Add implicit conversion between rtc:PacketTime and int64_t.
This is a preparation for deleting rtc::PacketTime. Next step, after
downstream code has been updated to not access the |timestamp| member,
is to make rtc::PacketTime an alias for int64_t.
Also delete the unused member rtc::PacketTime::not_before.
Bug: webrtc:9584
Change-Id: Iba9d2d55047d69565ad62b1beb525591fd432ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/108860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25468}diff --git a/media/base/fakenetworkinterface.h b/media/base/fakenetworkinterface.h
index 0f14659..dfd3407 100644
--- a/media/base/fakenetworkinterface.h
+++ b/media/base/fakenetworkinterface.h
@@ -169,9 +169,9 @@
static_cast<rtc::TypedMessageData<rtc::CopyOnWriteBuffer>*>(msg->pdata);
if (dest_) {
if (msg->message_id == ST_RTP) {
- dest_->OnPacketReceived(&msg_data->data(), rtc::CreatePacketTime(0));
+ dest_->OnPacketReceived(&msg_data->data(), rtc::TimeMicros());
} else {
- dest_->OnRtcpReceived(&msg_data->data(), rtc::CreatePacketTime(0));
+ dest_->OnRtcpReceived(&msg_data->data(), rtc::TimeMicros());
}
}
delete msg_data;
diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index d0f12c0..e376667 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -1379,7 +1379,7 @@
const rtc::PacketTime& packet_time) {
const webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
- packet_time.timestamp);
+ packet_time);
switch (delivery_result) {
case webrtc::PacketReceiver::DELIVERY_OK:
return;
@@ -1427,7 +1427,7 @@
}
if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
- packet_time.timestamp) !=
+ packet_time) !=
webrtc::PacketReceiver::DELIVERY_OK) {
RTC_LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
return;
@@ -1441,7 +1441,7 @@
// filter RTCP anymore incoming RTCP packets could've been going to audio (so
// logging failures spam the log).
call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
- packet_time.timestamp);
+ packet_time);
}
void WebRtcVideoChannel::OnReadyToSend(bool ready) {
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 4fc25cb..f63eb3e 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -2036,7 +2036,7 @@
webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
- packet_time.timestamp);
+ packet_time);
if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
return;
}
@@ -2088,8 +2088,8 @@
SetRawAudioSink(ssrc, std::move(proxy_sink));
}
- delivery_result = call_->Receiver()->DeliverPacket(
- webrtc::MediaType::AUDIO, *packet, packet_time.timestamp);
+ delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
+ *packet, packet_time);
RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
}
@@ -2100,7 +2100,7 @@
// Forward packet to Call as well.
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
- packet_time.timestamp);
+ packet_time);
}
void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
diff --git a/p2p/base/asyncstuntcpsocket.cc b/p2p/base/asyncstuntcpsocket.cc
index 8bb796f..d0185c3 100644
--- a/p2p/base/asyncstuntcpsocket.cc
+++ b/p2p/base/asyncstuntcpsocket.cc
@@ -112,7 +112,7 @@
}
SignalReadPacket(this, data, expected_pkt_len, remote_addr,
- rtc::CreatePacketTime(0));
+ rtc::TimeMicros());
*len -= actual_length;
if (*len > 0) {
diff --git a/p2p/base/dtlstransport.cc b/p2p/base/dtlstransport.cc
index eaf0540..9b74c1c 100644
--- a/p2p/base/dtlstransport.cc
+++ b/p2p/base/dtlstransport.cc
@@ -652,7 +652,7 @@
do {
ret = dtls_->Read(buf, sizeof(buf), &read, &read_error);
if (ret == rtc::SR_SUCCESS) {
- SignalReadPacket(this, buf, read, rtc::CreatePacketTime(0), 0);
+ SignalReadPacket(this, buf, read, rtc::TimeMicros(), 0);
} else if (ret == rtc::SR_EOS) {
// Remote peer shut down the association with no error.
RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed";
diff --git a/p2p/base/fakeicetransport.h b/p2p/base/fakeicetransport.h
index 646aed4..652e69d 100644
--- a/p2p/base/fakeicetransport.h
+++ b/p2p/base/fakeicetransport.h
@@ -266,7 +266,7 @@
if (dest_) {
last_sent_packet_ = packet;
dest_->SignalReadPacket(dest_, packet.data<char>(), packet.size(),
- rtc::CreatePacketTime(0), 0);
+ rtc::TimeMicros(), 0);
}
}
diff --git a/p2p/base/fakepackettransport.h b/p2p/base/fakepackettransport.h
index e57bc17..52b3921 100644
--- a/p2p/base/fakepackettransport.h
+++ b/p2p/base/fakepackettransport.h
@@ -119,7 +119,7 @@
last_sent_packet_ = packet;
if (dest_) {
dest_->SignalReadPacket(dest_, packet.data<char>(), packet.size(),
- CreatePacketTime(0), 0);
+ TimeMicros(), 0);
}
}
diff --git a/p2p/base/p2ptransportchannel_unittest.cc b/p2p/base/p2ptransportchannel_unittest.cc
index ce89b99..99151cb 100644
--- a/p2p/base/p2ptransportchannel_unittest.cc
+++ b/p2p/base/p2ptransportchannel_unittest.cc
@@ -3203,7 +3203,7 @@
msg.AddFingerprint();
rtc::ByteBufferWriter buf;
msg.Write(&buf);
- conn->OnReadPacket(buf.Data(), buf.Length(), rtc::CreatePacketTime(0));
+ conn->OnReadPacket(buf.Data(), buf.Length(), rtc::TimeMicros());
}
void OnReadyToSend(rtc::PacketTransportInternal* transport) {
@@ -3612,7 +3612,7 @@
clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
conn1->ReceivedPing();
- conn1->OnReadPacket("ABC", 3, rtc::CreatePacketTime(0));
+ conn1->OnReadPacket("ABC", 3, rtc::TimeMicros());
EXPECT_TRUE_SIMULATED_WAIT(ch.receiving(), kShortTimeout, clock);
EXPECT_TRUE_SIMULATED_WAIT(!ch.receiving(), kShortTimeout, clock);
}
@@ -3803,7 +3803,7 @@
Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
ASSERT_TRUE(conn2 != nullptr);
conn2->ReceivedPingResponse(LOW_RTT, "id"); // Become writable and receiving.
- conn2->OnReadPacket("ABC", 3, rtc::CreatePacketTime(0));
+ conn2->OnReadPacket("ABC", 3, rtc::TimeMicros());
EXPECT_EQ(conn2, ch.selected_connection());
conn2->ReceivedPingResponse(LOW_RTT, "id"); // Become writable.
@@ -3831,7 +3831,7 @@
// selected connection was nominated by the controlling side.
conn2->ReceivedPing();
conn2->ReceivedPingResponse(LOW_RTT, "id");
- conn2->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
+ conn2->OnReadPacket("XYZ", 3, rtc::TimeMicros());
EXPECT_EQ_WAIT(conn3, ch.selected_connection(), kDefaultTimeout);
}
@@ -3860,12 +3860,12 @@
// Advance the clock by 1ms so that the last data receiving timestamp of
// conn2 is larger.
SIMULATED_WAIT(false, 1, clock);
- conn2->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
+ conn2->OnReadPacket("XYZ", 3, rtc::TimeMicros());
EXPECT_EQ(1, reset_selected_candidate_pair_switches());
EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
// conn1 also receives data; it becomes selected due to priority again.
- conn1->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
+ conn1->OnReadPacket("XYZ", 3, rtc::TimeMicros());
EXPECT_EQ(1, reset_selected_candidate_pair_switches());
EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
@@ -3874,7 +3874,7 @@
SIMULATED_WAIT(false, 1, clock);
// Need to become writable again because it was pruned.
conn2->ReceivedPingResponse(LOW_RTT, "id");
- conn2->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
+ conn2->OnReadPacket("XYZ", 3, rtc::TimeMicros());
EXPECT_EQ(1, reset_selected_candidate_pair_switches());
EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
@@ -3904,7 +3904,7 @@
// conn1 received data; it is the selected connection.
// Advance the clock to have a non-zero last-data-receiving time.
SIMULATED_WAIT(false, 1, clock);
- conn1->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
+ conn1->OnReadPacket("XYZ", 3, rtc::TimeMicros());
EXPECT_EQ(1, reset_selected_candidate_pair_switches());
EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
@@ -4125,7 +4125,7 @@
// conn2.
NominateConnection(conn1);
SIMULATED_WAIT(false, 1, clock);
- conn1->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
+ conn1->OnReadPacket("XYZ", 3, rtc::TimeMicros());
SIMULATED_WAIT(conn2->pruned(), 100, clock);
EXPECT_FALSE(conn2->pruned());
}
diff --git a/p2p/base/turnport_unittest.cc b/p2p/base/turnport_unittest.cc
index 16f89e9..6100dbe 100644
--- a/p2p/base/turnport_unittest.cc
+++ b/p2p/base/turnport_unittest.cc
@@ -1031,8 +1031,7 @@
std::string test_packet = "Test packet";
EXPECT_FALSE(turn_port_->HandleIncomingPacket(
socket_.get(), test_packet.data(), test_packet.size(),
- rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
- rtc::CreatePacketTime(0)));
+ rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0), rtc::TimeMicros()));
}
// Tests that a shared-socket-TurnPort creates its own socket after
diff --git a/pc/channel.cc b/pc/channel.cc
index 8e1d5ff..6a16eb3 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -446,9 +446,8 @@
if (parsed_packet.arrival_time_ms() > 0) {
timestamp = parsed_packet.arrival_time_ms() * 1000;
}
- rtc::PacketTime packet_time(timestamp, /*not_before=*/0);
- OnPacketReceived(/*rtcp=*/false, parsed_packet.Buffer(), packet_time);
+ OnPacketReceived(/*rtcp=*/false, parsed_packet.Buffer(), timestamp);
}
void BaseChannel::UpdateRtpHeaderExtensionMap(
diff --git a/pc/rtptransport.cc b/pc/rtptransport.cc
index 9e994e9..c510359 100644
--- a/pc/rtptransport.cc
+++ b/pc/rtptransport.cc
@@ -195,8 +195,8 @@
return;
}
- if (time.timestamp != -1) {
- parsed_packet.set_arrival_time_ms((time.timestamp + 500) / 1000);
+ if (time != -1) {
+ parsed_packet.set_arrival_time_ms((time + 500) / 1000);
}
rtp_demuxer_.OnRtpPacket(parsed_packet);
}
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index e16648a..3f02fb7 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -699,6 +699,9 @@
defines = []
deps = [
":checks",
+
+ # For deprecation of rtc::PacketTime, in asyncpacketsocket.h.
+ ":deprecation",
":stringutils",
"..:webrtc_common",
"../api:array_view",
diff --git a/rtc_base/asyncpacketsocket.h b/rtc_base/asyncpacketsocket.h
index bb0b3bc..e671f0d 100644
--- a/rtc_base/asyncpacketsocket.h
+++ b/rtc_base/asyncpacketsocket.h
@@ -12,6 +12,7 @@
#define RTC_BASE_ASYNCPACKETSOCKET_H_
#include "rtc_base/constructormagic.h"
+#include "rtc_base/deprecation.h"
#include "rtc_base/dscp.h"
#include "rtc_base/socket.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
@@ -53,21 +54,21 @@
// This structure will have the information about when packet is actually
// received by socket.
struct PacketTime {
- PacketTime() : timestamp(-1), not_before(-1) {}
- PacketTime(int64_t timestamp, int64_t not_before)
- : timestamp(timestamp), not_before(not_before) {}
+ PacketTime() : timestamp(-1) {}
+ // Intentionally implicit.
+ PacketTime(int64_t timestamp) : timestamp(timestamp) {}
+ // Deprecated
+ PacketTime(int64_t timestamp, int64_t /* not_before */)
+ : timestamp(timestamp) {}
+
+ operator int64_t() const { return timestamp; }
int64_t timestamp; // Receive time after socket delivers the data.
-
- // Earliest possible time the data could have arrived, indicating the
- // potential error in the |timestamp| value, in case the system, is busy. For
- // example, the time of the last select() call.
- // If unknown, this value will be set to zero.
- int64_t not_before;
};
-inline PacketTime CreatePacketTime(int64_t not_before) {
- return PacketTime(TimeMicros(), not_before);
+// Deprecated
+inline PacketTime CreatePacketTime(int64_t /* not_before */) {
+ return TimeMicros();
}
// Provides the ability to receive packets asynchronously. Sends are not
diff --git a/rtc_base/asynctcpsocket.cc b/rtc_base/asynctcpsocket.cc
index 087a98e..666b335 100644
--- a/rtc_base/asynctcpsocket.cc
+++ b/rtc_base/asynctcpsocket.cc
@@ -321,7 +321,7 @@
return;
SignalReadPacket(this, data + kPacketLenSize, pkt_len, remote_addr,
- CreatePacketTime(0));
+ TimeMicros());
*len -= kPacketLenSize + pkt_len;
if (*len > 0) {
diff --git a/rtc_base/asyncudpsocket.cc b/rtc_base/asyncudpsocket.cc
index 9cd9e14..2f9011c 100644
--- a/rtc_base/asyncudpsocket.cc
+++ b/rtc_base/asyncudpsocket.cc
@@ -122,9 +122,8 @@
// TODO: Make sure that we got all of the packet.
// If we did not, then we should resize our buffer to be large enough.
- SignalReadPacket(
- this, buf_, static_cast<size_t>(len), remote_addr,
- (timestamp > -1 ? PacketTime(timestamp, 0) : CreatePacketTime(0)));
+ SignalReadPacket(this, buf_, static_cast<size_t>(len), remote_addr,
+ (timestamp > -1 ? timestamp : TimeMicros()));
}
void AsyncUDPSocket::OnWriteEvent(AsyncSocket* socket) {
diff --git a/rtc_base/testclient.cc b/rtc_base/testclient.cc
index 7c151c7..56a9f8c 100644
--- a/rtc_base/testclient.cc
+++ b/rtc_base/testclient.cc
@@ -97,7 +97,7 @@
std::unique_ptr<Packet> packet = NextPacket(kTimeoutMs);
if (packet) {
res = (packet->size == size && memcmp(packet->buf, buf, size) == 0 &&
- CheckTimestamp(packet->packet_time.timestamp));
+ CheckTimestamp(packet->packet_time));
if (addr)
*addr = packet->addr;
}
diff --git a/rtc_tools/network_tester/test_controller.cc b/rtc_tools/network_tester/test_controller.cc
index e5bd92e..4bdc993 100644
--- a/rtc_tools/network_tester/test_controller.cc
+++ b/rtc_tools/network_tester/test_controller.cc
@@ -109,7 +109,7 @@
break;
}
case NetworkTesterPacket::TEST_DATA: {
- packet.set_arrival_timestamp(packet_time.timestamp);
+ packet.set_arrival_timestamp(packet_time);
packet.set_packet_size(len);
packet_logger_.LogPacket(packet);
break;