Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 113 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/rtp_packet_infos_unittest.cc b/third_party/libwebrtc/api/rtp_packet_infos_unittest.cc
new file mode 100644
index 0000000000..a90cfa03e2
--- /dev/null
+++ b/third_party/libwebrtc/api/rtp_packet_infos_unittest.cc
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/rtp_packet_infos.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::SizeIs;
+
+template <typename Iterator>
+RtpPacketInfos::vector_type ToVector(Iterator begin, Iterator end) {
+  return RtpPacketInfos::vector_type(begin, end);
+}
+
+}  // namespace
+
+TEST(RtpPacketInfosTest, BasicFunctionality) {
+  RtpPacketInfo p0(/*ssrc=*/123, /*csrcs=*/{1, 2}, /*rtp_timestamp=*/89,
+                   /*receive_time=*/Timestamp::Millis(7));
+  p0.set_audio_level(5);
+  p0.set_absolute_capture_time(AbsoluteCaptureTime{
+      .absolute_capture_timestamp = 45, .estimated_capture_clock_offset = 78});
+
+  RtpPacketInfo p1(/*ssrc=*/456, /*csrcs=*/{3, 4}, /*rtp_timestamp=*/89,
+                   /*receive_time=*/Timestamp::Millis(1));
+  p1.set_audio_level(4);
+  p1.set_absolute_capture_time(AbsoluteCaptureTime{
+      .absolute_capture_timestamp = 13, .estimated_capture_clock_offset = 21});
+
+  RtpPacketInfo p2(/*ssrc=*/789, /*csrcs=*/{5, 6}, /*rtp_timestamp=*/88,
+                   /*receive_time=*/Timestamp::Millis(7));
+  p2.set_audio_level(1);
+  p2.set_absolute_capture_time(AbsoluteCaptureTime{
+      .absolute_capture_timestamp = 99, .estimated_capture_clock_offset = 78});
+
+  RtpPacketInfos x({p0, p1, p2});
+
+  ASSERT_THAT(x, SizeIs(3));
+
+  EXPECT_EQ(x[0], p0);
+  EXPECT_EQ(x[1], p1);
+  EXPECT_EQ(x[2], p2);
+
+  EXPECT_EQ(x.front(), p0);
+  EXPECT_EQ(x.back(), p2);
+
+  EXPECT_THAT(ToVector(x.begin(), x.end()), ElementsAre(p0, p1, p2));
+  EXPECT_THAT(ToVector(x.rbegin(), x.rend()), ElementsAre(p2, p1, p0));
+
+  EXPECT_THAT(ToVector(x.cbegin(), x.cend()), ElementsAre(p0, p1, p2));
+  EXPECT_THAT(ToVector(x.crbegin(), x.crend()), ElementsAre(p2, p1, p0));
+
+  EXPECT_FALSE(x.empty());
+}
+
+TEST(RtpPacketInfosTest, CopyShareData) {
+  RtpPacketInfo p0(/*ssrc=*/123, /*csrcs=*/{1, 2}, /*rtp_timestamp=*/89,
+                   /*receive_time=*/Timestamp::Millis(7));
+  p0.set_audio_level(5);
+  p0.set_absolute_capture_time(AbsoluteCaptureTime{
+      .absolute_capture_timestamp = 45, .estimated_capture_clock_offset = 78});
+
+  RtpPacketInfo p1(/*ssrc=*/456, /*csrcs=*/{3, 4}, /*rtp_timestamp=*/89,
+                   /*receive_time=*/Timestamp::Millis(1));
+  p1.set_audio_level(4);
+  p1.set_absolute_capture_time(AbsoluteCaptureTime{
+      .absolute_capture_timestamp = 13, .estimated_capture_clock_offset = 21});
+
+  RtpPacketInfo p2(/*ssrc=*/789, /*csrcs=*/{5, 6}, /*rtp_timestamp=*/88,
+                   /*receive_time=*/Timestamp::Millis(7));
+  p2.set_audio_level(1);
+  p2.set_absolute_capture_time(AbsoluteCaptureTime{
+      .absolute_capture_timestamp = 99, .estimated_capture_clock_offset = 78});
+
+  RtpPacketInfos lhs({p0, p1, p2});
+  RtpPacketInfos rhs = lhs;
+
+  ASSERT_THAT(lhs, SizeIs(3));
+  ASSERT_THAT(rhs, SizeIs(3));
+
+  for (size_t i = 0; i < lhs.size(); ++i) {
+    EXPECT_EQ(lhs[i], rhs[i]);
+  }
+
+  EXPECT_EQ(lhs.front(), rhs.front());
+  EXPECT_EQ(lhs.back(), rhs.back());
+
+  EXPECT_EQ(lhs.begin(), rhs.begin());
+  EXPECT_EQ(lhs.end(), rhs.end());
+  EXPECT_EQ(lhs.rbegin(), rhs.rbegin());
+  EXPECT_EQ(lhs.rend(), rhs.rend());
+
+  EXPECT_EQ(lhs.cbegin(), rhs.cbegin());
+  EXPECT_EQ(lhs.cend(), rhs.cend());
+  EXPECT_EQ(lhs.crbegin(), rhs.crbegin());
+  EXPECT_EQ(lhs.crend(), rhs.crend());
+
+  EXPECT_EQ(lhs.empty(), rhs.empty());
+}
+
+}  // namespace webrtc