diff options
Diffstat (limited to 'third_party/libwebrtc/api/stats')
| -rw-r--r-- | third_party/libwebrtc/api/stats/OWNERS | 3 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/stats/rtc_stats.h | 390 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/stats/rtc_stats_collector_callback.h | 30 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/stats/rtc_stats_report.h | 137 | ||||
| -rw-r--r-- | third_party/libwebrtc/api/stats/rtcstats_objects.h | 519 |
5 files changed, 1079 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/stats/OWNERS b/third_party/libwebrtc/api/stats/OWNERS new file mode 100644 index 0000000000..1556231b31 --- /dev/null +++ b/third_party/libwebrtc/api/stats/OWNERS @@ -0,0 +1,3 @@ +set noparent +hbos@webrtc.org +hta@webrtc.org diff --git a/third_party/libwebrtc/api/stats/rtc_stats.h b/third_party/libwebrtc/api/stats/rtc_stats.h new file mode 100644 index 0000000000..6cc39a309f --- /dev/null +++ b/third_party/libwebrtc/api/stats/rtc_stats.h @@ -0,0 +1,390 @@ +/* + * Copyright 2016 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_STATS_RTC_STATS_H_ +#define API_STATS_RTC_STATS_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <map> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/types/optional.h" +#include "api/units/timestamp.h" +#include "rtc_base/checks.h" +#include "rtc_base/system/rtc_export.h" +#include "rtc_base/system/rtc_export_template.h" + +namespace webrtc { + +class RTCStatsMemberInterface; + +// Abstract base class for RTCStats-derived dictionaries, see +// https://w3c.github.io/webrtc-stats/. +// +// All derived classes must have the following static variable defined: +// static const char kType[]; +// It is used as a unique class identifier and a string representation of the +// class type, see https://w3c.github.io/webrtc-stats/#rtcstatstype-str*. +// Use the `WEBRTC_RTCSTATS_IMPL` macro when implementing subclasses, see macro +// for details. +// +// Derived classes list their dictionary members, RTCStatsMember<T>, as public +// fields, allowing the following: +// +// RTCFooStats foo("fooId", Timestamp::Micros(GetCurrentTime())); +// foo.bar = 42; +// foo.baz = std::vector<std::string>(); +// foo.baz->push_back("hello world"); +// uint32_t x = *foo.bar; +// +// Pointers to all the members are available with `Members`, allowing iteration: +// +// for (const RTCStatsMemberInterface* member : foo.Members()) { +// printf("%s = %s\n", member->name(), member->ValueToString().c_str()); +// } +class RTC_EXPORT RTCStats { + public: + RTCStats(const std::string& id, Timestamp timestamp) + : id_(id), timestamp_(timestamp) {} + + virtual ~RTCStats() {} + + virtual std::unique_ptr<RTCStats> copy() const = 0; + + const std::string& id() const { return id_; } + // Time relative to the UNIX epoch (Jan 1, 1970, UTC), in microseconds. + Timestamp timestamp() const { return timestamp_; } + + // Returns the static member variable `kType` of the implementing class. + virtual const char* type() const = 0; + // Returns a vector of pointers to all the `RTCStatsMemberInterface` members + // of this class. This allows for iteration of members. For a given class, + // `Members` always returns the same members in the same order. + std::vector<const RTCStatsMemberInterface*> Members() const; + // Checks if the two stats objects are of the same type and have the same + // member values. Timestamps are not compared. These operators are exposed for + // testing. + bool operator==(const RTCStats& other) const; + bool operator!=(const RTCStats& other) const; + + // Creates a JSON readable string representation of the stats + // object, listing all of its members (names and values). + std::string ToJson() const; + + // Downcasts the stats object to an `RTCStats` subclass `T`. DCHECKs that the + // object is of type `T`. + template <typename T> + const T& cast_to() const { + RTC_DCHECK_EQ(type(), T::kType); + return static_cast<const T&>(*this); + } + + protected: + // Gets a vector of all members of this `RTCStats` object, including members + // derived from parent classes. `additional_capacity` is how many more members + // shall be reserved in the vector (so that subclasses can allocate a vector + // with room for both parent and child members without it having to resize). + virtual std::vector<const RTCStatsMemberInterface*> + MembersOfThisObjectAndAncestors(size_t additional_capacity) const; + + std::string const id_; + Timestamp timestamp_; +}; + +// All `RTCStats` classes should use these macros. +// `WEBRTC_RTCSTATS_DECL` is placed in a public section of the class definition. +// `WEBRTC_RTCSTATS_IMPL` is placed outside the class definition (in a .cc). +// +// These macros declare (in _DECL) and define (in _IMPL) the static `kType` and +// overrides methods as required by subclasses of `RTCStats`: `copy`, `type` and +// `MembersOfThisObjectAndAncestors`. The |...| argument is a list of addresses +// to each member defined in the implementing class. The list must have at least +// one member. +// +// (Since class names need to be known to implement these methods this cannot be +// part of the base `RTCStats`. While these methods could be implemented using +// templates, that would only work for immediate subclasses. Subclasses of +// subclasses also have to override these methods, resulting in boilerplate +// code. Using a macro avoids this and works for any `RTCStats` class, including +// grandchildren.) +// +// Sample usage: +// +// rtcfoostats.h: +// class RTCFooStats : public RTCStats { +// public: +// WEBRTC_RTCSTATS_DECL(); +// +// RTCFooStats(const std::string& id, Timestamp timestamp); +// +// RTCStatsMember<int32_t> foo; +// RTCStatsMember<int32_t> bar; +// }; +// +// rtcfoostats.cc: +// WEBRTC_RTCSTATS_IMPL(RTCFooStats, RTCStats, "foo-stats" +// &foo, +// &bar); +// +// RTCFooStats::RTCFooStats(const std::string& id, Timestamp timestamp) +// : RTCStats(id, timestamp), +// foo("foo"), +// bar("bar") { +// } +// +#define WEBRTC_RTCSTATS_DECL() \ + protected: \ + std::vector<const webrtc::RTCStatsMemberInterface*> \ + MembersOfThisObjectAndAncestors(size_t local_var_additional_capacity) \ + const override; \ + \ + public: \ + static const char kType[]; \ + \ + std::unique_ptr<webrtc::RTCStats> copy() const override; \ + const char* type() const override + +#define WEBRTC_RTCSTATS_IMPL(this_class, parent_class, type_str, ...) \ + const char this_class::kType[] = type_str; \ + \ + std::unique_ptr<webrtc::RTCStats> this_class::copy() const { \ + return std::make_unique<this_class>(*this); \ + } \ + \ + const char* this_class::type() const { \ + return this_class::kType; \ + } \ + \ + std::vector<const webrtc::RTCStatsMemberInterface*> \ + this_class::MembersOfThisObjectAndAncestors( \ + size_t local_var_additional_capacity) const { \ + const webrtc::RTCStatsMemberInterface* local_var_members[] = { \ + __VA_ARGS__}; \ + size_t local_var_members_count = \ + sizeof(local_var_members) / sizeof(local_var_members[0]); \ + std::vector<const webrtc::RTCStatsMemberInterface*> \ + local_var_members_vec = parent_class::MembersOfThisObjectAndAncestors( \ + local_var_members_count + local_var_additional_capacity); \ + RTC_DCHECK_GE( \ + local_var_members_vec.capacity() - local_var_members_vec.size(), \ + local_var_members_count + local_var_additional_capacity); \ + local_var_members_vec.insert(local_var_members_vec.end(), \ + &local_var_members[0], \ + &local_var_members[local_var_members_count]); \ + return local_var_members_vec; \ + } + +// A version of WEBRTC_RTCSTATS_IMPL() where "..." is omitted, used to avoid a +// compile error on windows. This is used if the stats dictionary does not +// declare any members of its own (but perhaps its parent dictionary does). +#define WEBRTC_RTCSTATS_IMPL_NO_MEMBERS(this_class, parent_class, type_str) \ + const char this_class::kType[] = type_str; \ + \ + std::unique_ptr<webrtc::RTCStats> this_class::copy() const { \ + return std::make_unique<this_class>(*this); \ + } \ + \ + const char* this_class::type() const { \ + return this_class::kType; \ + } \ + \ + std::vector<const webrtc::RTCStatsMemberInterface*> \ + this_class::MembersOfThisObjectAndAncestors( \ + size_t local_var_additional_capacity) const { \ + return parent_class::MembersOfThisObjectAndAncestors(0); \ + } + +// Interface for `RTCStats` members, which have a name and a value of a type +// defined in a subclass. Only the types listed in `Type` are supported, these +// are implemented by `RTCStatsMember<T>`. The value of a member may be +// undefined, the value can only be read if `is_defined`. +class RTCStatsMemberInterface { + public: + // Member value types. + enum Type { + kBool, // bool + kInt32, // int32_t + kUint32, // uint32_t + kInt64, // int64_t + kUint64, // uint64_t + kDouble, // double + kString, // std::string + + kSequenceBool, // std::vector<bool> + kSequenceInt32, // std::vector<int32_t> + kSequenceUint32, // std::vector<uint32_t> + kSequenceInt64, // std::vector<int64_t> + kSequenceUint64, // std::vector<uint64_t> + kSequenceDouble, // std::vector<double> + kSequenceString, // std::vector<std::string> + + kMapStringUint64, // std::map<std::string, uint64_t> + kMapStringDouble, // std::map<std::string, double> + }; + + virtual ~RTCStatsMemberInterface() {} + + const char* name() const { return name_; } + virtual Type type() const = 0; + virtual bool is_sequence() const = 0; + virtual bool is_string() const = 0; + virtual bool is_defined() const = 0; + // Type and value comparator. The names are not compared. These operators are + // exposed for testing. + bool operator==(const RTCStatsMemberInterface& other) const { + return IsEqual(other); + } + bool operator!=(const RTCStatsMemberInterface& other) const { + return !(*this == other); + } + virtual std::string ValueToString() const = 0; + // This is the same as ValueToString except for kInt64 and kUint64 types, + // where the value is represented as a double instead of as an integer. + // Since JSON stores numbers as floating point numbers, very large integers + // cannot be accurately represented, so we prefer to display them as doubles + // instead. + virtual std::string ValueToJson() const = 0; + + template <typename T> + const T& cast_to() const { + RTC_DCHECK_EQ(type(), T::StaticType()); + return static_cast<const T&>(*this); + } + + protected: + explicit RTCStatsMemberInterface(const char* name) : name_(name) {} + + virtual bool IsEqual(const RTCStatsMemberInterface& other) const = 0; + + const char* const name_; +}; + +// Template implementation of `RTCStatsMemberInterface`. +// The supported types are the ones described by +// `RTCStatsMemberInterface::Type`. +template <typename T> +class RTCStatsMember : public RTCStatsMemberInterface { + public: + explicit RTCStatsMember(const char* name) + : RTCStatsMemberInterface(name), value_() {} + RTCStatsMember(const char* name, const T& value) + : RTCStatsMemberInterface(name), value_(value) {} + RTCStatsMember(const char* name, T&& value) + : RTCStatsMemberInterface(name), value_(std::move(value)) {} + explicit RTCStatsMember(const RTCStatsMember<T>& other) + : RTCStatsMemberInterface(other.name_), value_(other.value_) {} + explicit RTCStatsMember(RTCStatsMember<T>&& other) + : RTCStatsMemberInterface(other.name_), value_(std::move(other.value_)) {} + + static Type StaticType(); + Type type() const override { return StaticType(); } + bool is_sequence() const override; + bool is_string() const override; + bool is_defined() const override { return value_.has_value(); } + std::string ValueToString() const override; + std::string ValueToJson() const override; + + template <typename U> + inline T ValueOrDefault(U default_value) const { + return value_.value_or(default_value); + } + + // Assignment operators. + T& operator=(const T& value) { + value_ = value; + return value_.value(); + } + T& operator=(const T&& value) { + value_ = std::move(value); + return value_.value(); + } + + // Getter methods that look the same as absl::optional<T>. Please prefer these + // in order to unblock replacing RTCStatsMember<T> with absl::optional<T> in + // the future (https://crbug.com/webrtc/15164). + bool has_value() const { return value_.has_value(); } + const T& value() const { return value_.value(); } + T& value() { return value_.value(); } + T& operator*() { + RTC_DCHECK(value_); + return *value_; + } + const T& operator*() const { + RTC_DCHECK(value_); + return *value_; + } + T* operator->() { + RTC_DCHECK(value_); + return &(*value_); + } + const T* operator->() const { + RTC_DCHECK(value_); + return &(*value_); + } + + protected: + bool IsEqual(const RTCStatsMemberInterface& other) const override { + if (type() != other.type()) + return false; + const RTCStatsMember<T>& other_t = + static_cast<const RTCStatsMember<T>&>(other); + return value_ == other_t.value_; + } + + private: + absl::optional<T> value_; +}; + +namespace rtc_stats_internal { + +typedef std::map<std::string, uint64_t> MapStringUint64; +typedef std::map<std::string, double> MapStringDouble; + +} // namespace rtc_stats_internal + +#define WEBRTC_DECLARE_RTCSTATSMEMBER(T) \ + template <> \ + RTC_EXPORT RTCStatsMemberInterface::Type RTCStatsMember<T>::StaticType(); \ + template <> \ + RTC_EXPORT bool RTCStatsMember<T>::is_sequence() const; \ + template <> \ + RTC_EXPORT bool RTCStatsMember<T>::is_string() const; \ + template <> \ + RTC_EXPORT std::string RTCStatsMember<T>::ValueToString() const; \ + template <> \ + RTC_EXPORT std::string RTCStatsMember<T>::ValueToJson() const; \ + extern template class RTC_EXPORT_TEMPLATE_DECLARE(RTC_EXPORT) \ + RTCStatsMember<T> + +WEBRTC_DECLARE_RTCSTATSMEMBER(bool); +WEBRTC_DECLARE_RTCSTATSMEMBER(int32_t); +WEBRTC_DECLARE_RTCSTATSMEMBER(uint32_t); +WEBRTC_DECLARE_RTCSTATSMEMBER(int64_t); +WEBRTC_DECLARE_RTCSTATSMEMBER(uint64_t); +WEBRTC_DECLARE_RTCSTATSMEMBER(double); +WEBRTC_DECLARE_RTCSTATSMEMBER(std::string); +WEBRTC_DECLARE_RTCSTATSMEMBER(std::vector<bool>); +WEBRTC_DECLARE_RTCSTATSMEMBER(std::vector<int32_t>); +WEBRTC_DECLARE_RTCSTATSMEMBER(std::vector<uint32_t>); +WEBRTC_DECLARE_RTCSTATSMEMBER(std::vector<int64_t>); +WEBRTC_DECLARE_RTCSTATSMEMBER(std::vector<uint64_t>); +WEBRTC_DECLARE_RTCSTATSMEMBER(std::vector<double>); +WEBRTC_DECLARE_RTCSTATSMEMBER(std::vector<std::string>); +WEBRTC_DECLARE_RTCSTATSMEMBER(rtc_stats_internal::MapStringUint64); +WEBRTC_DECLARE_RTCSTATSMEMBER(rtc_stats_internal::MapStringDouble); + +} // namespace webrtc + +#endif // API_STATS_RTC_STATS_H_ diff --git a/third_party/libwebrtc/api/stats/rtc_stats_collector_callback.h b/third_party/libwebrtc/api/stats/rtc_stats_collector_callback.h new file mode 100644 index 0000000000..506cc63e6f --- /dev/null +++ b/third_party/libwebrtc/api/stats/rtc_stats_collector_callback.h @@ -0,0 +1,30 @@ +/* + * Copyright 2016 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_STATS_RTC_STATS_COLLECTOR_CALLBACK_H_ +#define API_STATS_RTC_STATS_COLLECTOR_CALLBACK_H_ + +#include "api/scoped_refptr.h" +#include "api/stats/rtc_stats_report.h" +#include "rtc_base/ref_count.h" + +namespace webrtc { + +class RTCStatsCollectorCallback : public rtc::RefCountInterface { + public: + ~RTCStatsCollectorCallback() override = default; + + virtual void OnStatsDelivered( + const rtc::scoped_refptr<const RTCStatsReport>& report) = 0; +}; + +} // namespace webrtc + +#endif // API_STATS_RTC_STATS_COLLECTOR_CALLBACK_H_ diff --git a/third_party/libwebrtc/api/stats/rtc_stats_report.h b/third_party/libwebrtc/api/stats/rtc_stats_report.h new file mode 100644 index 0000000000..1dce4d89b2 --- /dev/null +++ b/third_party/libwebrtc/api/stats/rtc_stats_report.h @@ -0,0 +1,137 @@ +/* + * Copyright 2016 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_STATS_RTC_STATS_REPORT_H_ +#define API_STATS_RTC_STATS_REPORT_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <map> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "api/ref_counted_base.h" +#include "api/scoped_refptr.h" +#include "api/stats/rtc_stats.h" +#include "api/units/timestamp.h" +// TODO(tommi): Remove this include after fixing iwyu issue in chromium. +// See: third_party/blink/renderer/platform/peerconnection/rtc_stats.cc +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// A collection of stats. +// This is accessible as a map from `RTCStats::id` to `RTCStats`. +class RTC_EXPORT RTCStatsReport final + : public rtc::RefCountedNonVirtual<RTCStatsReport> { + public: + typedef std::map<std::string, std::unique_ptr<const RTCStats>> StatsMap; + + class RTC_EXPORT ConstIterator { + public: + ConstIterator(ConstIterator&& other); + ~ConstIterator(); + + ConstIterator& operator++(); + ConstIterator& operator++(int); + const RTCStats& operator*() const; + const RTCStats* operator->() const; + bool operator==(const ConstIterator& other) const; + bool operator!=(const ConstIterator& other) const; + + private: + friend class RTCStatsReport; + ConstIterator(const rtc::scoped_refptr<const RTCStatsReport>& report, + StatsMap::const_iterator it); + + // Reference report to make sure it is kept alive. + rtc::scoped_refptr<const RTCStatsReport> report_; + StatsMap::const_iterator it_; + }; + + static rtc::scoped_refptr<RTCStatsReport> Create(Timestamp timestamp); + + explicit RTCStatsReport(Timestamp timestamp); + + RTCStatsReport(const RTCStatsReport& other) = delete; + rtc::scoped_refptr<RTCStatsReport> Copy() const; + + Timestamp timestamp() const { return timestamp_; } + void AddStats(std::unique_ptr<const RTCStats> stats); + // On success, returns a non-owning pointer to `stats`. If the stats ID is not + // unique, `stats` is not inserted and nullptr is returned. + template <typename T> + T* TryAddStats(std::unique_ptr<T> stats) { + T* stats_ptr = stats.get(); + if (!stats_ + .insert(std::make_pair(std::string(stats->id()), std::move(stats))) + .second) { + return nullptr; + } + return stats_ptr; + } + const RTCStats* Get(const std::string& id) const; + size_t size() const { return stats_.size(); } + + // Gets the stat object of type `T` by ID, where `T` is any class descending + // from `RTCStats`. + // Returns null if there is no stats object for the given ID or it is the + // wrong type. + template <typename T> + const T* GetAs(const std::string& id) const { + const RTCStats* stats = Get(id); + if (!stats || stats->type() != T::kType) { + return nullptr; + } + return &stats->cast_to<const T>(); + } + + // Removes the stats object from the report, returning ownership of it or null + // if there is no object with `id`. + std::unique_ptr<const RTCStats> Take(const std::string& id); + // Takes ownership of all the stats in `other`, leaving it empty. + void TakeMembersFrom(rtc::scoped_refptr<RTCStatsReport> other); + + // Stats iterators. Stats are ordered lexicographically on `RTCStats::id`. + ConstIterator begin() const; + ConstIterator end() const; + + // Gets the subset of stats that are of type `T`, where `T` is any class + // descending from `RTCStats`. + template <typename T> + std::vector<const T*> GetStatsOfType() const { + std::vector<const T*> stats_of_type; + for (const RTCStats& stats : *this) { + if (stats.type() == T::kType) + stats_of_type.push_back(&stats.cast_to<const T>()); + } + return stats_of_type; + } + + // Creates a JSON readable string representation of the report, + // listing all of its stats objects. + std::string ToJson() const; + + protected: + friend class rtc::RefCountedNonVirtual<RTCStatsReport>; + ~RTCStatsReport() = default; + + private: + Timestamp timestamp_; + StatsMap stats_; +}; + +} // namespace webrtc + +#endif // API_STATS_RTC_STATS_REPORT_H_ diff --git a/third_party/libwebrtc/api/stats/rtcstats_objects.h b/third_party/libwebrtc/api/stats/rtcstats_objects.h new file mode 100644 index 0000000000..c28b635660 --- /dev/null +++ b/third_party/libwebrtc/api/stats/rtcstats_objects.h @@ -0,0 +1,519 @@ +/* + * Copyright 2016 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_STATS_RTCSTATS_OBJECTS_H_ +#define API_STATS_RTCSTATS_OBJECTS_H_ + +#include <stdint.h> + +#include <map> +#include <memory> +#include <string> +#include <vector> + +#include "api/stats/rtc_stats.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// https://w3c.github.io/webrtc-stats/#certificatestats-dict* +class RTC_EXPORT RTCCertificateStats final : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCCertificateStats(std::string id, Timestamp timestamp); + RTCCertificateStats(const RTCCertificateStats& other); + ~RTCCertificateStats() override; + + RTCStatsMember<std::string> fingerprint; + RTCStatsMember<std::string> fingerprint_algorithm; + RTCStatsMember<std::string> base64_certificate; + RTCStatsMember<std::string> issuer_certificate_id; +}; + +// https://w3c.github.io/webrtc-stats/#codec-dict* +class RTC_EXPORT RTCCodecStats final : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCCodecStats(std::string id, Timestamp timestamp); + RTCCodecStats(const RTCCodecStats& other); + ~RTCCodecStats() override; + + RTCStatsMember<std::string> transport_id; + RTCStatsMember<uint32_t> payload_type; + RTCStatsMember<std::string> mime_type; + RTCStatsMember<uint32_t> clock_rate; + RTCStatsMember<uint32_t> channels; + RTCStatsMember<std::string> sdp_fmtp_line; +}; + +// https://w3c.github.io/webrtc-stats/#dcstats-dict* +class RTC_EXPORT RTCDataChannelStats final : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCDataChannelStats(std::string id, Timestamp timestamp); + RTCDataChannelStats(const RTCDataChannelStats& other); + ~RTCDataChannelStats() override; + + RTCStatsMember<std::string> label; + RTCStatsMember<std::string> protocol; + RTCStatsMember<int32_t> data_channel_identifier; + RTCStatsMember<std::string> state; + RTCStatsMember<uint32_t> messages_sent; + RTCStatsMember<uint64_t> bytes_sent; + RTCStatsMember<uint32_t> messages_received; + RTCStatsMember<uint64_t> bytes_received; +}; + +// https://w3c.github.io/webrtc-stats/#candidatepair-dict* +class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCIceCandidatePairStats(std::string id, Timestamp timestamp); + RTCIceCandidatePairStats(const RTCIceCandidatePairStats& other); + ~RTCIceCandidatePairStats() override; + + RTCStatsMember<std::string> transport_id; + RTCStatsMember<std::string> local_candidate_id; + RTCStatsMember<std::string> remote_candidate_id; + RTCStatsMember<std::string> state; + // Obsolete: priority + RTCStatsMember<uint64_t> priority; + RTCStatsMember<bool> nominated; + // `writable` does not exist in the spec and old comments suggest it used to + // exist but was incorrectly implemented. + // TODO(https://crbug.com/webrtc/14171): Standardize and/or modify + // implementation. + RTCStatsMember<bool> writable; + RTCStatsMember<uint64_t> packets_sent; + RTCStatsMember<uint64_t> packets_received; + RTCStatsMember<uint64_t> bytes_sent; + RTCStatsMember<uint64_t> bytes_received; + RTCStatsMember<double> total_round_trip_time; + RTCStatsMember<double> current_round_trip_time; + RTCStatsMember<double> available_outgoing_bitrate; + RTCStatsMember<double> available_incoming_bitrate; + RTCStatsMember<uint64_t> requests_received; + RTCStatsMember<uint64_t> requests_sent; + RTCStatsMember<uint64_t> responses_received; + RTCStatsMember<uint64_t> responses_sent; + RTCStatsMember<uint64_t> consent_requests_sent; + RTCStatsMember<uint64_t> packets_discarded_on_send; + RTCStatsMember<uint64_t> bytes_discarded_on_send; + RTCStatsMember<double> last_packet_received_timestamp; + RTCStatsMember<double> last_packet_sent_timestamp; +}; + +// https://w3c.github.io/webrtc-stats/#icecandidate-dict* +class RTC_EXPORT RTCIceCandidateStats : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCIceCandidateStats(const RTCIceCandidateStats& other); + ~RTCIceCandidateStats() override; + + RTCStatsMember<std::string> transport_id; + // Obsolete: is_remote + RTCStatsMember<bool> is_remote; + RTCStatsMember<std::string> network_type; + RTCStatsMember<std::string> ip; + RTCStatsMember<std::string> address; + RTCStatsMember<int32_t> port; + RTCStatsMember<std::string> protocol; + RTCStatsMember<std::string> relay_protocol; + RTCStatsMember<std::string> candidate_type; + RTCStatsMember<int32_t> priority; + RTCStatsMember<std::string> url; + RTCStatsMember<std::string> foundation; + RTCStatsMember<std::string> related_address; + RTCStatsMember<int32_t> related_port; + RTCStatsMember<std::string> username_fragment; + RTCStatsMember<std::string> tcp_type; + + // The following metrics are NOT exposed to JavaScript. We should consider + // standardizing or removing them. + RTCStatsMember<bool> vpn; + RTCStatsMember<std::string> network_adapter_type; + + protected: + RTCIceCandidateStats(std::string id, Timestamp timestamp, bool is_remote); +}; + +// In the spec both local and remote varieties are of type RTCIceCandidateStats. +// But here we define them as subclasses of `RTCIceCandidateStats` because the +// `kType` need to be different ("RTCStatsType type") in the local/remote case. +// https://w3c.github.io/webrtc-stats/#rtcstatstype-str* +// This forces us to have to override copy() and type(). +class RTC_EXPORT RTCLocalIceCandidateStats final : public RTCIceCandidateStats { + public: + static const char kType[]; + RTCLocalIceCandidateStats(std::string id, Timestamp timestamp); + std::unique_ptr<RTCStats> copy() const override; + const char* type() const override; +}; + +class RTC_EXPORT RTCRemoteIceCandidateStats final + : public RTCIceCandidateStats { + public: + static const char kType[]; + RTCRemoteIceCandidateStats(std::string id, Timestamp timestamp); + std::unique_ptr<RTCStats> copy() const override; + const char* type() const override; +}; + +// https://w3c.github.io/webrtc-stats/#pcstats-dict* +class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCPeerConnectionStats(std::string id, Timestamp timestamp); + RTCPeerConnectionStats(const RTCPeerConnectionStats& other); + ~RTCPeerConnectionStats() override; + + RTCStatsMember<uint32_t> data_channels_opened; + RTCStatsMember<uint32_t> data_channels_closed; +}; + +// https://w3c.github.io/webrtc-stats/#streamstats-dict* +class RTC_EXPORT RTCRtpStreamStats : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCRtpStreamStats(const RTCRtpStreamStats& other); + ~RTCRtpStreamStats() override; + + RTCStatsMember<uint32_t> ssrc; + RTCStatsMember<std::string> kind; + RTCStatsMember<std::string> transport_id; + RTCStatsMember<std::string> codec_id; + + protected: + RTCRtpStreamStats(std::string id, Timestamp timestamp); +}; + +// https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict* +class RTC_EXPORT RTCReceivedRtpStreamStats : public RTCRtpStreamStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCReceivedRtpStreamStats(const RTCReceivedRtpStreamStats& other); + ~RTCReceivedRtpStreamStats() override; + + RTCStatsMember<double> jitter; + RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550 + + protected: + RTCReceivedRtpStreamStats(std::string id, Timestamp timestamp); +}; + +// https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* +class RTC_EXPORT RTCSentRtpStreamStats : public RTCRtpStreamStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCSentRtpStreamStats(const RTCSentRtpStreamStats& other); + ~RTCSentRtpStreamStats() override; + + RTCStatsMember<uint64_t> packets_sent; + RTCStatsMember<uint64_t> bytes_sent; + + protected: + RTCSentRtpStreamStats(std::string id, Timestamp timestamp); +}; + +// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict* +class RTC_EXPORT RTCInboundRtpStreamStats final + : public RTCReceivedRtpStreamStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCInboundRtpStreamStats(std::string id, Timestamp timestamp); + RTCInboundRtpStreamStats(const RTCInboundRtpStreamStats& other); + ~RTCInboundRtpStreamStats() override; + + RTCStatsMember<std::string> playout_id; + RTCStatsMember<std::string> track_identifier; + RTCStatsMember<std::string> mid; + RTCStatsMember<std::string> remote_id; + RTCStatsMember<uint32_t> packets_received; + RTCStatsMember<uint64_t> packets_discarded; + RTCStatsMember<uint64_t> fec_packets_received; + RTCStatsMember<uint64_t> fec_bytes_received; + RTCStatsMember<uint64_t> fec_packets_discarded; + // Inbound FEC SSRC. Only present if a mechanism like FlexFEC is negotiated. + RTCStatsMember<uint32_t> fec_ssrc; + RTCStatsMember<uint64_t> bytes_received; + RTCStatsMember<uint64_t> header_bytes_received; + // Inbound RTX stats. Only defined when RTX is used and it is therefore + // possible to distinguish retransmissions. + RTCStatsMember<uint64_t> retransmitted_packets_received; + RTCStatsMember<uint64_t> retransmitted_bytes_received; + RTCStatsMember<uint32_t> rtx_ssrc; + + RTCStatsMember<double> last_packet_received_timestamp; + RTCStatsMember<double> jitter_buffer_delay; + RTCStatsMember<double> jitter_buffer_target_delay; + RTCStatsMember<double> jitter_buffer_minimum_delay; + RTCStatsMember<uint64_t> jitter_buffer_emitted_count; + RTCStatsMember<uint64_t> total_samples_received; + RTCStatsMember<uint64_t> concealed_samples; + RTCStatsMember<uint64_t> silent_concealed_samples; + RTCStatsMember<uint64_t> concealment_events; + RTCStatsMember<uint64_t> inserted_samples_for_deceleration; + RTCStatsMember<uint64_t> removed_samples_for_acceleration; + RTCStatsMember<double> audio_level; + RTCStatsMember<double> total_audio_energy; + RTCStatsMember<double> total_samples_duration; + // Stats below are only implemented or defined for video. + RTCStatsMember<uint32_t> frames_received; + RTCStatsMember<uint32_t> frame_width; + RTCStatsMember<uint32_t> frame_height; + RTCStatsMember<double> frames_per_second; + RTCStatsMember<uint32_t> frames_decoded; + RTCStatsMember<uint32_t> key_frames_decoded; + RTCStatsMember<uint32_t> frames_dropped; + RTCStatsMember<double> total_decode_time; + RTCStatsMember<double> total_processing_delay; + RTCStatsMember<double> total_assembly_time; + RTCStatsMember<uint32_t> frames_assembled_from_multiple_packets; + // TODO(https://crbug.com/webrtc/15600): Implement framesRendered, which is + // incremented at the same time that totalInterFrameDelay and + // totalSquaredInterFrameDelay is incremented. (Dividing inter-frame delay by + // framesDecoded is slightly wrong.) + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesrendered + // + // TODO(https://crbug.com/webrtc/15601): Inter-frame, pause and freeze metrics + // all related to when the frame is rendered, but our implementation measures + // at delivery to sink, not at actual render time. When we have an actual + // frame rendered callback, move the calculating of these metrics to there in + // order to make them more accurate. + RTCStatsMember<double> total_inter_frame_delay; + RTCStatsMember<double> total_squared_inter_frame_delay; + RTCStatsMember<uint32_t> pause_count; + RTCStatsMember<double> total_pauses_duration; + RTCStatsMember<uint32_t> freeze_count; + RTCStatsMember<double> total_freezes_duration; + // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype + RTCStatsMember<std::string> content_type; + // Only populated if audio/video sync is enabled. + // TODO(https://crbug.com/webrtc/14177): Expose even if A/V sync is off? + RTCStatsMember<double> estimated_playout_timestamp; + // Only defined for video. + // In JavaScript, this is only exposed if HW exposure is allowed. + RTCStatsMember<std::string> decoder_implementation; + // FIR and PLI counts are only defined for |kind == "video"|. + RTCStatsMember<uint32_t> fir_count; + RTCStatsMember<uint32_t> pli_count; + RTCStatsMember<uint32_t> nack_count; + RTCStatsMember<uint64_t> qp_sum; + // This is a remnant of the legacy getStats() API. When the "video-timing" + // header extension is used, + // https://webrtc.github.io/webrtc-org/experiments/rtp-hdrext/video-timing/, + // `googTimingFrameInfo` is exposed with the value of + // TimingFrameInfo::ToString(). + // TODO(https://crbug.com/webrtc/14586): Unship or standardize this metric. + RTCStatsMember<std::string> goog_timing_frame_info; + // In JavaScript, this is only exposed if HW exposure is allowed. + RTCStatsMember<bool> power_efficient_decoder; + + // The following metrics are NOT exposed to JavaScript. We should consider + // standardizing or removing them. + RTCStatsMember<uint64_t> jitter_buffer_flushes; + RTCStatsMember<uint64_t> delayed_packet_outage_samples; + RTCStatsMember<double> relative_packet_arrival_delay; + RTCStatsMember<uint32_t> interruption_count; + RTCStatsMember<double> total_interruption_duration; + RTCStatsMember<double> min_playout_delay; +}; + +// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict* +class RTC_EXPORT RTCOutboundRtpStreamStats final + : public RTCSentRtpStreamStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCOutboundRtpStreamStats(std::string id, Timestamp timestamp); + RTCOutboundRtpStreamStats(const RTCOutboundRtpStreamStats& other); + ~RTCOutboundRtpStreamStats() override; + + RTCStatsMember<std::string> media_source_id; + RTCStatsMember<std::string> remote_id; + RTCStatsMember<std::string> mid; + RTCStatsMember<std::string> rid; + RTCStatsMember<uint64_t> retransmitted_packets_sent; + RTCStatsMember<uint64_t> header_bytes_sent; + RTCStatsMember<uint64_t> retransmitted_bytes_sent; + RTCStatsMember<double> target_bitrate; + RTCStatsMember<uint32_t> frames_encoded; + RTCStatsMember<uint32_t> key_frames_encoded; + RTCStatsMember<double> total_encode_time; + RTCStatsMember<uint64_t> total_encoded_bytes_target; + RTCStatsMember<uint32_t> frame_width; + RTCStatsMember<uint32_t> frame_height; + RTCStatsMember<double> frames_per_second; + RTCStatsMember<uint32_t> frames_sent; + RTCStatsMember<uint32_t> huge_frames_sent; + RTCStatsMember<double> total_packet_send_delay; + RTCStatsMember<std::string> quality_limitation_reason; + RTCStatsMember<std::map<std::string, double>> quality_limitation_durations; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges + RTCStatsMember<uint32_t> quality_limitation_resolution_changes; + // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype + RTCStatsMember<std::string> content_type; + // In JavaScript, this is only exposed if HW exposure is allowed. + // Only implemented for video. + // TODO(https://crbug.com/webrtc/14178): Implement for audio as well. + RTCStatsMember<std::string> encoder_implementation; + // FIR and PLI counts are only defined for |kind == "video"|. + RTCStatsMember<uint32_t> fir_count; + RTCStatsMember<uint32_t> pli_count; + RTCStatsMember<uint32_t> nack_count; + RTCStatsMember<uint64_t> qp_sum; + RTCStatsMember<bool> active; + // In JavaScript, this is only exposed if HW exposure is allowed. + RTCStatsMember<bool> power_efficient_encoder; + RTCStatsMember<std::string> scalability_mode; + + // RTX ssrc. Only present if RTX is negotiated. + RTCStatsMember<uint32_t> rtx_ssrc; +}; + +// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict* +class RTC_EXPORT RTCRemoteInboundRtpStreamStats final + : public RTCReceivedRtpStreamStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCRemoteInboundRtpStreamStats(std::string id, Timestamp timestamp); + RTCRemoteInboundRtpStreamStats(const RTCRemoteInboundRtpStreamStats& other); + ~RTCRemoteInboundRtpStreamStats() override; + + RTCStatsMember<std::string> local_id; + RTCStatsMember<double> round_trip_time; + RTCStatsMember<double> fraction_lost; + RTCStatsMember<double> total_round_trip_time; + RTCStatsMember<int32_t> round_trip_time_measurements; +}; + +// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict* +class RTC_EXPORT RTCRemoteOutboundRtpStreamStats final + : public RTCSentRtpStreamStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCRemoteOutboundRtpStreamStats(std::string id, Timestamp timestamp); + RTCRemoteOutboundRtpStreamStats(const RTCRemoteOutboundRtpStreamStats& other); + ~RTCRemoteOutboundRtpStreamStats() override; + + RTCStatsMember<std::string> local_id; + RTCStatsMember<double> remote_timestamp; + RTCStatsMember<uint64_t> reports_sent; + RTCStatsMember<double> round_trip_time; + RTCStatsMember<uint64_t> round_trip_time_measurements; + RTCStatsMember<double> total_round_trip_time; +}; + +// https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats +class RTC_EXPORT RTCMediaSourceStats : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCMediaSourceStats(const RTCMediaSourceStats& other); + ~RTCMediaSourceStats() override; + + RTCStatsMember<std::string> track_identifier; + RTCStatsMember<std::string> kind; + + protected: + RTCMediaSourceStats(std::string id, Timestamp timestamp); +}; + +// https://w3c.github.io/webrtc-stats/#dom-rtcaudiosourcestats +class RTC_EXPORT RTCAudioSourceStats final : public RTCMediaSourceStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCAudioSourceStats(std::string id, Timestamp timestamp); + RTCAudioSourceStats(const RTCAudioSourceStats& other); + ~RTCAudioSourceStats() override; + + RTCStatsMember<double> audio_level; + RTCStatsMember<double> total_audio_energy; + RTCStatsMember<double> total_samples_duration; + RTCStatsMember<double> echo_return_loss; + RTCStatsMember<double> echo_return_loss_enhancement; +}; + +// https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats +class RTC_EXPORT RTCVideoSourceStats final : public RTCMediaSourceStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCVideoSourceStats(std::string id, Timestamp timestamp); + RTCVideoSourceStats(const RTCVideoSourceStats& other); + ~RTCVideoSourceStats() override; + + RTCStatsMember<uint32_t> width; + RTCStatsMember<uint32_t> height; + RTCStatsMember<uint32_t> frames; + RTCStatsMember<double> frames_per_second; +}; + +// https://w3c.github.io/webrtc-stats/#transportstats-dict* +class RTC_EXPORT RTCTransportStats final : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCTransportStats(std::string id, Timestamp timestamp); + RTCTransportStats(const RTCTransportStats& other); + ~RTCTransportStats() override; + + RTCStatsMember<uint64_t> bytes_sent; + RTCStatsMember<uint64_t> packets_sent; + RTCStatsMember<uint64_t> bytes_received; + RTCStatsMember<uint64_t> packets_received; + RTCStatsMember<std::string> rtcp_transport_stats_id; + RTCStatsMember<std::string> dtls_state; + RTCStatsMember<std::string> selected_candidate_pair_id; + RTCStatsMember<std::string> local_certificate_id; + RTCStatsMember<std::string> remote_certificate_id; + RTCStatsMember<std::string> tls_version; + RTCStatsMember<std::string> dtls_cipher; + RTCStatsMember<std::string> dtls_role; + RTCStatsMember<std::string> srtp_cipher; + RTCStatsMember<uint32_t> selected_candidate_pair_changes; + RTCStatsMember<std::string> ice_role; + RTCStatsMember<std::string> ice_local_username_fragment; + RTCStatsMember<std::string> ice_state; +}; + +// https://w3c.github.io/webrtc-stats/#playoutstats-dict* +class RTC_EXPORT RTCAudioPlayoutStats final : public RTCStats { + public: + WEBRTC_RTCSTATS_DECL(); + + RTCAudioPlayoutStats(const std::string& id, Timestamp timestamp); + RTCAudioPlayoutStats(const RTCAudioPlayoutStats& other); + ~RTCAudioPlayoutStats() override; + + RTCStatsMember<std::string> kind; + RTCStatsMember<double> synthesized_samples_duration; + RTCStatsMember<uint64_t> synthesized_samples_events; + RTCStatsMember<double> total_samples_duration; + RTCStatsMember<double> total_playout_delay; + RTCStatsMember<uint64_t> total_samples_count; +}; + +} // namespace webrtc + +#endif // API_STATS_RTCSTATS_OBJECTS_H_ |