/* * 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_parameters.h" #include #include #include #include #include "api/array_view.h" #include "rtc_base/strings/string_builder.h" namespace webrtc { const char* DegradationPreferenceToString( DegradationPreference degradation_preference) { switch (degradation_preference) { case DegradationPreference::DISABLED: return "disabled"; case DegradationPreference::MAINTAIN_FRAMERATE: return "maintain-framerate"; case DegradationPreference::MAINTAIN_RESOLUTION: return "maintain-resolution"; case DegradationPreference::BALANCED: return "balanced"; } RTC_CHECK_NOTREACHED(); } const double kDefaultBitratePriority = 1.0; RtcpFeedback::RtcpFeedback() = default; RtcpFeedback::RtcpFeedback(RtcpFeedbackType type) : type(type) {} RtcpFeedback::RtcpFeedback(RtcpFeedbackType type, RtcpFeedbackMessageType message_type) : type(type), message_type(message_type) {} RtcpFeedback::RtcpFeedback(const RtcpFeedback& rhs) = default; RtcpFeedback::~RtcpFeedback() = default; RtpCodecCapability::RtpCodecCapability() = default; RtpCodecCapability::~RtpCodecCapability() = default; RtpHeaderExtensionCapability::RtpHeaderExtensionCapability() = default; RtpHeaderExtensionCapability::RtpHeaderExtensionCapability( absl::string_view uri) : uri(uri) {} RtpHeaderExtensionCapability::RtpHeaderExtensionCapability( absl::string_view uri, int preferred_id) : uri(uri), preferred_id(preferred_id) {} RtpHeaderExtensionCapability::RtpHeaderExtensionCapability( absl::string_view uri, int preferred_id, RtpTransceiverDirection direction) : uri(uri), preferred_id(preferred_id), direction(direction) {} RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() = default; RtpExtension::RtpExtension() = default; RtpExtension::RtpExtension(absl::string_view uri, int id) : uri(uri), id(id) {} RtpExtension::RtpExtension(absl::string_view uri, int id, bool encrypt) : uri(uri), id(id), encrypt(encrypt) {} RtpExtension::~RtpExtension() = default; RtpFecParameters::RtpFecParameters() = default; RtpFecParameters::RtpFecParameters(FecMechanism mechanism) : mechanism(mechanism) {} RtpFecParameters::RtpFecParameters(FecMechanism mechanism, uint32_t ssrc) : ssrc(ssrc), mechanism(mechanism) {} RtpFecParameters::RtpFecParameters(const RtpFecParameters& rhs) = default; RtpFecParameters::~RtpFecParameters() = default; RtpRtxParameters::RtpRtxParameters() = default; RtpRtxParameters::RtpRtxParameters(uint32_t ssrc) : ssrc(ssrc) {} RtpRtxParameters::RtpRtxParameters(const RtpRtxParameters& rhs) = default; RtpRtxParameters::~RtpRtxParameters() = default; RtpEncodingParameters::RtpEncodingParameters() = default; RtpEncodingParameters::RtpEncodingParameters(const RtpEncodingParameters& rhs) = default; RtpEncodingParameters::~RtpEncodingParameters() = default; RtpCodecParameters::RtpCodecParameters() = default; RtpCodecParameters::RtpCodecParameters(const RtpCodecParameters& rhs) = default; RtpCodecParameters::~RtpCodecParameters() = default; RtpCapabilities::RtpCapabilities() = default; RtpCapabilities::~RtpCapabilities() = default; RtcpParameters::RtcpParameters() = default; RtcpParameters::RtcpParameters(const RtcpParameters& rhs) = default; RtcpParameters::~RtcpParameters() = default; RtpParameters::RtpParameters() = default; RtpParameters::RtpParameters(const RtpParameters& rhs) = default; RtpParameters::~RtpParameters() = default; std::string RtpExtension::ToString() const { char buf[256]; rtc::SimpleStringBuilder sb(buf); sb << "{uri: " << uri; sb << ", id: " << id; if (encrypt) { sb << ", encrypt"; } sb << '}'; return sb.str(); } constexpr char RtpExtension::kEncryptHeaderExtensionsUri[]; constexpr char RtpExtension::kAudioLevelUri[]; constexpr char RtpExtension::kTimestampOffsetUri[]; constexpr char RtpExtension::kAbsSendTimeUri[]; constexpr char RtpExtension::kAbsoluteCaptureTimeUri[]; constexpr char RtpExtension::kVideoRotationUri[]; constexpr char RtpExtension::kVideoContentTypeUri[]; constexpr char RtpExtension::kVideoTimingUri[]; constexpr char RtpExtension::kGenericFrameDescriptorUri00[]; constexpr char RtpExtension::kDependencyDescriptorUri[]; constexpr char RtpExtension::kVideoLayersAllocationUri[]; constexpr char RtpExtension::kTransportSequenceNumberUri[]; constexpr char RtpExtension::kTransportSequenceNumberV2Uri[]; constexpr char RtpExtension::kPlayoutDelayUri[]; constexpr char RtpExtension::kColorSpaceUri[]; constexpr char RtpExtension::kMidUri[]; constexpr char RtpExtension::kRidUri[]; constexpr char RtpExtension::kRepairedRidUri[]; constexpr char RtpExtension::kVideoFrameTrackingIdUri[]; constexpr char RtpExtension::kCsrcAudioLevelsUri[]; constexpr int RtpExtension::kMinId; constexpr int RtpExtension::kMaxId; constexpr int RtpExtension::kMaxValueSize; constexpr int RtpExtension::kOneByteHeaderExtensionMaxId; constexpr int RtpExtension::kOneByteHeaderExtensionMaxValueSize; bool RtpExtension::IsSupportedForAudio(absl::string_view uri) { return uri == webrtc::RtpExtension::kAudioLevelUri || uri == webrtc::RtpExtension::kAbsSendTimeUri || uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri || uri == webrtc::RtpExtension::kTransportSequenceNumberUri || uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri || uri == webrtc::RtpExtension::kMidUri || uri == webrtc::RtpExtension::kRidUri || uri == webrtc::RtpExtension::kRepairedRidUri || uri == webrtc::RtpExtension::kCsrcAudioLevelsUri; } bool RtpExtension::IsSupportedForVideo(absl::string_view uri) { return uri == webrtc::RtpExtension::kTimestampOffsetUri || uri == webrtc::RtpExtension::kAbsSendTimeUri || uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri || uri == webrtc::RtpExtension::kVideoRotationUri || uri == webrtc::RtpExtension::kTransportSequenceNumberUri || uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri || uri == webrtc::RtpExtension::kPlayoutDelayUri || uri == webrtc::RtpExtension::kVideoContentTypeUri || uri == webrtc::RtpExtension::kVideoTimingUri || uri == webrtc::RtpExtension::kMidUri || uri == webrtc::RtpExtension::kGenericFrameDescriptorUri00 || uri == webrtc::RtpExtension::kDependencyDescriptorUri || uri == webrtc::RtpExtension::kColorSpaceUri || uri == webrtc::RtpExtension::kRidUri || uri == webrtc::RtpExtension::kRepairedRidUri || uri == webrtc::RtpExtension::kVideoLayersAllocationUri || uri == webrtc::RtpExtension::kVideoFrameTrackingIdUri; } bool RtpExtension::IsEncryptionSupported(absl::string_view uri) { return #if defined(ENABLE_EXTERNAL_AUTH) // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri" // here and filter out later if external auth is really used in // srtpfilter. External auth is used by Chromium and replaces the // extension header value of "kAbsSendTimeUri", so it must not be // encrypted (which can't be done by Chromium). uri != webrtc::RtpExtension::kAbsSendTimeUri && #endif uri != webrtc::RtpExtension::kEncryptHeaderExtensionsUri; } // Returns whether a header extension with the given URI exists. // Note: This does not differentiate between encrypted and non-encrypted // extensions, so use with care! static bool HeaderExtensionWithUriExists( const std::vector& extensions, absl::string_view uri) { for (const auto& extension : extensions) { if (extension.uri == uri) { return true; } } return false; } const RtpExtension* RtpExtension::FindHeaderExtensionByUri( const std::vector& extensions, absl::string_view uri, Filter filter) { const webrtc::RtpExtension* fallback_extension = nullptr; for (const auto& extension : extensions) { if (extension.uri != uri) { continue; } switch (filter) { case kDiscardEncryptedExtension: // We only accept an unencrypted extension. if (!extension.encrypt) { return &extension; } break; case kPreferEncryptedExtension: // We prefer an encrypted extension but we can fall back to an // unencrypted extension. if (extension.encrypt) { return &extension; } else { fallback_extension = &extension; } break; case kRequireEncryptedExtension: // We only accept an encrypted extension. if (extension.encrypt) { return &extension; } break; } } // Returning fallback extension (if any) return fallback_extension; } const RtpExtension* RtpExtension::FindHeaderExtensionByUriAndEncryption( const std::vector& extensions, absl::string_view uri, bool encrypt) { for (const auto& extension : extensions) { if (extension.uri == uri && extension.encrypt == encrypt) { return &extension; } } return nullptr; } const std::vector RtpExtension::DeduplicateHeaderExtensions( const std::vector& extensions, Filter filter) { std::vector filtered; // If we do not discard encrypted extensions, add them first if (filter != kDiscardEncryptedExtension) { for (const auto& extension : extensions) { if (!extension.encrypt) { continue; } if (!HeaderExtensionWithUriExists(filtered, extension.uri)) { filtered.push_back(extension); } } } // If we do not require encrypted extensions, add missing, non-encrypted // extensions. if (filter != kRequireEncryptedExtension) { for (const auto& extension : extensions) { if (extension.encrypt) { continue; } if (!HeaderExtensionWithUriExists(filtered, extension.uri)) { filtered.push_back(extension); } } } // Sort the returned vector to make comparisons of header extensions reliable. // In order of priority, we sort by uri first, then encrypt and id last. std::sort(filtered.begin(), filtered.end(), [](const RtpExtension& a, const RtpExtension& b) { return std::tie(a.uri, a.encrypt, a.id) < std::tie(b.uri, b.encrypt, b.id); }); return filtered; } } // namespace webrtc