Fix circular dependencies in webrtc_common.
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 592039f..94dd865 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -59,6 +59,8 @@
"proxy.h",
"rtcerror.cc",
"rtcerror.h",
+ "rtp_headers.cc",
+ "rtp_headers.h",
"rtpparameters.cc",
"rtpparameters.h",
"rtpreceiverinterface.h",
@@ -84,6 +86,7 @@
]
deps = [
+ ":array_view",
":optional",
":rtc_stats_api",
":video_frame_api",
@@ -93,10 +96,14 @@
# Basically, don't add stuff here. You might break sensitive downstream
# targets like pnacl. API should not depend on anything outside of this
# file, really. All these should arguably go away in time.
+ "..:typedefs",
"..:webrtc_common",
"../modules/audio_processing:audio_processing_statistics",
+ "../rtc_base:checks",
+ "../rtc_base:deprecation",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
+ "../rtc_base:stringutils",
]
# This is needed until bugs.webrtc.org/7504 is removed so this target can
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index d2c074d..01dc124 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -29,7 +29,9 @@
"..:array_view",
"..:optional",
"../..:webrtc_common",
+ "../../:typedefs",
"../../rtc_base:checks",
+ "../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:sanitizer",
]
diff --git a/api/ortc/rtptransportinterface.h b/api/ortc/rtptransportinterface.h
index 3c58fad..716a297 100644
--- a/api/ortc/rtptransportinterface.h
+++ b/api/ortc/rtptransportinterface.h
@@ -16,6 +16,7 @@
#include "api/optional.h"
#include "api/ortc/packettransportinterface.h"
#include "api/rtcerror.h"
+#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
diff --git a/api/rtp_headers.cc b/api/rtp_headers.cc
new file mode 100644
index 0000000..a0b1a15
--- /dev/null
+++ b/api/rtp_headers.cc
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2017 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 "api/rtp_headers.h"
+
+#include <string.h>
+#include <algorithm>
+#include <limits>
+#include <type_traits>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/stringutils.h"
+
+namespace webrtc {
+
+RTPHeaderExtension::RTPHeaderExtension()
+ : hasTransmissionTimeOffset(false),
+ transmissionTimeOffset(0),
+ hasAbsoluteSendTime(false),
+ absoluteSendTime(0),
+ hasTransportSequenceNumber(false),
+ transportSequenceNumber(0),
+ hasAudioLevel(false),
+ voiceActivity(false),
+ audioLevel(0),
+ hasVideoRotation(false),
+ videoRotation(kVideoRotation_0),
+ hasVideoContentType(false),
+ videoContentType(VideoContentType::UNSPECIFIED),
+ has_video_timing(false) {}
+
+RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
+ default;
+
+RTPHeaderExtension& RTPHeaderExtension::operator=(
+ const RTPHeaderExtension& other) = default;
+
+RTPHeader::RTPHeader()
+ : markerBit(false),
+ payloadType(0),
+ sequenceNumber(0),
+ timestamp(0),
+ ssrc(0),
+ numCSRCs(0),
+ arrOfCSRCs(),
+ paddingLength(0),
+ headerLength(0),
+ payload_type_frequency(0),
+ extension() {}
+
+RTPHeader::RTPHeader(const RTPHeader& other) = default;
+
+RTPHeader& RTPHeader::operator=(const RTPHeader& other) = default;
+
+} // namespace webrtc
diff --git a/api/rtp_headers.h b/api/rtp_headers.h
new file mode 100644
index 0000000..2624a33
--- /dev/null
+++ b/api/rtp_headers.h
@@ -0,0 +1,178 @@
+/*
+ * Copyright (c) 2017 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_RTP_HEADERS_H_
+#define API_RTP_HEADERS_H_
+
+#include <stddef.h>
+#include <string.h>
+#include <ostream>
+#include <string>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/optional.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_timing.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/deprecation.h"
+#include "common_types.h" // NOLINT(build/include)
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+// Class to represent the value of RTP header extensions that are
+// variable-length strings (e.g., RtpStreamId and RtpMid).
+// Unlike std::string, it can be copied with memcpy and cleared with memset.
+//
+// Empty value represents unset header extension (use empty() to query).
+class StringRtpHeaderExtension {
+ public:
+ // String RTP header extensions are limited to 16 bytes because it is the
+ // maximum length that can be encoded with one-byte header extensions.
+ static constexpr size_t kMaxSize = 16;
+
+ static bool IsLegalName(rtc::ArrayView<const char> name);
+
+ StringRtpHeaderExtension() { value_[0] = 0; }
+ explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) {
+ Set(value.data(), value.size());
+ }
+ StringRtpHeaderExtension(const StringRtpHeaderExtension&) = default;
+ StringRtpHeaderExtension& operator=(const StringRtpHeaderExtension&) =
+ default;
+
+ bool empty() const { return value_[0] == 0; }
+ const char* data() const { return value_; }
+ size_t size() const { return strnlen(value_, kMaxSize); }
+
+ void Set(rtc::ArrayView<const uint8_t> value) {
+ Set(reinterpret_cast<const char*>(value.data()), value.size());
+ }
+ void Set(const char* data, size_t size);
+
+ friend bool operator==(const StringRtpHeaderExtension& lhs,
+ const StringRtpHeaderExtension& rhs) {
+ return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0;
+ }
+ friend bool operator!=(const StringRtpHeaderExtension& lhs,
+ const StringRtpHeaderExtension& rhs) {
+ return !(lhs == rhs);
+ }
+
+ private:
+ char value_[kMaxSize];
+};
+
+// StreamId represents RtpStreamId which is a string.
+typedef StringRtpHeaderExtension StreamId;
+
+// Mid represents RtpMid which is a string.
+typedef StringRtpHeaderExtension Mid;
+
+struct RTPHeaderExtension {
+ RTPHeaderExtension();
+ RTPHeaderExtension(const RTPHeaderExtension& other);
+ RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
+
+ bool hasTransmissionTimeOffset;
+ int32_t transmissionTimeOffset;
+ bool hasAbsoluteSendTime;
+ uint32_t absoluteSendTime;
+ bool hasTransportSequenceNumber;
+ uint16_t transportSequenceNumber;
+
+ // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
+ // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
+ bool hasAudioLevel;
+ bool voiceActivity;
+ uint8_t audioLevel;
+
+ // For Coordination of Video Orientation. See
+ // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
+ // ts_126114v120700p.pdf
+ bool hasVideoRotation;
+ VideoRotation videoRotation;
+
+ // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
+ // a corresponding bool flag.
+ bool hasVideoContentType;
+ VideoContentType videoContentType;
+
+ bool has_video_timing;
+ VideoSendTiming video_timing;
+
+ PlayoutDelay playout_delay = {-1, -1};
+
+ // For identification of a stream when ssrc is not signaled. See
+ // https://tools.ietf.org/html/draft-ietf-avtext-rid-09
+ // TODO(danilchap): Update url from draft to release version.
+ StreamId stream_id;
+ StreamId repaired_stream_id;
+
+ // For identifying the media section used to interpret this RTP packet. See
+ // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
+ Mid mid;
+};
+
+struct RTPHeader {
+ RTPHeader();
+ RTPHeader(const RTPHeader& other);
+ RTPHeader& operator=(const RTPHeader& other);
+
+ bool markerBit;
+ uint8_t payloadType;
+ uint16_t sequenceNumber;
+ uint32_t timestamp;
+ uint32_t ssrc;
+ uint8_t numCSRCs;
+ uint32_t arrOfCSRCs[kRtpCsrcSize];
+ size_t paddingLength;
+ size_t headerLength;
+ int payload_type_frequency;
+ RTPHeaderExtension extension;
+};
+
+// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
+// RTCP mode is described by RFC 5506.
+enum class RtcpMode { kOff, kCompound, kReducedSize };
+
+enum NetworkState {
+ kNetworkUp,
+ kNetworkDown,
+};
+
+struct RtpKeepAliveConfig final {
+ // If no packet has been sent for |timeout_interval_ms|, send a keep-alive
+ // packet. The keep-alive packet is an empty (no payload) RTP packet with a
+ // payload type of 20 as long as the other end has not negotiated the use of
+ // this value. If this value has already been negotiated, then some other
+ // unused static payload type from table 5 of RFC 3551 shall be used and set
+ // in |payload_type|.
+ int64_t timeout_interval_ms = -1;
+ uint8_t payload_type = 20;
+
+ bool operator==(const RtpKeepAliveConfig& o) const {
+ return timeout_interval_ms == o.timeout_interval_ms &&
+ payload_type == o.payload_type;
+ }
+ bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
+};
+
+// Currently only VP8/VP9 specific.
+struct RtpPayloadState {
+ int16_t picture_id = -1;
+};
+
+} // namespace webrtc
+
+#endif // API_RTP_HEADERS_H_
diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
index f3cb7f1..6f3623d 100644
--- a/api/video_codecs/BUILD.gn
+++ b/api/video_codecs/BUILD.gn
@@ -26,6 +26,7 @@
"..:optional",
"..:video_frame_api",
"../..:webrtc_common",
+ "../../:typedefs",
"../../common_video",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",