Delete rtc::PacketTime (was an alias for int64_t)
Followup to https://webrtc-review.googlesource.com/c/91860.
Bug: webrtc:9584
Change-Id: Icadf73d6c275ef32167357fc33b3c08158fa096f
Reviewed-on: https://webrtc-review.googlesource.com/c/114545
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26109}diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index b1b4cee..34e1b50 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h
@@ -77,8 +77,8 @@
absl::optional<PacketOptions> packet_options_;
bool is_rtcp_;
// If using a PacketReceiver for incoming degradation, populate with
- // appropriate MediaType and PacketTime. This type/timing will be kept and
- // forwarded. The PacketTime might be altered to reflect time spent in fake
+ // appropriate MediaType and packet time. This type/timing will be kept and
+ // forwarded. The packet time might be altered to reflect time spent in fake
// network pipe.
MediaType media_type_;
absl::optional<int64_t> packet_time_us_;
@@ -124,8 +124,8 @@
// Implements the PacketReceiver interface. When/if packets are delivered,
// they will be passed directly to the receiver instance given in
- // SetReceiver(), without passing through a Demuxer. The receive time in
- // PacketTime will be increased by the amount of time the packet spent in the
+ // SetReceiver(), without passing through a Demuxer. The receive time
+ // will be increased by the amount of time the packet spent in the
// fake network pipe.
PacketReceiver::DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
diff --git a/media/engine/webrtcvideoengine_unittest.cc b/media/engine/webrtcvideoengine_unittest.cc
index f63c2c0..31dfb55 100644
--- a/media/engine/webrtcvideoengine_unittest.cc
+++ b/media/engine/webrtcvideoengine_unittest.cc
@@ -7042,7 +7042,7 @@
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(0, renderer_.num_rendered_frames());
- channel_->OnPacketReceived(&packet1, rtc::PacketTime());
+ channel_->OnPacketReceived(&packet1, /*packet_time_us=*/-1);
std::vector<webrtc::RtpSource> sources = channel_->GetSources(kSsrc);
EXPECT_EQ(1u, sources.size());
@@ -7052,7 +7052,7 @@
// a new packet.
int64_t timeDeltaMs = 1;
fake_clock_.AdvanceTime(webrtc::TimeDelta::ms(timeDeltaMs));
- channel_->OnPacketReceived(&packet1, rtc::PacketTime());
+ channel_->OnPacketReceived(&packet1, /*packet_time_us=*/-1);
int64_t timestamp2 = channel_->GetSources(kSsrc)[0].timestamp_ms();
EXPECT_EQ(timestamp2, timestamp1 + timeDeltaMs);
@@ -7077,7 +7077,7 @@
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(0, renderer_.num_rendered_frames());
- channel_->OnPacketReceived(&packet1, rtc::PacketTime());
+ channel_->OnPacketReceived(&packet1, /*packet_time_us=*/-1);
{
ASSERT_EQ(2u, channel_->GetSources(kSsrc).size());
@@ -7106,7 +7106,7 @@
int64_t timeDeltaMs = 1;
fake_clock_.AdvanceTime(webrtc::TimeDelta::ms(timeDeltaMs));
- channel_->OnPacketReceived(&packet2, rtc::PacketTime());
+ channel_->OnPacketReceived(&packet2, /*packet_time_us=*/-1);
{
ASSERT_EQ(2u, channel_->GetSources(kSsrc).size());
diff --git a/pc/channel.cc b/pc/channel.cc
index 2635069..8a2fcb5 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -460,10 +460,8 @@
}
void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) {
- // Reconstruct the PacketTime from the |parsed_packet|.
- // RtpPacketReceived.arrival_time_ms = (PacketTime + 500) / 1000;
- // Note: The |not_before| field is always 0 here. This field is not currently
- // used, so it should be fine.
+ // Take packet time from the |parsed_packet|.
+ // RtpPacketReceived.arrival_time_ms = (timestamp_us + 500) / 1000;
int64_t timestamp_us = -1;
if (parsed_packet.arrival_time_ms() > 0) {
timestamp_us = parsed_packet.arrival_time_ms() * 1000;
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index d8af370..896e5d8 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -862,9 +862,6 @@
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 b35ba0c..0e31c2b 100644
--- a/rtc_base/asyncpacketsocket.h
+++ b/rtc_base/asyncpacketsocket.h
@@ -12,7 +12,6 @@
#define RTC_BASE_ASYNCPACKETSOCKET_H_
#include "rtc_base/constructormagic.h"
-#include "rtc_base/deprecation.h"
#include "rtc_base/dscp.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/socket.h"
@@ -52,10 +51,6 @@
PacketInfo info_signaled_after_sent;
};
-// TODO(bugs.webrtc.org/9584): Compatibility alias, delete as soon as downstream
-// code is updated.
-typedef int64_t PacketTime;
-
// Provides the ability to receive packets asynchronously. Sends are not
// buffered since it is acceptable to drop packets under high load.
class AsyncPacketSocket : public sigslot::has_slots<> {
