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Use ArrayView versions of SendRtp and SendRtcp

Browse source 
This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.

Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}
This commit is contained in:
Harald Alvestrand 2023-08-04 12:18:45 +00:00 committed by WebRTC LUCI CQ
parent 8dc55689d2
commit 34d82df2ba
11 changed files with 32 additions and 27 deletions
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16
api/call/transport.h
View file

@ -48,6 +48,8 @@ class Transport {
// New style functions. Default implementations are to accomodate
// subclasses that haven't been converted to new style yet.
// TODO(bugs.webrtc.org/14870): Deprecate and remove old functions.
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wdeprecated-declarations"
virtual bool SendRtp(rtc::ArrayView<const uint8_t> packet,
const PacketOptions& options) {
return SendRtp(packet.data(), packet.size(), options);
@ -55,11 +57,17 @@ class Transport {
virtual bool SendRtcp(rtc::ArrayView<const uint8_t> packet) {
return SendRtcp(packet.data(), packet.size());
}
#pragma clang diagnostic pop
// Old style functions.
virtual bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) = 0;
virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
[[deprecated("Use ArrayView version")]] virtual bool
SendRtp(const uint8_t* packet, size_t length, const PacketOptions& options) {
return SendRtp(rtc::MakeArrayView(packet, length), options);
}
[[deprecated("Use ArrayView version")]] virtual bool SendRtcp(
const uint8_t* packet,
size_t length) {
return SendRtcp(rtc::MakeArrayView(packet, length));
}
protected:
virtual ~Transport() {}

8
call/fake_network_pipe.cc
View file

@ -356,10 +356,12 @@ void FakeNetworkPipe::DeliverNetworkPacket(NetworkPacket* packet) {
return;
}
if (packet->is_rtcp()) {
transport->SendRtcp(packet->data(), packet->data_length());
transport->SendRtcp(
rtc::MakeArrayView(packet->data(), packet->data_length()));
} else {
transport->SendRtp(packet->data(), packet->data_length(),
packet->packet_options());
transport->SendRtp(
rtc::MakeArrayView(packet->data(), packet->data_length()),
packet->packet_options());
}
} else if (receiver_) {
int64_t packet_time_us = packet->packet_time_us().value_or(-1);

3
media/engine/webrtc_video_engine_unittest.cc
View file

@ -5589,8 +5589,7 @@ TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) {
// Packets should also self-identify their dscp in PacketOptions.
const uint8_t kData[10] = {0};
EXPECT_TRUE(ChannelImplAsTransport(send_channel.get())
->SendRtcp(kData, sizeof(kData)));
EXPECT_TRUE(ChannelImplAsTransport(send_channel.get())->SendRtcp(kData));
EXPECT_EQ(rtc::DSCP_CS1, network_interface->options().dscp);
send_channel->SetInterface(nullptr);

4
media/engine/webrtc_voice_engine_unittest.cc
View file

@ -3255,9 +3255,7 @@ TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
// Packets should also self-identify their dscp in PacketOptions.
const uint8_t kData[10] = {0};
EXPECT_TRUE(SendImplFromPointer(channel.get())
->transport()
->SendRtcp(kData, sizeof(kData)));
EXPECT_TRUE(SendImplFromPointer(channel.get())->transport()->SendRtcp(kData));
EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp);
channel->SetInterface(nullptr);

2
modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
View file

@ -416,7 +416,7 @@ bool DEPRECATED_RtpSenderEgress::SendPacketToNetwork(
const PacedPacketInfo& pacing_info) {
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
bytes_sent = transport_->SendRtp(packet, options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {

6
modules/rtp_rtcp/source/rtcp_sender.cc
View file

@ -242,7 +242,7 @@ int32_t RTCPSender::SendLossNotification(const FeedbackState& feedback_state,
bool buffering_allowed) {
int32_t error_code = -1;
auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
transport_->SendRtcp(packet.data(), packet.size());
transport_->SendRtcp(packet);
error_code = 0;
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
@ -659,7 +659,7 @@ int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
const uint16_t* nack_list) {
int32_t error_code = -1;
auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
if (transport_->SendRtcp(packet.data(), packet.size())) {
if (transport_->SendRtcp(packet)) {
error_code = 0;
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
@ -938,7 +938,7 @@ void RTCPSender::SendCombinedRtcpPacket(
}
RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
if (transport_->SendRtcp(packet.data(), packet.size())) {
if (transport_->SendRtcp(packet)) {
if (event_log_)
event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
}

2
modules/rtp_rtcp/source/rtp_sender_egress.cc
View file

@ -522,7 +522,7 @@ bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
RTC_DCHECK_RUN_ON(worker_queue_);
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
bytes_sent = transport_->SendRtp(packet, options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {

2
video/end_to_end_tests/retransmission_tests.cc
View file

@ -166,7 +166,7 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
nack.SetPacketIds(nack_list, 1);
rtc::Buffer buffer = nack.Build();
EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
EXPECT_TRUE(receive_transport_->SendRtcp(buffer));
}
return SEND_PACKET;

2
video/end_to_end_tests/rtp_rtcp_tests.cc
View file

@ -336,7 +336,7 @@ void RtpRtcpEndToEndTest::TestRtpStatePreservation(
rtcp::RapidResyncRequest force_send_sr_back_request;
rtc::Buffer packet = force_send_sr_back_request.Build();
static_cast<webrtc::Transport*>(receive_transport_.get())
->SendRtcp(packet.data(), packet.size());
->SendRtcp(packet);
}
CreateFrameGeneratorCapturer(30, 1280, 720);
StartVideoSources();

9
video/transport_adapter.cc
View file

@ -22,20 +22,19 @@ TransportAdapter::TransportAdapter(Transport* transport)
TransportAdapter::~TransportAdapter() = default;
bool TransportAdapter::SendRtp(const uint8_t* packet,
size_t length,
bool TransportAdapter::SendRtp(rtc::ArrayView<const uint8_t> packet,
const PacketOptions& options) {
if (!enabled_.load())
return false;
return transport_->SendRtp(packet, length, options);
return transport_->SendRtp(packet, options);
}
bool TransportAdapter::SendRtcp(const uint8_t* packet, size_t length) {
bool TransportAdapter::SendRtcp(rtc::ArrayView<const uint8_t> packet) {
if (!enabled_.load())
return false;
return transport_->SendRtcp(packet, length);
return transport_->SendRtcp(packet);
}
void TransportAdapter::Enable() {

5
video/transport_adapter.h
View file

@ -25,10 +25,9 @@ class TransportAdapter : public Transport {
explicit TransportAdapter(Transport* transport);
~TransportAdapter() override;
bool SendRtp(const uint8_t* packet,
size_t length,
bool SendRtp(rtc::ArrayView<const uint8_t> packet,
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* packet, size_t length) override;
bool SendRtcp(rtc::ArrayView<const uint8_t> packet) override;
void Enable();
void Disable();