231 lines
9.1 KiB
C++
231 lines
9.1 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/rtp_parameters.h"
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#include <algorithm>
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#include <string>
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#include <utility>
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#include "api/array_view.h"
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#include "rtc_base/strings/string_builder.h"
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namespace webrtc {
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const char* DegradationPreferenceToString(
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DegradationPreference degradation_preference) {
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switch (degradation_preference) {
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case DegradationPreference::DISABLED:
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return "disabled";
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case DegradationPreference::MAINTAIN_FRAMERATE:
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return "maintain-framerate";
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case DegradationPreference::MAINTAIN_RESOLUTION:
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return "maintain-resolution";
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case DegradationPreference::BALANCED:
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return "balanced";
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}
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RTC_CHECK_NOTREACHED();
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}
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const double kDefaultBitratePriority = 1.0;
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RtcpFeedback::RtcpFeedback() = default;
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RtcpFeedback::RtcpFeedback(RtcpFeedbackType type) : type(type) {}
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RtcpFeedback::RtcpFeedback(RtcpFeedbackType type,
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RtcpFeedbackMessageType message_type)
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: type(type), message_type(message_type) {}
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RtcpFeedback::RtcpFeedback(const RtcpFeedback& rhs) = default;
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RtcpFeedback::~RtcpFeedback() = default;
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RtpCodecCapability::RtpCodecCapability() = default;
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RtpCodecCapability::~RtpCodecCapability() = default;
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RtpHeaderExtensionCapability::RtpHeaderExtensionCapability() = default;
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RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
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absl::string_view uri)
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: uri(uri) {}
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RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
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absl::string_view uri,
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int preferred_id)
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: uri(uri), preferred_id(preferred_id) {}
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RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
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absl::string_view uri,
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int preferred_id,
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RtpTransceiverDirection direction)
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: uri(uri), preferred_id(preferred_id), direction(direction) {}
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RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() = default;
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RtpExtension::RtpExtension() = default;
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RtpExtension::RtpExtension(absl::string_view uri, int id) : uri(uri), id(id) {}
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RtpExtension::RtpExtension(absl::string_view uri, int id, bool encrypt)
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: uri(uri), id(id), encrypt(encrypt) {}
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RtpExtension::~RtpExtension() = default;
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RtpFecParameters::RtpFecParameters() = default;
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RtpFecParameters::RtpFecParameters(FecMechanism mechanism)
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: mechanism(mechanism) {}
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RtpFecParameters::RtpFecParameters(FecMechanism mechanism, uint32_t ssrc)
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: ssrc(ssrc), mechanism(mechanism) {}
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RtpFecParameters::RtpFecParameters(const RtpFecParameters& rhs) = default;
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RtpFecParameters::~RtpFecParameters() = default;
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RtpRtxParameters::RtpRtxParameters() = default;
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RtpRtxParameters::RtpRtxParameters(uint32_t ssrc) : ssrc(ssrc) {}
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RtpRtxParameters::RtpRtxParameters(const RtpRtxParameters& rhs) = default;
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RtpRtxParameters::~RtpRtxParameters() = default;
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RtpEncodingParameters::RtpEncodingParameters() = default;
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RtpEncodingParameters::RtpEncodingParameters(const RtpEncodingParameters& rhs) =
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default;
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RtpEncodingParameters::~RtpEncodingParameters() = default;
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RtpCodecParameters::RtpCodecParameters() = default;
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RtpCodecParameters::RtpCodecParameters(const RtpCodecParameters& rhs) = default;
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RtpCodecParameters::~RtpCodecParameters() = default;
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RtpCapabilities::RtpCapabilities() = default;
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RtpCapabilities::~RtpCapabilities() = default;
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RtcpParameters::RtcpParameters() = default;
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RtcpParameters::RtcpParameters(const RtcpParameters& rhs) = default;
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RtcpParameters::~RtcpParameters() = default;
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RtpParameters::RtpParameters() = default;
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RtpParameters::RtpParameters(const RtpParameters& rhs) = default;
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RtpParameters::~RtpParameters() = default;
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std::string RtpExtension::ToString() const {
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char buf[256];
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rtc::SimpleStringBuilder sb(buf);
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sb << "{uri: " << uri;
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sb << ", id: " << id;
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if (encrypt) {
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sb << ", encrypt";
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}
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sb << '}';
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return sb.str();
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}
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constexpr char RtpExtension::kEncryptHeaderExtensionsUri[];
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constexpr char RtpExtension::kAudioLevelUri[];
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constexpr char RtpExtension::kTimestampOffsetUri[];
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constexpr char RtpExtension::kAbsSendTimeUri[];
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constexpr char RtpExtension::kAbsoluteCaptureTimeUri[];
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constexpr char RtpExtension::kVideoRotationUri[];
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constexpr char RtpExtension::kVideoContentTypeUri[];
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constexpr char RtpExtension::kVideoTimingUri[];
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constexpr char RtpExtension::kGenericFrameDescriptorUri00[];
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constexpr char RtpExtension::kDependencyDescriptorUri[];
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constexpr char RtpExtension::kVideoLayersAllocationUri[];
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constexpr char RtpExtension::kTransportSequenceNumberUri[];
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constexpr char RtpExtension::kTransportSequenceNumberV2Uri[];
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constexpr char RtpExtension::kPlayoutDelayUri[];
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constexpr char RtpExtension::kColorSpaceUri[];
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constexpr char RtpExtension::kMidUri[];
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constexpr char RtpExtension::kRidUri[];
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constexpr char RtpExtension::kRepairedRidUri[];
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constexpr int RtpExtension::kMinId;
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constexpr int RtpExtension::kMaxId;
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constexpr int RtpExtension::kMaxValueSize;
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constexpr int RtpExtension::kOneByteHeaderExtensionMaxId;
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constexpr int RtpExtension::kOneByteHeaderExtensionMaxValueSize;
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bool RtpExtension::IsSupportedForAudio(absl::string_view uri) {
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return uri == webrtc::RtpExtension::kAudioLevelUri ||
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uri == webrtc::RtpExtension::kAbsSendTimeUri ||
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uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
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uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
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uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri ||
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uri == webrtc::RtpExtension::kMidUri ||
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uri == webrtc::RtpExtension::kRidUri ||
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uri == webrtc::RtpExtension::kRepairedRidUri;
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}
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bool RtpExtension::IsSupportedForVideo(absl::string_view uri) {
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return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
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uri == webrtc::RtpExtension::kAbsSendTimeUri ||
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uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
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uri == webrtc::RtpExtension::kVideoRotationUri ||
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uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
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uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri ||
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uri == webrtc::RtpExtension::kPlayoutDelayUri ||
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uri == webrtc::RtpExtension::kVideoContentTypeUri ||
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uri == webrtc::RtpExtension::kVideoTimingUri ||
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uri == webrtc::RtpExtension::kMidUri ||
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uri == webrtc::RtpExtension::kGenericFrameDescriptorUri00 ||
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uri == webrtc::RtpExtension::kDependencyDescriptorUri ||
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uri == webrtc::RtpExtension::kColorSpaceUri ||
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uri == webrtc::RtpExtension::kRidUri ||
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uri == webrtc::RtpExtension::kRepairedRidUri ||
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uri == webrtc::RtpExtension::kVideoLayersAllocationUri;
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}
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bool RtpExtension::IsEncryptionSupported(absl::string_view uri) {
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return uri == webrtc::RtpExtension::kAudioLevelUri ||
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uri == webrtc::RtpExtension::kTimestampOffsetUri ||
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#if !defined(ENABLE_EXTERNAL_AUTH)
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// TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri"
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// here and filter out later if external auth is really used in
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// srtpfilter. External auth is used by Chromium and replaces the
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// extension header value of "kAbsSendTimeUri", so it must not be
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// encrypted (which can't be done by Chromium).
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uri == webrtc::RtpExtension::kAbsSendTimeUri ||
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#endif
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uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
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uri == webrtc::RtpExtension::kVideoRotationUri ||
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uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
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uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri ||
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uri == webrtc::RtpExtension::kPlayoutDelayUri ||
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uri == webrtc::RtpExtension::kVideoContentTypeUri ||
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uri == webrtc::RtpExtension::kMidUri ||
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uri == webrtc::RtpExtension::kRidUri ||
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uri == webrtc::RtpExtension::kRepairedRidUri ||
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uri == webrtc::RtpExtension::kVideoLayersAllocationUri;
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}
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const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
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const std::vector<RtpExtension>& extensions,
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absl::string_view uri) {
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for (const auto& extension : extensions) {
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if (extension.uri == uri) {
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return &extension;
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}
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}
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return nullptr;
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}
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std::vector<RtpExtension> RtpExtension::FilterDuplicateNonEncrypted(
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const std::vector<RtpExtension>& extensions) {
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std::vector<RtpExtension> filtered;
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for (auto extension = extensions.begin(); extension != extensions.end();
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++extension) {
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if (extension->encrypt) {
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filtered.push_back(*extension);
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continue;
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}
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// Only add non-encrypted extension if no encrypted with the same URI
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// is also present...
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if (std::any_of(extension + 1, extensions.end(),
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[&](const RtpExtension& check) {
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return extension->uri == check.uri;
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})) {
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continue;
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}
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// ...and has not been added before.
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if (!FindHeaderExtensionByUri(filtered, extension->uri)) {
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filtered.push_back(*extension);
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}
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}
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return filtered;
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}
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} // namespace webrtc
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