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To free up RtpVideoHeader::generic field for codec agnostic details from an rtp header extension. Bug: webrtc:10342 Change-Id: I7b9d869b2ecfedb96dfd860be47ed8dffa058749 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166175 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30396}
71 lines
2.4 KiB
C++
71 lines
2.4 KiB
C++
/*
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* Copyright (c) 2019 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 "modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h"
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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using ::testing::SizeIs;
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TEST(VideoRtpDepacketizerGeneric, NonExtendedHeaderNoFrameId) {
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const size_t kRtpPayloadSize = 10;
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const uint8_t kPayload[kRtpPayloadSize] = {0x01};
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rtc::CopyOnWriteBuffer rtp_payload(kPayload);
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VideoRtpDepacketizerGeneric depacketizer;
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absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
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depacketizer.Parse(rtp_payload);
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ASSERT_TRUE(parsed);
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EXPECT_EQ(parsed->video_header.generic, absl::nullopt);
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EXPECT_THAT(parsed->video_payload, SizeIs(kRtpPayloadSize - 1));
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}
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TEST(VideoRtpDepacketizerGeneric, ExtendedHeaderParsesFrameId) {
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const size_t kRtpPayloadSize = 10;
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const uint8_t kPayload[kRtpPayloadSize] = {0x05, 0x13, 0x37};
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rtc::CopyOnWriteBuffer rtp_payload(kPayload);
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VideoRtpDepacketizerGeneric depacketizer;
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absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
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depacketizer.Parse(rtp_payload);
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ASSERT_TRUE(parsed);
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const auto* generic_header = absl::get_if<RTPVideoHeaderLegacyGeneric>(
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&parsed->video_header.video_type_header);
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ASSERT_TRUE(generic_header);
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EXPECT_EQ(generic_header->picture_id, 0x1337);
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EXPECT_THAT(parsed->video_payload, SizeIs(kRtpPayloadSize - 3));
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}
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TEST(VideoRtpDepacketizerGeneric, PassRtpPayloadAsVideoPayload) {
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const uint8_t kPayload[] = {0x01, 0x25, 0x52};
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rtc::CopyOnWriteBuffer rtp_payload(kPayload);
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VideoRtpDepacketizerGeneric depacketizer;
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absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
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depacketizer.Parse(rtp_payload);
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ASSERT_TRUE(parsed);
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// Check there was no memcpy involved by verifying return and original buffers
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// point to the same buffer.
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EXPECT_EQ(parsed->video_payload.cdata(), rtp_payload.cdata() + 1);
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}
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} // namespace
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} // namespace webrtc
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