4;
- std::unique_ptr<RtpPacketToSend> packet(
- new RtpPacketToSend(kNoExtensions, kRtpHeaderSize + kDtmfSize));
+ auto packet = std::make_unique<RtpPacketToSend>(kNoExtensions,
+ kRtpHeaderSize + kDtmfSize);
packet->SetPayloadType(dtmf_current_event_.payload_type);
packet->SetMarker(marker_bit);
packet->SetSsrc(rtp_sender_->SSRC());
@@ -383,10 +383,9 @@
packet->set_packet_type(RtpPacketMediaType::kAudio);
packet->set_allow_retransmission(true);
- result = rtp_sender_->SendToNetwork(std::move(packet));
- send_count--;
- } while (send_count > 0 && result);
-
- return result;
+ packets.push_back(std::move(packet));
+ }
+ rtp_sender_->EnqueuePackets(std::move(packets));
+ return true;
}
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 65c49f7..c47edfc 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -201,8 +201,9 @@
packet->set_allow_retransmission(true);
// Packet should be stored in a send bucket.
- EXPECT_TRUE(
- rtp_sender_->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
+ std::vector<std::unique_ptr<RtpPacketToSend>> packets(1);
+ packets[0] = std::make_unique<RtpPacketToSend>(*packet);
+ rtp_sender_->EnqueuePackets(std::move(packets));
return packet;
}
@@ -232,10 +233,12 @@
size_t GenerateAndSendPadding(size_t target_size_bytes) {
size_t generated_bytes = 0;
+ std::vector<std::unique_ptr<RtpPacketToSend>> packets;
for (auto& packet : GeneratePadding(target_size_bytes)) {
generated_bytes += packet->payload_size() + packet->padding_size();
- rtp_sender_->SendToNetwork(std::move(packet));
+ packets.push_back(std::move(packet));
}
+ rtp_sender_->EnqueuePackets(std::move(packets));
return generated_bytes;
}
@@ -339,7 +342,8 @@
}
TEST_F(RtpSenderTest, SendToNetworkForwardsPacketsToPacer) {
- auto packet =
+ std::vector<std::unique_ptr<RtpPacketToSend>> packets(1);
+ packets[0] =
BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, Timestamp::Zero());
Timestamp now = clock_->CurrentTime();
@@ -347,7 +351,8 @@
EnqueuePackets(ElementsAre(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::capture_time, now))))));
- EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet)));
+
+ rtp_sender_->EnqueuePackets(std::move(packets));
}
TEST_F(RtpSenderTest, ReSendPacketForwardsPacketsToPacer) {
@@ -398,7 +403,7 @@
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
Sequence(GeneratePadding(kPaddingTargetBytes));
ASSERT_THAT(padding_packets, SizeIs(1));
- rtp_sender_->SendToNetwork(std::move(padding_packets[0]));
+ rtp_sender_->EnqueuePackets(std::move(padding_packets));
}
// Send a regular video packet again.
