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Frame marking RTP header extension (PART 1: implement extension)

Browse Source 
Bug: webrtc:7765
Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb
Reviewed-on: https://webrtc-review.googlesource.com/85200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24695}
This commit is contained in:
Johnny Lee
2018-09-11 16:50:49 -04:00
committed by Commit Bot
parent bfd412ef71
commit e0c8b230e7
22 changed files with 191 additions and 2 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
AUTHORS
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@ -90,6 +90,7 @@ Temasys Communications <*@temasys.io>
The Chromium Authors <*@chromium.org>
The WebRTC Authors <*@webrtc.org>
Videxio AS <*@videxio.com>
Vidyo, Inc. <*@vidyo.com>
Vonage Holdings Corp. <*@vonage.com>
Wire Swiss GmbH <*@wire.com>
Miguel Paris <mparisdiaz@gmail.com>

4
api/rtp_headers.cc
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@ -34,7 +34,9 @@ RTPHeaderExtension::RTPHeaderExtension()
videoRotation(kVideoRotation_0),
hasVideoContentType(false),
videoContentType(VideoContentType::UNSPECIFIED),
has_video_timing(false) {}
has_video_timing(false),
has_frame_marking(false),
frame_marking({false, false, false, false, false, 0xFF, 0, 0}) {}
RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
default;

4
api/rtp_headers.h
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@ -19,6 +19,7 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame_marking.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
@ -116,6 +117,9 @@ struct RTPHeaderExtension {
bool has_video_timing;
VideoSendTiming video_timing;
bool has_frame_marking;
FrameMarking frame_marking;
PlayoutDelay playout_delay = {-1, -1};
// For identification of a stream when ssrc is not signaled. See

7
api/rtpparameters.cc
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@ -129,6 +129,10 @@ const int RtpExtension::kVideoTimingDefaultId = 8;
const char RtpExtension::kMidUri[] = "urn:ietf:params:rtp-hdrext:sdes:mid";
const int RtpExtension::kMidDefaultId = 9;
const char RtpExtension::kFrameMarkingUri[] =
"http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07";
const int RtpExtension::kFrameMarkingDefaultId = 10;
const char RtpExtension::kEncryptHeaderExtensionsUri[] =
"urn:ietf:params:rtp-hdrext:encrypt";
@ -149,7 +153,8 @@ bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
uri == webrtc::RtpExtension::kPlayoutDelayUri ||
uri == webrtc::RtpExtension::kVideoContentTypeUri ||
uri == webrtc::RtpExtension::kVideoTimingUri ||
uri == webrtc::RtpExtension::kMidUri;
uri == webrtc::RtpExtension::kMidUri ||
uri == webrtc::RtpExtension::kFrameMarkingUri;
}
bool RtpExtension::IsEncryptionSupported(const std::string& uri) {

4
api/rtpparameters.h
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@ -282,6 +282,10 @@ struct RtpExtension {
static const char kVideoTimingUri[];
static const int kVideoTimingDefaultId;
// Header extension for video frame marking.
static const char kFrameMarkingUri[];
static const int kFrameMarkingDefaultId;
// Header extension for transport sequence number, see url for details:
// http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
static const char kTransportSequenceNumberUri[];

1
api/video/BUILD.gn
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@ -19,6 +19,7 @@ rtc_source_set("video_frame") {
"video_frame.h",
"video_frame_buffer.cc",
"video_frame_buffer.h",
"video_frame_marking.h",
"video_rotation.h",
"video_sink_interface.h",
"video_source_interface.cc",

29
api/video/video_frame_marking.h Normal file
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@ -0,0 +1,29 @@
/*
* 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 API_VIDEO_VIDEO_FRAME_MARKING_H_
#define API_VIDEO_VIDEO_FRAME_MARKING_H_
namespace webrtc {
struct FrameMarking {
bool start_of_frame;
bool end_of_frame;
bool independent_frame;
bool discardable_frame;
bool base_layer_sync;
uint8_t temporal_id;
uint8_t layer_id;
uint8_t tl0_pic_idx;
};
} // namespace webrtc
#endif // API_VIDEO_VIDEO_FRAME_MARKING_H_

2
logging/rtc_event_log/rtc_event_log2text.cc
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@ -157,6 +157,8 @@ webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
webrtc::RtpExtension::kVideoContentTypeDefaultId);
default_map.Register<webrtc::VideoTimingExtension>(
webrtc::RtpExtension::kVideoTimingDefaultId);
default_map.Register<webrtc::FrameMarkingExtension>(
webrtc::RtpExtension::kFrameMarkingDefaultId);
default_map.Register<webrtc::TransportSequenceNumber>(
webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
default_map.Register<webrtc::PlayoutDelayLimits>(

2
logging/rtc_event_log/rtc_event_log_parser_new.cc
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@ -228,6 +228,8 @@ webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
webrtc::RtpExtension::kVideoContentTypeDefaultId);
default_map.Register<VideoTimingExtension>(
webrtc::RtpExtension::kVideoTimingDefaultId);
default_map.Register<FrameMarkingExtension>(
webrtc::RtpExtension::kFrameMarkingDefaultId);
default_map.Register<TransportSequenceNumber>(
webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
default_map.Register<PlayoutDelayLimits>(

3
media/engine/webrtcvideoengine.cc
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@ -509,6 +509,9 @@ RtpCapabilities WebRtcVideoEngine::GetCapabilities() const {
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri,
webrtc::RtpExtension::kVideoTimingDefaultId));
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kFrameMarkingUri,
webrtc::RtpExtension::kFrameMarkingDefaultId));
// TODO(bugs.webrtc.org/4050): Add MID header extension as capability once MID
// demuxing is completed.
// capabilities.header_extensions.push_back(webrtc::RtpExtension(

1
modules/rtp_rtcp/include/rtp_rtcp_defines.h
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@ -103,6 +103,7 @@ enum RTPExtensionType {
kRtpExtensionPlayoutDelay,
kRtpExtensionVideoContentType,
kRtpExtensionVideoTiming,
kRtpExtensionFrameMarking,
kRtpExtensionRtpStreamId,
kRtpExtensionRepairedRtpStreamId,
kRtpExtensionMid,

1
modules/rtp_rtcp/source/rtp_header_extension_map.cc
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@ -38,6 +38,7 @@ constexpr ExtensionInfo kExtensions[] = {
CreateExtensionInfo<PlayoutDelayLimits>(),
CreateExtensionInfo<VideoContentTypeExtension>(),
CreateExtensionInfo<VideoTimingExtension>(),
CreateExtensionInfo<FrameMarkingExtension>(),
CreateExtensionInfo<RtpStreamId>(),
CreateExtensionInfo<RepairedRtpStreamId>(),
CreateExtensionInfo<RtpMid>(),

80
modules/rtp_rtcp/source/rtp_header_extensions.cc
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@ -350,6 +350,86 @@ bool VideoTimingExtension::Write(rtc::ArrayView<uint8_t> data,
return true;
}
// Frame Marking.
//
// Meta-information about an RTP stream outside the encrypted media payload,
// useful for an RTP switch to do codec-agnostic selective forwarding
// without decrypting the payload.
//
// For non-scalable streams:
// 0 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | L = 0 |S|E|I|D|0 0 0 0|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// For scalable streams:
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | L = 2 |S|E|I|D|B| TID | LID | TL0PICIDX |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr RTPExtensionType FrameMarkingExtension::kId;
constexpr const char FrameMarkingExtension::kUri[];
bool FrameMarkingExtension::IsScalable(uint8_t temporal_id, uint8_t layer_id) {
return temporal_id != kNoTemporalIdx || layer_id != kNoSpatialIdx;
}
bool FrameMarkingExtension::Parse(rtc::ArrayView<const uint8_t> data,
FrameMarking* frame_marking) {
RTC_DCHECK(frame_marking);
if (data.size() != 1 && data.size() != 3)
return false;
frame_marking->start_of_frame = (data[0] & 0x80) != 0;
frame_marking->end_of_frame = (data[0] & 0x40) != 0;
frame_marking->independent_frame = (data[0] & 0x20) != 0;
frame_marking->discardable_frame = (data[0] & 0x10) != 0;
if (data.size() == 3) {
frame_marking->base_layer_sync = (data[0] & 0x08) != 0;
frame_marking->temporal_id = data[0] & 0x7;
frame_marking->layer_id = data[1];
frame_marking->tl0_pic_idx = data[2];
} else {
// non-scalable
frame_marking->base_layer_sync = false;
frame_marking->temporal_id = kNoTemporalIdx;
frame_marking->layer_id = kNoSpatialIdx;
frame_marking->tl0_pic_idx = 0;
}
return true;
}
size_t FrameMarkingExtension::ValueSize(const FrameMarking& frame_marking) {
if (IsScalable(frame_marking.temporal_id, frame_marking.layer_id))
return 3;
else
return 1;
}
bool FrameMarkingExtension::Write(rtc::ArrayView<uint8_t> data,
const FrameMarking& frame_marking) {
RTC_DCHECK_GE(data.size(), 1);
RTC_CHECK_LE(frame_marking.temporal_id, 0x07);
data[0] = frame_marking.start_of_frame ? 0x80 : 0x00;
data[0] |= frame_marking.end_of_frame ? 0x40 : 0x00;
data[0] |= frame_marking.independent_frame ? 0x20 : 0x00;
data[0] |= frame_marking.discardable_frame ? 0x10 : 0x00;
if (IsScalable(frame_marking.temporal_id, frame_marking.layer_id)) {
RTC_DCHECK_EQ(data.size(), 3);
data[0] |= frame_marking.base_layer_sync ? 0x08 : 0x00;
data[0] |= frame_marking.temporal_id & 0x07;
data[1] = frame_marking.layer_id;
data[2] = frame_marking.tl0_pic_idx;
}
return true;
}
bool BaseRtpStringExtension::Parse(rtc::ArrayView<const uint8_t> data,
StringRtpHeaderExtension* str) {
if (data.empty() || data[0] == 0) // Valid string extension can't be empty.

16
modules/rtp_rtcp/source/rtp_header_extensions.h
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@ -153,6 +153,22 @@ class VideoTimingExtension {
uint8_t idx);
};
class FrameMarkingExtension {
public:
static constexpr RTPExtensionType kId = kRtpExtensionFrameMarking;
static constexpr const char kUri[] =
"http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07";
static bool Parse(rtc::ArrayView<const uint8_t> data,
FrameMarking* frame_marking);
static size_t ValueSize(const FrameMarking& frame_marking);
static bool Write(rtc::ArrayView<uint8_t> data,
const FrameMarking& frame_marking);
private:
static bool IsScalable(uint8_t temporal_id, uint8_t layer_id);
};
// Base extension class for RTP header extensions which are strings.
// Subclasses must defined kId and kUri static constexpr members.
class BaseRtpStringExtension {

2
modules/rtp_rtcp/source/rtp_packet_received.cc
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@ -61,6 +61,8 @@ void RtpPacketReceived::GetHeader(RTPHeader* header) const {
&header->extension.videoContentType);
header->extension.has_video_timing =
GetExtension<VideoTimingExtension>(&header->extension.video_timing);
header->extension.has_frame_marking =
GetExtension<FrameMarkingExtension>(&header->extension.frame_marking);
GetExtension<RtpStreamId>(&header->extension.stream_id);
GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
GetExtension<RtpMid>(&header->extension.mid);

6
modules/rtp_rtcp/source/rtp_receiver_video.cc
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@ -74,6 +74,7 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
rtp_header->video_header().rotation = kVideoRotation_0;
rtp_header->video_header().content_type = VideoContentType::UNSPECIFIED;
rtp_header->video_header().video_timing.flags = VideoSendTiming::kInvalid;
rtp_header->video_header().frame_marking.temporal_id = kNoTemporalIdx;
// Retrieve the video rotation information.
if (rtp_header->header.extension.hasVideoRotation) {
@ -94,6 +95,11 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
rtp_header->video_header().playout_delay =
rtp_header->header.extension.playout_delay;
if (rtp_header->header.extension.has_frame_marking) {
rtp_header->video_header().frame_marking =
rtp_header->header.extension.frame_marking;
}
return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
parsed_payload.payload_length,
rtp_header) == 0

2
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -50,6 +50,8 @@ RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
return kRtpExtensionVideoContentType;
if (extension == RtpExtension::kVideoTimingUri)
return kRtpExtensionVideoTiming;
if (extension == RtpExtension::kFrameMarkingUri)
return kRtpExtensionFrameMarking;
if (extension == RtpExtension::kMidUri)
return kRtpExtensionMid;
RTC_NOTREACHED() << "Looking up unsupported RTP extension.";

13
modules/rtp_rtcp/source/rtp_utility.cc
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@ -236,6 +236,10 @@ bool RtpHeaderParser::Parse(
header->extension.has_video_timing = false;
header->extension.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false};
header->extension.has_frame_marking = false;
header->extension.frame_marking = {false, false, false, false, false,
kNoTemporalIdx, 0, 0};
if (X) {
/* RTP header extension, RFC 3550.
0 1 2 3
@ -454,6 +458,15 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
&header->extension.video_timing);
break;
}
case kRtpExtensionFrameMarking: {
if (!FrameMarkingExtension::Parse(rtc::MakeArrayView(ptr, len + 1),
&header->extension.frame_marking)) {
RTC_LOG(LS_WARNING) << "Incorrect frame marking len: " << len;
return;
}
header->extension.has_frame_marking = true;
break;
}
case kRtpExtensionRtpStreamId: {
header->extension.stream_id.Set(rtc::MakeArrayView(ptr, len + 1));
break;

2
modules/rtp_rtcp/source/rtp_video_header.h
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@ -13,6 +13,7 @@
#include "absl/container/inlined_vector.h"
#include "absl/types/variant.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame_marking.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "common_types.h" // NOLINT(build/include)
@ -56,6 +57,7 @@ struct RTPVideoHeader {
PlayoutDelay playout_delay;
VideoSendTiming video_timing;
FrameMarking frame_marking;
RTPVideoTypeHeader video_type_header;
};

8
modules/video_coding/frame_object.cc
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@ -159,5 +159,13 @@ absl::optional<RTPVideoHeader> RtpFrameObject::GetRtpVideoHeader() const {
return packet->video_header;
}
absl::optional<FrameMarking> RtpFrameObject::GetFrameMarking() const {
rtc::CritScope lock(&packet_buffer_->crit_);
VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
if (!packet)
return absl::nullopt;
return packet->video_header.frame_marking;
}
} // namespace video_coding
} // namespace webrtc

1
modules/video_coding/frame_object.h
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@ -41,6 +41,7 @@ class RtpFrameObject : public EncodedFrame {
int64_t RenderTime() const override;
bool delayed_by_retransmission() const override;
absl::optional<RTPVideoHeader> GetRtpVideoHeader() const;
absl::optional<FrameMarking> GetFrameMarking() const;
private:
rtc::scoped_refptr<PacketBuffer> packet_buffer_;

4
test/fuzzers/rtp_packet_fuzzer.cc
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@ -95,6 +95,10 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
VideoSendTiming timing;
packet.GetExtension<VideoTimingExtension>(&timing);
break;
case kRtpExtensionFrameMarking:
FrameMarking frame_marking;
packet.GetExtension<FrameMarkingExtension>(&frame_marking);
break;
case kRtpExtensionRtpStreamId: {
std::string rsid;
packet.GetExtension<RtpStreamId>(&rsid);