| /* | |
| * 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 <algorithm> | |
| #include <string> | |
| #include <tuple> | |
| #include <utility> | |
| #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; | |
| } | |
| 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<RtpExtension>& 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<RtpExtension>& 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<RtpExtension>& 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> RtpExtension::DeduplicateHeaderExtensions( | |
| const std::vector<RtpExtension>& extensions, | |
| Filter filter) { | |
| std::vector<RtpExtension> 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 |