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Upstream commit: https://webrtc.googlesource.com/src/+/00ca0044d4e06fa95706f6ec22c4fbeef9c679f9 Unify helpers IsRtpPacket and IsRtcpPacket Bug: None Change-Id: Ibe942de433435d256cd6827440136936d4b274d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225022 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34419}
34 lines
1.1 KiB
C++
34 lines
1.1 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "test/direct_transport.h"
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#include <string.h>
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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TEST(DemuxerTest, Demuxing) {
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constexpr uint8_t kVideoPayloadType = 100;
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constexpr uint8_t kAudioPayloadType = 101;
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constexpr size_t kPacketSize = 12;
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Demuxer demuxer({{kVideoPayloadType, MediaType::VIDEO},
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{kAudioPayloadType, MediaType::AUDIO}});
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uint8_t data[kPacketSize];
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memset(data, 0, kPacketSize);
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data[0] = 0x80;
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data[1] = kVideoPayloadType;
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EXPECT_EQ(demuxer.GetMediaType(data, kPacketSize), MediaType::VIDEO);
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data[1] = kAudioPayloadType;
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EXPECT_EQ(demuxer.GetMediaType(data, kPacketSize), MediaType::AUDIO);
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}
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} // namespace test
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} // namespace webrtc
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