chromium/third_party/openscreen/src/cast/streaming/impl/rtp_packetizer_unittest.cc

// Copyright 2019 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "cast/streaming/impl/rtp_packetizer.h"

#include <chrono>
#include <memory>
#include <optional>

#include "cast/streaming/impl/frame_crypto.h"
#include "cast/streaming/impl/rtp_defines.h"
#include "cast/streaming/impl/rtp_packet_parser.h"
#include "cast/streaming/ssrc.h"
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "util/chrono_helpers.h"
#include "util/crypto/random_bytes.h"

ElementsAreArray;

namespace openscreen::cast {

namespace {

constexpr RtpPayloadType kPayloadType = …;

// Returns true if |needle| is fully within |haystack|.
bool IsSubspan(ByteView needle, ByteView haystack) { … }

class RtpPacketizerTest : public testing::Test { … };

// Tests that all packets are generated for one key frame, followed by 9 "delta"
// frames. The key frame is larger than the other frames, as is typical in a
// real-world usage scenario.
TEST_F(RtpPacketizerTest, GeneratesPacketsForSequenceOfFrames) { … }

// Tests that all packets are generated for a key frame that includes a playout
// delay change. Only the first packet should mention the playout delay change.
TEST_F(RtpPacketizerTest, GeneratesPacketsForFrameWithLatencyChange) { … }

// Tests that a single, valid RTP packet is generated for a frame with no data
// payload. Having no payload is valid with some codecs (e.g., complete audio
// silence can be represented by an empty payload).
TEST_F(RtpPacketizerTest, GeneratesOnePacketForFrameWithNoPayload) { … }

// Tests that re-generating the same packet for re-transmission works, including
// a different sequence counter value in the packet each time.
TEST_F(RtpPacketizerTest, GeneratesPacketForRetransmission) { … }

}  // namespace
}  // namespace openscreen::cast