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Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.

Browse Source 
Bug: none
Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642
Reviewed-on: https://webrtc-review.googlesource.com/87561
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23904}
This commit is contained in:
philipel
2018-07-09 15:52:29 +02:00
committed by Commit Bot
parent d4c5d63a94
commit 1a4746a563
16 changed files with 152 additions and 50 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
modules/BUILD.gn
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@ -56,6 +56,7 @@ rtc_source_set("module_api") {
"../api/transport:network_control",
"../api/video:video_frame",
"../api/video:video_frame_i420",
"../modules/rtp_rtcp:rtp_video_header",
"../rtc_base:deprecation",
"../rtc_base:rtc_base_approved",
"video_coding:codec_globals_headers",

38
modules/include/module_common_types.h
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@ -24,9 +24,7 @@
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types_public.h"
#include "modules/include/module_fec_types.h"
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/numerics/safe_conversions.h"
@ -38,40 +36,6 @@ namespace webrtc {
// TODO(nisse): Deprecated, use webrtc::VideoCodecType instead.
using RtpVideoCodecTypes = VideoCodecType;
// TODO(philipel): Change from union to abls::variant.
union RTPVideoTypeHeader {
RTPVideoHeaderVP8 VP8;
RTPVideoHeaderVP9 VP9;
RTPVideoHeaderH264 H264;
};
// Since RTPVideoHeader is used as a member of a union, it can't have a
// non-trivial default constructor.
struct RTPVideoHeader {
RTPVideoHeaderVP8& vp8() { return codecHeader.VP8; }
const RTPVideoHeaderVP8& vp8() const { return codecHeader.VP8; }
RTPVideoHeaderVP9& vp9() { return codecHeader.VP9; }
const RTPVideoHeaderVP9& vp9() const { return codecHeader.VP9; }
RTPVideoHeaderH264& h264() { return codecHeader.H264; }
const RTPVideoHeaderH264& h264() const { return codecHeader.H264; }
uint16_t width; // size
uint16_t height;
VideoRotation rotation;
PlayoutDelay playout_delay;
VideoContentType content_type;
VideoSendTiming video_timing;
bool is_first_packet_in_frame;
uint8_t simulcastIdx; // Index if the simulcast encoder creating
// this frame, 0 if not using simulcast.
VideoCodecType codec;
RTPVideoTypeHeader codecHeader;
};
struct WebRtcRTPHeader {
RTPVideoHeader& video_header() { return video; }
const RTPVideoHeader& video_header() const { return video; }

14
modules/rtp_rtcp/BUILD.gn
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@ -258,6 +258,20 @@ rtc_source_set("rtcp_transceiver") {
]
}
rtc_source_set("rtp_video_header") {
visibility = [ "*" ]
sources = [
"source/rtp_video_header.cc",
"source/rtp_video_header.h",
]
deps = [
"../../:webrtc_common",
"../../api/video:video_frame",
"../../modules/video_coding:codec_globals_headers",
"//third_party/abseil-cpp/absl/types:variant",
]
}
rtc_source_set("fec_test_helper") {
testonly = true
sources = [

1
modules/rtp_rtcp/source/fec_test_helper.cc
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@ -123,7 +123,6 @@ std::unique_ptr<AugmentedPacket> AugmentedPacketGenerator::NextPacket(
for (size_t i = 0; i < length; ++i)
packet->data[i + kRtpHeaderSize] = offset + i;
packet->length = length + kRtpHeaderSize;
memset(&packet->header, 0, sizeof(WebRtcRTPHeader));
packet->header.frameType = kVideoFrameDelta;
packet->header.header.headerLength = kRtpHeaderSize;
packet->header.header.markerBit = (num_packets_ == 1);

3
modules/rtp_rtcp/source/rtp_receiver_impl.cc
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@ -159,8 +159,7 @@ bool RtpReceiverImpl::IncomingRtpPacket(const RTPHeader& rtp_header,
return false;
}
WebRtcRTPHeader webrtc_rtp_header;
memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
WebRtcRTPHeader webrtc_rtp_header{};
webrtc_rtp_header.header = rtp_header;
CheckCSRC(webrtc_rtp_header);

8
modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -1682,7 +1682,7 @@ TEST_P(RtpSenderVideoTest, KeyFrameHasCVO) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoRotation, kVideoRotationExtensionId));
RTPVideoHeader hdr = {0};
RTPVideoHeader hdr;
hdr.rotation = kVideoRotation_0;
rtp_sender_video_->SendVideo(kVideoCodecGeneric, kVideoFrameKey, kPayload,
kTimestamp, 0, kFrame, sizeof(kFrame), nullptr,
@ -1704,7 +1704,7 @@ TEST_P(RtpSenderVideoTest, TimingFrameHasPacketizationTimstampSet) {
const int64_t kCaptureTimestamp = fake_clock_.TimeInMilliseconds();
RTPVideoHeader hdr = {0};
RTPVideoHeader hdr;
hdr.video_timing.flags = VideoSendTiming::kTriggeredByTimer;
hdr.video_timing.encode_start_delta_ms = kEncodeStartDeltaMs;
hdr.video_timing.encode_finish_delta_ms = kEncodeFinishDeltaMs;
@ -1727,7 +1727,7 @@ TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenChanged) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoRotation, kVideoRotationExtensionId));
RTPVideoHeader hdr = {0};
RTPVideoHeader hdr;
hdr.rotation = kVideoRotation_90;
EXPECT_TRUE(rtp_sender_video_->SendVideo(
kVideoCodecGeneric, kVideoFrameKey, kPayload, kTimestamp, 0, kFrame,
@ -1749,7 +1749,7 @@ TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenNonZero) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoRotation, kVideoRotationExtensionId));
RTPVideoHeader hdr = {0};
RTPVideoHeader hdr;
hdr.rotation = kVideoRotation_90;
EXPECT_TRUE(rtp_sender_video_->SendVideo(
kVideoCodecGeneric, kVideoFrameKey, kPayload, kTimestamp, 0, kFrame,

27
modules/rtp_rtcp/source/rtp_video_header.cc Normal file
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@ -0,0 +1,27 @@
/*
* Copyright (c) 2018 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 "modules/rtp_rtcp/source/rtp_video_header.h"
namespace webrtc {
RTPVideoHeader::RTPVideoHeader()
: width(),
height(),
rotation(),
playout_delay(),
content_type(),
video_timing(),
is_first_packet_in_frame(),
simulcastIdx(),
codec() {}
RTPVideoHeader::RTPVideoHeader(const RTPVideoHeader& other) = default;
} // namespace webrtc

88
modules/rtp_rtcp/source/rtp_video_header.h Normal file
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@ -0,0 +1,88 @@
/*
* Copyright (c) 2018 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.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
#include "absl/types/variant.h"
#include "api/video/video_content_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
namespace webrtc {
using RTPVideoTypeHeader =
absl::variant<RTPVideoHeaderVP8, RTPVideoHeaderVP9, RTPVideoHeaderH264>;
struct RTPVideoHeader {
RTPVideoHeader();
RTPVideoHeader(const RTPVideoHeader& other);
// TODO(philipel): Remove when downstream projects have been updated.
RTPVideoHeaderVP8& vp8() {
if (!absl::holds_alternative<RTPVideoHeaderVP8>(video_type_header))
video_type_header.emplace<RTPVideoHeaderVP8>();
return absl::get<RTPVideoHeaderVP8>(video_type_header);
}
// TODO(philipel): Remove when downstream projects have been updated.
const RTPVideoHeaderVP8& vp8() const {
if (!absl::holds_alternative<RTPVideoHeaderVP8>(video_type_header))
video_type_header.emplace<RTPVideoHeaderVP8>();
return absl::get<RTPVideoHeaderVP8>(video_type_header);
}
// TODO(philipel): Remove when downstream projects have been updated.
RTPVideoHeaderVP9& vp9() {
if (!absl::holds_alternative<RTPVideoHeaderVP9>(video_type_header))
video_type_header.emplace<RTPVideoHeaderVP9>();
return absl::get<RTPVideoHeaderVP9>(video_type_header);
}
// TODO(philipel): Remove when downstream projects have been updated.
const RTPVideoHeaderVP9& vp9() const {
if (!absl::holds_alternative<RTPVideoHeaderVP9>(video_type_header))
video_type_header.emplace<RTPVideoHeaderVP9>();
return absl::get<RTPVideoHeaderVP9>(video_type_header);
}
// TODO(philipel): Remove when downstream projects have been updated.
RTPVideoHeaderH264& h264() {
if (!absl::holds_alternative<RTPVideoHeaderH264>(video_type_header))
video_type_header.emplace<RTPVideoHeaderH264>();
return absl::get<RTPVideoHeaderH264>(video_type_header);
}
// TODO(philipel): Remove when downstream projects have been updated.
const RTPVideoHeaderH264& h264() const {
if (!absl::holds_alternative<RTPVideoHeaderH264>(video_type_header))
video_type_header.emplace<RTPVideoHeaderH264>();
return absl::get<RTPVideoHeaderH264>(video_type_header);
}
uint16_t width;
uint16_t height;
VideoRotation rotation;
PlayoutDelay playout_delay;
VideoContentType content_type;
VideoSendTiming video_timing;
bool is_first_packet_in_frame;
uint8_t simulcastIdx;
VideoCodecType codec;
// TODO(philipel): remove mutable when downstream projects have been updated.
mutable RTPVideoTypeHeader video_type_header;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_

5
modules/video_coding/BUILD.gn
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@ -179,6 +179,7 @@ rtc_static_library("video_coding") {
"../rtp_rtcp:rtp_rtcp_format",
"../utility:utility",
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@ -221,6 +222,10 @@ rtc_source_set("codec_globals_headers") {
"codecs/vp8/include/vp8_globals.h",
"codecs/vp9/include/vp9_globals.h",
]
deps = [
"../../rtc_base:checks",
]
}
rtc_source_set("video_coding_utility") {

2
modules/video_coding/codecs/h264/include/h264_globals.h
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@ -16,6 +16,8 @@
#include <string>
#include "rtc_base/checks.h"
namespace webrtc {
// The packetization types that we support: single, aggregated, and fragmented.

2
modules/video_coding/frame_object.cc
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@ -161,7 +161,7 @@ absl::optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
if (!packet)
return absl::nullopt;
return packet->video_header.codecHeader;
return packet->video_header.video_type_header;
}
} // namespace video_coding

2
modules/video_coding/packet.cc
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@ -86,7 +86,7 @@ void VCMPacket::Reset() {
insertStartCode = false;
width = 0;
height = 0;
memset(&video_header, 0, sizeof(RTPVideoHeader));
video_header = {};
}
void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader) {

7
modules/video_coding/rtp_frame_reference_finder.cc
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@ -13,6 +13,7 @@
#include <algorithm>
#include <limits>
#include "absl/types/variant.h"
#include "modules/video_coding/frame_object.h"
#include "modules/video_coding/packet_buffer.h"
#include "rtc_base/checks.h"
@ -247,7 +248,8 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
return kDrop;
}
const RTPVideoHeaderVP8& codec_header = rtp_codec_header->VP8;
const RTPVideoHeaderVP8& codec_header =
absl::get<RTPVideoHeaderVP8>(*rtp_codec_header);
if (codec_header.pictureId == kNoPictureId ||
codec_header.temporalIdx == kNoTemporalIdx ||
@ -398,7 +400,8 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp9(
return kDrop;
}
const RTPVideoHeaderVP9& codec_header = rtp_codec_header->VP9;
const RTPVideoHeaderVP9& codec_header =
absl::get<RTPVideoHeaderVP9>(*rtp_codec_header);
if (codec_header.picture_id == kNoPictureId ||
codec_header.temporal_idx == kNoTemporalIdx ||

1
video/BUILD.gn
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@ -72,6 +72,7 @@ rtc_static_library("video") {
"../call:rtp_interfaces",
"../call:video_stream_api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/rtp_rtcp:rtp_video_header",
"../modules/video_coding:codec_globals_headers",
"../modules/video_coding:nack_module",
"../modules/video_coding:packet",

1
video/payload_router.cc
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@ -230,7 +230,6 @@ EncodedImageCallback::Result PayloadRouter::OnEncodedImage(
return Result(Result::ERROR_SEND_FAILED);
RTPVideoHeader rtp_video_header;
memset(&rtp_video_header, 0, sizeof(RTPVideoHeader));
if (codec_specific_info)
CopyCodecSpecific(codec_specific_info, &rtp_video_header);

2
video/payload_router.h
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@ -16,6 +16,7 @@
#include "api/video_codecs/video_encoder.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
@ -24,7 +25,6 @@ namespace webrtc {
class RTPFragmentationHeader;
class RtpRtcp;
struct RTPVideoHeader;
// Currently only VP8/VP9 specific.
struct RtpPayloadState {