mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-12 21:30:45 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions

Fix circular dependencies in webrtc_common.

Browse source 
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}
This commit is contained in:
Patrik Höglund 2017-12-15 14:40:10 +01:00 committed by Commit Bot
parent d5d0540b86
commit 3e113438b1
57 changed files with 559 additions and 316 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1
.gn
View file

@ -21,6 +21,7 @@ secondary_source = "//build/secondary/"
# their includes checked for proper dependencies when you run either
# "gn check" or "gn gen --check".
check_targets = [
":webrtc_common",
"//api/*",
"//audio/*",
"//backup/*",

15
BUILD.gn
View file

@ -350,11 +350,24 @@ if (!build_with_chromium) {
}
}
rtc_source_set("typedefs") {
sources = [
"typedefs.h",
]
}
rtc_static_library("webrtc_common") {
sources = [
"common_types.cc",
"common_types.h",
"typedefs.h",
]
deps = [
":typedefs",
"api:array_view",
"api:optional",
"rtc_base:checks",
"rtc_base:deprecation",
"rtc_base:stringutils",
]
if (!build_with_chromium && is_clang) {

7
api/BUILD.gn
View file

@ -59,6 +59,8 @@ rtc_static_library("libjingle_peerconnection_api") {
"proxy.h",
"rtcerror.cc",
"rtcerror.h",
"rtp_headers.cc",
"rtp_headers.h",
"rtpparameters.cc",
"rtpparameters.h",
"rtpreceiverinterface.h",
@ -84,6 +86,7 @@ rtc_static_library("libjingle_peerconnection_api") {
]
deps = [
":array_view",
":optional",
":rtc_stats_api",
":video_frame_api",
@ -93,10 +96,14 @@ rtc_static_library("libjingle_peerconnection_api") {
# 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

2
api/audio_codecs/BUILD.gn
View file

@ -29,7 +29,9 @@ rtc_source_set("audio_codecs_api") {
"..:array_view",
"..:optional",
"../..:webrtc_common",
"../../:typedefs",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:sanitizer",
]

1
api/ortc/rtptransportinterface.h
View file

@ -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 {

62
api/rtp_headers.cc Normal file
View file

@ -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

178
api/rtp_headers.h Normal file
View file

@ -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_

1
api/video_codecs/BUILD.gn
View file

@ -26,6 +26,7 @@ rtc_source_set("video_codecs_api") {
"..:optional",
"..:video_frame_api",
"../..:webrtc_common",
"../../:typedefs",
"../../common_video",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",

1
audio/utility/BUILD.gn
View file

@ -21,6 +21,7 @@ rtc_static_library("audio_frame_operations") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../modules:module_api",
"../../modules/audio_coding:audio_format_conversion",
"../../rtc_base:checks",

4
call/BUILD.gn
View file

@ -24,6 +24,7 @@ rtc_source_set("call_interfaces") {
":rtp_interfaces",
":video_stream_api",
"..:webrtc_common",
"../:typedefs",
"../api:audio_mixer_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
@ -70,6 +71,7 @@ rtc_source_set("rtp_receiver") {
":rtp_interfaces",
"..:webrtc_common",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
@ -176,6 +178,7 @@ rtc_source_set("video_stream_api") {
"../api:optional",
"../api:transport_api",
"../common_video:common_video",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
]
}
@ -204,6 +207,7 @@ if (rtc_include_tests) {
":rtp_sender",
"..:webrtc_common",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../api:mock_audio_mixer",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../logging:rtc_event_log_api",

1
call/flexfec_receive_stream.h
View file

@ -16,6 +16,7 @@
#include <string>
#include <vector>
#include "api/rtp_headers.h"
#include "api/call/transport.h"
#include "api/rtpparameters.h"
#include "call/rtp_packet_sink_interface.h"

1
call/rtcp_demuxer.cc
View file

@ -10,6 +10,7 @@
#include "call/rtcp_demuxer.h"
#include "api/rtp_headers.h"
#include "call/rtcp_packet_sink_interface.h"
#include "call/rtp_rtcp_demuxer_helper.h"
#include "common_types.h" // NOLINT(build/include)

1
call/rtcp_demuxer_unittest.cc
View file

@ -13,6 +13,7 @@
#include <memory>
#include <set>
#include "api/rtp_headers.h"
#include "call/rtcp_packet_sink_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"

4
call/video_receive_stream.h
View file

@ -16,12 +16,16 @@
#include <string>
#include <vector>
#include "api/rtp_headers.h"
#include "api/call/transport.h"
#include "api/rtpparameters.h"
#include "api/video/video_content_type.h"
#include "api/video/video_timing.h"
#include "call/rtp_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "common_video/include/frame_callback.h"
#include "media/base/videosinkinterface.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/platform_file.h"
namespace webrtc {

2
call/video_send_stream.h
View file

@ -18,12 +18,14 @@
#include "api/call/transport.h"
#include "api/rtpparameters.h"
#include "api/rtp_headers.h"
#include "call/rtp_config.h"
#include "call/video_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "common_video/include/frame_callback.h"
#include "media/base/videosinkinterface.h"
#include "media/base/videosourceinterface.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/platform_file.h"
namespace webrtc {

5
common_audio/BUILD.gn
View file

@ -62,6 +62,7 @@ rtc_static_library("common_audio") {
deps = [
":sinc_resampler",
"..:webrtc_common",
"../:typedefs",
"../api:optional",
"../rtc_base:checks",
"../rtc_base:gtest_prod",
@ -223,6 +224,7 @@ rtc_source_set("common_audio_c") {
":common_audio_c_arm_asm",
":common_audio_cc",
"..:webrtc_common",
"../:typedefs",
"../rtc_base:checks",
"../rtc_base:compile_assert_c",
"../rtc_base:rtc_base_approved",
@ -241,6 +243,7 @@ rtc_source_set("common_audio_cc") {
public_configs = [ ":common_audio_config" ]
deps = [
"..:webrtc_common",
"../:typedefs",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
]
@ -252,6 +255,7 @@ rtc_source_set("sinc_resampler") {
]
deps = [
"..:webrtc_common",
"../:typedefs",
"../rtc_base:gtest_prod",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
@ -447,6 +451,7 @@ if (rtc_include_tests) {
":fir_filter_factory",
":sinc_resampler",
"..:webrtc_common",
"../:typedefs",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",

60
common_types.cc
View file

@ -20,66 +20,6 @@
namespace webrtc {
StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {}
constexpr size_t StreamId::kMaxSize;
bool StreamId::IsLegalName(rtc::ArrayView<const char> name) {
return (name.size() <= kMaxSize && name.size() > 0 &&
std::all_of(name.data(), name.data() + name.size(), isalnum));
}
void StreamId::Set(const char* data, size_t size) {
// If |data| contains \0, the stream id size might become less than |size|.
RTC_CHECK_LE(size, kMaxSize);
memcpy(value_, data, size);
if (size < kMaxSize)
value_[size] = 0;
}
// StreamId is used as member of RTPHeader that is sometimes copied with memcpy
// and thus assume trivial destructibility.
static_assert(std::is_trivially_destructible<StreamId>::value, "");
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;
VideoCodec::VideoCodec()
: codecType(kVideoCodecUnknown),
plName(),

241
common_types.h
View file

@ -19,9 +19,6 @@
#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 "typedefs.h" // NOLINT(build/include)
@ -685,244 +682,6 @@ struct PlayoutDelay {
int max_ms;
};
// 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;
};
struct RtpPacketCounter {
RtpPacketCounter()
: header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
void Add(const RtpPacketCounter& other) {
header_bytes += other.header_bytes;
payload_bytes += other.payload_bytes;
padding_bytes += other.padding_bytes;
packets += other.packets;
}
void Subtract(const RtpPacketCounter& other) {
RTC_DCHECK_GE(header_bytes, other.header_bytes);
header_bytes -= other.header_bytes;
RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
payload_bytes -= other.payload_bytes;
RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
padding_bytes -= other.padding_bytes;
RTC_DCHECK_GE(packets, other.packets);
packets -= other.packets;
}
void AddPacket(size_t packet_length, const RTPHeader& header) {
++packets;
header_bytes += header.headerLength;
padding_bytes += header.paddingLength;
payload_bytes +=
packet_length - (header.headerLength + header.paddingLength);
}
size_t TotalBytes() const {
return header_bytes + payload_bytes + padding_bytes;
}
size_t header_bytes; // Number of bytes used by RTP headers.
size_t payload_bytes; // Payload bytes, excluding RTP headers and padding.
size_t padding_bytes; // Number of padding bytes.
uint32_t packets; // Number of packets.
};
// Data usage statistics for a (rtp) stream.
struct StreamDataCounters {
StreamDataCounters();
void Add(const StreamDataCounters& other) {
transmitted.Add(other.transmitted);
retransmitted.Add(other.retransmitted);
fec.Add(other.fec);
if (other.first_packet_time_ms != -1 &&
(other.first_packet_time_ms < first_packet_time_ms ||
first_packet_time_ms == -1)) {
// Use oldest time.
first_packet_time_ms = other.first_packet_time_ms;
}
}
void Subtract(const StreamDataCounters& other) {
transmitted.Subtract(other.transmitted);
retransmitted.Subtract(other.retransmitted);
fec.Subtract(other.fec);
if (other.first_packet_time_ms != -1 &&
(other.first_packet_time_ms > first_packet_time_ms ||
first_packet_time_ms == -1)) {
// Use youngest time.
first_packet_time_ms = other.first_packet_time_ms;
}
}
int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
}
// Returns the number of bytes corresponding to the actual media payload (i.e.
// RTP headers, padding, retransmissions and fec packets are excluded).
// Note this function does not have meaning for an RTX stream.
size_t MediaPayloadBytes() const {
return transmitted.payload_bytes - retransmitted.payload_bytes -
fec.payload_bytes;
}
int64_t first_packet_time_ms; // Time when first packet is sent/received.
RtpPacketCounter transmitted; // Number of transmitted packets/bytes.
RtpPacketCounter retransmitted; // Number of retransmitted packets/bytes.
RtpPacketCounter fec; // Number of redundancy packets/bytes.
};
// Callback, called whenever byte/packet counts have been updated.
class StreamDataCountersCallback {
public:
virtual ~StreamDataCountersCallback() {}
virtual void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) = 0;
};
// 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 // COMMON_TYPES_H_

1
common_video/include/video_frame.h
View file

@ -16,6 +16,7 @@
// to refactor and clean up related interfaces, at which point it
// should be moved to somewhere under api/.
#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "typedefs.h" // NOLINT(build/include)

3
examples/BUILD.gn
View file

@ -509,6 +509,7 @@ if (is_linux || is_win) {
deps = [
"../api:video_frame_api_i420",
"../rtc_base:checks",
"../rtc_base:stringutils",
]
if (is_win) {
sources += [
@ -569,6 +570,7 @@ if (is_linux || is_win) {
deps = [
"..:webrtc_common",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../rtc_tools:command_line_parser",
]
if (!build_with_chromium && is_clang) {
@ -727,6 +729,7 @@ if (!build_with_chromium) {
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../system_wrappers:field_trial_default",
]
}

1
logging/BUILD.gn
View file

@ -71,6 +71,7 @@ rtc_source_set("rtc_event_log_api") {
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:array_view",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",

1
logging/rtc_event_log/rtc_stream_config.h
View file

@ -14,6 +14,7 @@
#include <string>
#include <vector>
#include "api/rtp_headers.h"
#include "api/rtpparameters.h"
#include "common_types.h" // NOLINT(build/include)

4
media/BUILD.gn
View file

@ -57,6 +57,7 @@ rtc_static_library("rtc_media_base") {
deps = [
"../rtc_base:checks",
"../rtc_base:sanitizer",
"../rtc_base:stringutils",
]
public_deps = []
sources = [
@ -133,6 +134,7 @@ rtc_static_library("rtc_audio_video") {
"../call:call_interfaces",
"../modules/video_coding:video_coding_utility",
"../rtc_base:checks",
"../rtc_base:stringutils",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
"//third_party/libyuv",
@ -347,6 +349,7 @@ if (rtc_include_tests) {
"../modules/video_coding:video_coding_utility",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:stringutils",
]
sources = [
"base/fakemediaengine.h",
@ -438,6 +441,7 @@ if (rtc_include_tests) {
"../pc:rtc_pc",
"../pc:rtc_pc_base",
"../rtc_base:checks",
"../rtc_base:stringutils",
"../test:field_trial",
]
sources = [

1
media/base/mediachannel.h
View file

@ -21,6 +21,7 @@
#include "api/optional.h"
#include "api/rtpparameters.h"
#include "api/rtpreceiverinterface.h"
#include "api/video/video_content_type.h"
#include "api/video/video_timing.h"
#include "call/video_config.h"
#include "media/base/codec.h"

4
modules/BUILD.gn
View file

@ -34,6 +34,7 @@ rtc_source_set("module_api_public") {
]
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:optional",
]
}
@ -46,9 +47,12 @@ rtc_source_set("module_api") {
deps = [
":module_api_public",
"..:webrtc_common",
"../:typedefs",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:video_frame_api",
"../api:video_frame_api_i420",
"../rtc_base:deprecation",
"../rtc_base:rtc_base_approved",
"video_coding:codec_globals_headers",
]

36
modules/audio_coding/BUILD.gn
View file

@ -65,6 +65,7 @@ rtc_static_library("rent_a_codec") {
"acm2/rent_a_codec.h",
]
deps = [
"../../:typedefs",
"../../rtc_base:checks",
"../../api:array_view",
"../../api:optional",
@ -96,6 +97,7 @@ rtc_source_set("audio_coding_module_typedefs") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
]
}
@ -132,6 +134,8 @@ rtc_static_library("audio_coding") {
}
deps = audio_coding_deps + [
"../../rtc_base:deprecation",
"../../:typedefs",
"../../rtc_base:checks",
"../../system_wrappers:metrics_api",
"..:module_api",
@ -177,6 +181,7 @@ rtc_static_library("cng") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
@ -241,6 +246,7 @@ rtc_source_set("g711_c") {
]
deps = [
"../..:webrtc_common",
"../../:typedefs",
]
}
@ -282,6 +288,7 @@ rtc_source_set("g722_c") {
]
deps = [
"../..:webrtc_common",
"../../:typedefs",
]
}
@ -461,6 +468,7 @@ rtc_source_set("ilbc_c") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../rtc_base:checks",
@ -569,6 +577,7 @@ rtc_static_library("isac_c") {
deps = [
":isac_common",
"../..:webrtc_common",
"../../:typedefs",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:compile_assert_c",
@ -675,6 +684,7 @@ rtc_source_set("isac_fix_c") {
deps = [
":isac_common",
"../..:webrtc_common",
"../../:typedefs",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../rtc_base:checks",
@ -799,6 +809,7 @@ rtc_source_set("pcm16b_c") {
public_configs = [ ":pcm16b_config" ]
deps = [
"../..:webrtc_common",
"../../:typedefs",
]
}
@ -856,6 +867,7 @@ rtc_source_set("webrtc_opus_c") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
@ -1042,6 +1054,8 @@ rtc_static_library("neteq") {
":neteq_decoder_enum",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
@ -1084,6 +1098,8 @@ rtc_source_set("neteq_tools_minimal") {
":neteq",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
@ -1121,7 +1137,9 @@ rtc_source_set("neteq_test_tools") {
":pcm16b",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -1169,6 +1187,7 @@ rtc_source_set("neteq_tools") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
@ -1195,6 +1214,7 @@ rtc_source_set("neteq_input_audio_tools") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -1308,6 +1328,7 @@ if (rtc_include_tests) {
":pcm16b_c",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:rtc_base_approved",
@ -1339,6 +1360,7 @@ if (rtc_include_tests) {
":neteq_test_support",
":neteq_test_tools",
"../..:webrtc_common",
"../../:typedefs",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
@ -1411,6 +1433,7 @@ if (rtc_include_tests) {
":audio_coding_module_typedefs",
":audio_format_conversion",
"..:module_api",
"../../:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../rtc_base:rtc_base_approved",
@ -1441,6 +1464,7 @@ if (rtc_include_tests) {
":audio_coding",
":audio_format_conversion",
"..:module_api",
"../../:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../rtc_base:rtc_base_approved",
@ -1519,6 +1543,7 @@ if (rtc_include_tests) {
defines = []
deps = [
"..:module_api",
"../../:typedefs",
"../../rtc_base:checks",
]
sources = [
@ -1567,7 +1592,9 @@ if (rtc_include_tests) {
rtc_test("audio_codec_speed_tests") {
testonly = true
defines = []
deps = []
deps = [
"../../:typedefs",
]
sources = [
"codecs/isac/fix/test/isac_speed_test.cc",
"codecs/opus/opus_speed_test.cc",
@ -1626,6 +1653,7 @@ if (rtc_include_tests) {
":pcm16b",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
@ -1653,6 +1681,7 @@ if (rtc_include_tests) {
":neteq_test_tools",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -1665,6 +1694,7 @@ if (rtc_include_tests) {
testonly = true
deps = audio_coding_deps + [
"../../:typedefs",
":audio_coding",
":neteq_input_audio_tools",
"../../api/audio_codecs/g711:audio_encoder_g711",
@ -1688,6 +1718,7 @@ if (rtc_include_tests) {
testonly = true
deps = audio_coding_deps + [
"../../:typedefs",
"../../system_wrappers:system_wrappers_default",
"../rtp_rtcp:rtp_rtcp_format",
"../../api:array_view",
@ -1768,6 +1799,7 @@ if (rtc_include_tests) {
":neteq",
":neteq_test_support",
"../..:webrtc_common",
"../../:typedefs",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@ -1915,6 +1947,7 @@ if (rtc_include_tests) {
deps = [
":g722",
"../..:webrtc_common",
"../../:typedefs",
]
}
@ -2098,6 +2131,7 @@ if (rtc_include_tests) {
":webrtc_opus",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",

1
modules/audio_coding/neteq/include/neteq.h
View file

@ -18,6 +18,7 @@
#include "api/audio_codecs/audio_decoder.h"
#include "api/optional.h"
#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
#include "rtc_base/constructormagic.h"

1
modules/audio_coding/neteq/tools/packet.h
View file

@ -14,6 +14,7 @@
#include <list>
#include <memory>
#include "api/rtp_headers.h" // NOLINT(build/include)
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)

1
modules/audio_coding/neteq/tools/rtp_generator.h
View file

@ -11,6 +11,7 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)

1
modules/audio_device/BUILD.gn
View file

@ -100,6 +100,7 @@ rtc_source_set("audio_device_generic") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../common_audio",
"../../rtc_base:checks",

1
modules/audio_mixer/BUILD.gn
View file

@ -36,6 +36,7 @@ rtc_static_library("audio_mixer_impl") {
":audio_frame_manipulator",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api:audio_mixer_api",
"../../audio/utility:audio_frame_operations",

10
modules/audio_processing/BUILD.gn
View file

@ -226,10 +226,12 @@ rtc_static_library("audio_processing") {
":audio_processing_statistics",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api:optional",
"../../audio/utility:audio_frame_operations",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:gtest_prod",
"../../rtc_base:protobuf_utils",
"../../rtc_base:sanitizer",
@ -341,6 +343,7 @@ rtc_source_set("audio_processing_c") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -371,6 +374,7 @@ rtc_source_set("apm_logging") {
"../../api:array_view",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:stringutils",
]
defines = []
if (apm_debug_dump) {
@ -404,6 +408,7 @@ rtc_source_set("aec_core") {
":apm_logging",
":audio_processing_statistics",
"../..:webrtc_common",
"../../:typedefs",
"../../common_audio:common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -582,6 +587,7 @@ if (rtc_include_tests) {
":audioproc_test_utils",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api:optional",
"../../common_audio:common_audio",
@ -768,6 +774,7 @@ if (rtc_include_tests) {
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:stringutils",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@ -819,6 +826,7 @@ if (rtc_include_tests) {
":audio_processing",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
@ -838,6 +846,7 @@ if (rtc_include_tests) {
deps = [
":audio_processing",
"../..:webrtc_common",
"../../:typedefs",
"../../system_wrappers",
"../../system_wrappers:metrics_default",
]
@ -892,6 +901,7 @@ if (rtc_include_tests) {
deps = [
":audioproc_debug_proto",
"../..:webrtc_common",
"../../:typedefs",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
]

2
modules/audio_processing/test/conversational_speech/BUILD.gn
View file

@ -45,6 +45,7 @@ rtc_static_library("lib") {
]
deps = [
"../../../..:webrtc_common",
"../../../../:typedefs",
"../../../../api:array_view",
"../../../../common_audio",
"../../../../rtc_base:checks",
@ -65,6 +66,7 @@ rtc_source_set("unittest") {
deps = [
":lib",
"../../../..:webrtc_common",
"../../../../:typedefs",
"../../../../api:array_view",
"../../../../api:optional",
"../../../../common_audio",

1
modules/congestion_controller/BUILD.gn
View file

@ -49,6 +49,7 @@ rtc_static_library("congestion_controller") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",

3
modules/desktop_capture/BUILD.gn
View file

@ -28,6 +28,7 @@ rtc_static_library("primitives") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../rtc_base:checks",
"../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
]
@ -89,6 +90,7 @@ if (rtc_include_tests) {
":desktop_capture_mock",
":primitives",
"../..:webrtc_common",
"../../:typedefs",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:cpu_features_api",
@ -333,6 +335,7 @@ rtc_static_library("desktop_capture_generic") {
deps = [
":primitives",
"../..:webrtc_common",
"../../:typedefs",
"../../api:refcountedbase",
"../../rtc_base:checks",
"../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.

1
modules/include/module_common_types.h
View file

@ -18,6 +18,7 @@
#include <limits>
#include "api/optional.h"
#include "api/rtp_headers.h"
#include "api/video/video_rotation.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types_public.h"

1
modules/media_file/BUILD.gn
View file

@ -35,6 +35,7 @@ rtc_static_library("media_file") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../common_audio",
"../../rtc_base:rtc_base_approved",
]

1
modules/pacing/BUILD.gn
View file

@ -34,6 +34,7 @@ rtc_static_library("pacing") {
deps = [
"..:module_api",
"../../:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../logging:rtc_event_log_api",

2
modules/remote_bitrate_estimator/BUILD.gn
View file

@ -46,6 +46,7 @@ rtc_static_library("remote_bitrate_estimator") {
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../modules:module_api",
"../../modules/rtp_rtcp:rtp_rtcp_format",
@ -140,6 +141,7 @@ if (rtc_include_tests) {
":remote_bitrate_estimator",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../rtc_base:checks",
"../../rtc_base:gtest_prod",

6
modules/rtp_rtcp/BUILD.gn
View file

@ -82,12 +82,14 @@ rtc_source_set("rtp_rtcp_format") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../common_video",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
@ -192,6 +194,7 @@ rtc_static_library("rtp_rtcp") {
":rtp_rtcp_format",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../api:optional",
@ -200,11 +203,13 @@ rtc_static_library("rtp_rtcp") {
"../../common_video",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:gtest_prod",
"../../rtc_base:rate_limiter",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../rtc_base:sequenced_task_checker",
"../../rtc_base:stringutils",
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
@ -398,6 +403,7 @@ if (rtc_include_tests) {
":rtp_rtcp_format",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../api:optional",

1
modules/rtp_rtcp/include/receive_statistics.h
View file

@ -17,6 +17,7 @@
#include "modules/include/module.h"
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {

21
modules/rtp_rtcp/include/rtp_rtcp_defines.cc
View file

@ -12,6 +12,27 @@
namespace webrtc {
StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {}
constexpr size_t StreamId::kMaxSize;
bool StreamId::IsLegalName(rtc::ArrayView<const char> name) {
return (name.size() <= kMaxSize && name.size() > 0 &&
std::all_of(name.data(), name.data() + name.size(), isalnum));
}
void StreamId::Set(const char* data, size_t size) {
// If |data| contains \0, the stream id size might become less than |size|.
RTC_CHECK_LE(size, kMaxSize);
memcpy(value_, data, size);
if (size < kMaxSize)
value_[size] = 0;
}
// StreamId is used as member of RTPHeader that is sometimes copied with memcpy
// and thus assume trivial destructibility.
static_assert(std::is_trivially_destructible<StreamId>::value, "");
PayloadUnion::PayloadUnion(const AudioPayload& payload)
: audio_payload_(payload) {}
PayloadUnion::PayloadUnion(const VideoPayload& payload)

96
modules/rtp_rtcp/include/rtp_rtcp_defines.h
View file

@ -16,6 +16,7 @@
#include <vector>
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types.h"
#include "rtc_base/deprecation.h"
@ -496,5 +497,100 @@ class TransportSequenceNumberAllocator {
virtual uint16_t AllocateSequenceNumber() = 0;
};
struct RtpPacketCounter {
RtpPacketCounter()
: header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
void Add(const RtpPacketCounter& other) {
header_bytes += other.header_bytes;
payload_bytes += other.payload_bytes;
padding_bytes += other.padding_bytes;
packets += other.packets;
}
void Subtract(const RtpPacketCounter& other) {
RTC_DCHECK_GE(header_bytes, other.header_bytes);
header_bytes -= other.header_bytes;
RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
payload_bytes -= other.payload_bytes;
RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
padding_bytes -= other.padding_bytes;
RTC_DCHECK_GE(packets, other.packets);
packets -= other.packets;
}
void AddPacket(size_t packet_length, const RTPHeader& header) {
++packets;
header_bytes += header.headerLength;
padding_bytes += header.paddingLength;
payload_bytes +=
packet_length - (header.headerLength + header.paddingLength);
}
size_t TotalBytes() const {
return header_bytes + payload_bytes + padding_bytes;
}
size_t header_bytes; // Number of bytes used by RTP headers.
size_t payload_bytes; // Payload bytes, excluding RTP headers and padding.
size_t padding_bytes; // Number of padding bytes.
uint32_t packets; // Number of packets.
};
// Data usage statistics for a (rtp) stream.
struct StreamDataCounters {
StreamDataCounters();
void Add(const StreamDataCounters& other) {
transmitted.Add(other.transmitted);
retransmitted.Add(other.retransmitted);
fec.Add(other.fec);
if (other.first_packet_time_ms != -1 &&
(other.first_packet_time_ms < first_packet_time_ms ||
first_packet_time_ms == -1)) {
// Use oldest time.
first_packet_time_ms = other.first_packet_time_ms;
}
}
void Subtract(const StreamDataCounters& other) {
transmitted.Subtract(other.transmitted);
retransmitted.Subtract(other.retransmitted);
fec.Subtract(other.fec);
if (other.first_packet_time_ms != -1 &&
(other.first_packet_time_ms > first_packet_time_ms ||
first_packet_time_ms == -1)) {
// Use youngest time.
first_packet_time_ms = other.first_packet_time_ms;
}
}
int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
}
// Returns the number of bytes corresponding to the actual media payload (i.e.
// RTP headers, padding, retransmissions and fec packets are excluded).
// Note this function does not have meaning for an RTX stream.
size_t MediaPayloadBytes() const {
return transmitted.payload_bytes - retransmitted.payload_bytes -
fec.payload_bytes;
}
int64_t first_packet_time_ms; // Time when first packet is sent/received.
RtpPacketCounter transmitted; // Number of transmitted packets/bytes.
RtpPacketCounter retransmitted; // Number of retransmitted packets/bytes.
RtpPacketCounter fec; // Number of redundancy packets/bytes.
};
// Callback, called whenever byte/packet counts have been updated.
class StreamDataCountersCallback {
public:
virtual ~StreamDataCountersCallback() {}
virtual void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) = 0;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_

1
modules/utility/BUILD.gn
View file

@ -32,6 +32,7 @@ rtc_static_library("utility") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../audio/utility:audio_frame_operations",
"../../common_audio",
"../../rtc_base:checks",

2
modules/video_capture/BUILD.gn
View file

@ -28,10 +28,12 @@ rtc_static_library("video_capture_module") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:video_frame_api_i420",
"../../common_video",
"../../media:rtc_media_base",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:stringutils",
"../../system_wrappers",
"//third_party/libyuv",
]

7
modules/video_coding/BUILD.gn
View file

@ -99,6 +99,7 @@ rtc_static_library("video_coding") {
"..:module_api",
"..:module_api_public",
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../api:video_frame_api_i420",
"../../call:video_stream_api",
@ -127,6 +128,7 @@ rtc_source_set("mock_headers") {
deps = [
":video_coding_utility",
"../..:webrtc_common",
"../../:typedefs",
"../../test:test_support",
]
}
@ -169,6 +171,7 @@ rtc_source_set("video_coding_utility") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../api/video_codecs:video_codecs_api",
"../../common_video",
@ -239,6 +242,7 @@ rtc_static_library("webrtc_i420") {
deps = [
":video_coding_utility",
"../..:webrtc_common",
"../../:typedefs",
"../../api:video_frame_api_i420",
"../../common_video:common_video",
"../../rtc_base:checks",
@ -301,6 +305,7 @@ rtc_static_library("webrtc_vp8") {
":video_coding_utility",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../api/video_codecs:video_codecs_api",
"../../common_video",
@ -431,6 +436,7 @@ if (rtc_include_tests) {
":video_coding_utility",
":webrtc_vp8",
"../..:webrtc_common",
"../../:typedefs",
"../../api:video_frame_api_i420",
"../../api/video_codecs:video_codecs_api",
"../../common_video:common_video",
@ -599,6 +605,7 @@ if (rtc_include_tests) {
":webrtc_vp9",
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
"../../api:video_frame_api",
"../../api:video_frame_api_i420",
"../../api/video_codecs:video_codecs_api",

3
p2p/BUILD.gn
View file

@ -96,6 +96,7 @@ rtc_static_library("rtc_p2p") {
"../api:ortc_api",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:stringutils",
"../system_wrappers:field_trial_api",
]
@ -183,6 +184,7 @@ if (rtc_include_tests) {
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:stringutils",
"../test:test_support",
"//testing/gmock",
"//testing/gtest",
@ -204,6 +206,7 @@ rtc_static_library("libstunprober") {
deps = [
":rtc_p2p",
"..:webrtc_common",
"../:typedefs",
"../rtc_base:checks",
"../rtc_base:rtc_base",
]

3
pc/BUILD.gn
View file

@ -78,6 +78,7 @@ rtc_static_library("rtc_pc_base") {
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_task_queue",
"../rtc_base:stringutils",
]
if (rtc_build_libsrtp) {
@ -186,6 +187,7 @@ rtc_static_library("peerconnection") {
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../stats",
"../system_wrappers",
"../system_wrappers:field_trial_api",
@ -441,6 +443,7 @@ if (rtc_include_tests) {
":peerconnection",
":rtc_pc_base",
"../rtc_base:checks",
"../rtc_base:stringutils",
]
if (is_android) {
deps += [ ":android_black_magic" ]

41
rtc_base/BUILD.gn
View file

@ -96,13 +96,18 @@ rtc_source_set("checks") {
]
deps = [
":safe_compare",
"..:webrtc_common",
"..:typedefs",
]
}
rtc_source_set("type_traits") {
rtc_source_set("rate_limiter") {
sources = [
"type_traits.h",
"rate_limiter.cc",
"rate_limiter.h",
]
deps = [
":rtc_base_approved",
"../system_wrappers",
]
}
@ -121,14 +126,25 @@ rtc_source_set("safe_compare") {
]
}
rtc_source_set("rate_limiter") {
rtc_source_set("stringutils") {
sources = [
"rate_limiter.cc",
"rate_limiter.h",
"stringutils.cc",
"stringutils.h",
]
deps = [
":rtc_base_approved",
"../system_wrappers",
":checks",
]
}
rtc_source_set("type_traits") {
sources = [
"type_traits.h",
]
}
rtc_source_set("deprecation") {
sources = [
"deprecation.h",
]
}
@ -146,7 +162,9 @@ rtc_source_set("rtc_base_approved_generic") {
deps = [
":checks",
":safe_compare",
":stringutils",
":type_traits",
"../:typedefs",
]
sources = [
@ -171,7 +189,6 @@ rtc_source_set("rtc_base_approved_generic") {
"copyonwritebuffer.h",
"criticalsection.cc",
"criticalsection.h",
"deprecation.h",
"event_tracer.cc",
"event_tracer.h",
"file.cc",
@ -216,8 +233,6 @@ rtc_source_set("rtc_base_approved_generic") {
"stringencode.cc",
"stringencode.h",
"stringize_macros.h",
"stringutils.cc",
"stringutils.h",
"swap_queue.h",
"template_util.h",
"thread_annotations.h",
@ -526,6 +541,7 @@ rtc_static_library("rtc_base_generic") {
defines = []
deps = [
":checks",
":stringutils",
"..:webrtc_common",
"../api:optional",
]
@ -845,6 +861,7 @@ rtc_source_set("rtc_base_tests_utils") {
deps = [
":checks",
":rtc_base",
":stringutils",
"../test:field_trial",
"../test:test_support",
]
@ -956,6 +973,7 @@ if (rtc_include_tests) {
":rtc_base_tests_utils",
":rtc_task_queue",
":safe_compare",
":stringutils",
"../api:array_view",
"../system_wrappers:system_wrappers",
"../test:test_support",
@ -1080,6 +1098,7 @@ if (rtc_include_tests) {
":checks",
":rtc_base_tests_main",
":rtc_base_tests_utils",
":stringutils",
"../api:optional",
"../test:test_support",
]

2
rtc_tools/BUILD.gn
View file

@ -157,6 +157,7 @@ if (!build_with_chromium) {
deps = [
"..:webrtc_common",
"../:typedefs",
"../common_video",
]
}
@ -379,6 +380,7 @@ if (rtc_include_tests) {
deps = [
"..:webrtc_common",
"../:typedefs",
"../common_audio",
"../modules/audio_processing",
"../modules/audio_processing:audioproc_debug_proto",

1
sdk/android/BUILD.gn
View file

@ -401,6 +401,7 @@ rtc_static_library("peerconnection_jni") {
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:stringutils",
"../../system_wrappers:field_trial_api",
]
}

3
system_wrappers/BUILD.gn
View file

@ -54,6 +54,7 @@ rtc_static_library("system_wrappers") {
":field_trial_api",
":metrics_api",
"..:webrtc_common",
"../:typedefs",
"../api:optional",
"../modules:module_api_public",
"../rtc_base:checks",
@ -112,6 +113,7 @@ rtc_source_set("cpu_features_api") {
]
deps = [
"..:webrtc_common",
"../:typedefs",
]
}
@ -215,6 +217,7 @@ if (rtc_include_tests) {
":metrics_default",
":system_wrappers",
"..:webrtc_common",
"../:typedefs",
"../rtc_base:rtc_base_approved",
"../test:test_main",
"//testing/gtest",

10
system_wrappers/include/ntp_time.h
View file

@ -12,6 +12,8 @@
#include <stdint.h>
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
class NtpTime {
@ -40,8 +42,12 @@ class NtpTime {
// NTP standard (RFC1305, section 3.1) explicitly state value 0 is invalid.
bool Valid() const { return value_ != 0; }
uint32_t seconds() const { return value_ / kFractionsPerSecond; }
uint32_t fractions() const { return value_ % kFractionsPerSecond; }
uint32_t seconds() const {
return rtc::dchecked_cast<uint32_t>(value_ / kFractionsPerSecond);
}
uint32_t fractions() const {
return rtc::dchecked_cast<uint32_t>(value_ % kFractionsPerSecond);
}
private:
uint64_t value_;

9
test/BUILD.gn
View file

@ -60,6 +60,7 @@ rtc_source_set("video_test_common") {
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:optional",
"../api:video_frame_api_i420",
"../api/video_codecs:video_codecs_api",
@ -151,6 +152,7 @@ rtc_source_set("test_support") {
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:array_view",
"../common_video",
"../rtc_base:gtest_prod",
@ -215,6 +217,7 @@ if (rtc_include_tests) {
":test_support",
":video_test_common",
"..:webrtc_common",
"../:typedefs",
"../api:video_frame_api",
"../api:video_frame_api_i420",
"../common_video",
@ -362,6 +365,7 @@ if (is_ios) {
"testsupport/iosfileutils.mm",
]
deps = [
"..:typedefs",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
@ -384,6 +388,7 @@ rtc_source_set("fileutils") {
]
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:optional",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
@ -470,6 +475,7 @@ rtc_source_set("direct_transport") {
}
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:transport_api",
"../call",
"../call:call_interfaces",
@ -507,6 +513,7 @@ rtc_source_set("fake_audio_device") {
}
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:array_view",
"../common_audio:common_audio",
"../modules/audio_device:audio_device",
@ -565,6 +572,7 @@ rtc_source_set("test_common") {
":test_support",
":video_test_common",
"..:webrtc_common",
"../:typedefs",
"../api:transport_api",
"../api:video_frame_api",
"../api:video_frame_api_i420",
@ -668,6 +676,7 @@ rtc_source_set("test_renderer_generic") {
deps = [
":test_support",
"..:webrtc_common",
"../:typedefs",
"../common_video",
"../media:rtc_media_base",
"../modules/media_file",

1
test/fuzzers/BUILD.gn
View file

@ -245,6 +245,7 @@ rtc_static_library("audio_decoder_fuzzer") {
]
deps = [
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../modules/rtp_rtcp:rtp_rtcp_format",

2
video/BUILD.gn
View file

@ -55,6 +55,7 @@ rtc_static_library("video") {
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:optional",
"../api:transport_api",
"../api:video_frame_api_i420",
@ -216,6 +217,7 @@ if (rtc_include_tests) {
]
deps = [
"..:webrtc_common",
"../:typedefs",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../common_video",

2
voice_engine/BUILD.gn
View file

@ -52,6 +52,7 @@ rtc_static_library("voice_engine") {
deps = [
":audio_level",
"..:webrtc_common",
"../:typedefs",
"../api:array_view",
"../api:audio_mixer_api",
"../api:call_api",
@ -94,6 +95,7 @@ rtc_static_library("audio_level") {
deps = [
"..:webrtc_common",
"../:typedefs",
"../common_audio",
"../modules:module_api",
"../rtc_base:rtc_base_approved",