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This creates the RtpCodec structure for the common fields used in RtpCodecParameters and RtpCodecCapability. Remove the unused fields from both that were defined from ORTC and never implemented as well. Bug: webrtc:15064 Change-Id: I37b4c83e2051a888fc99cc0d9f7aeb8d74f0421d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301182 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39862}
297 lines
11 KiB
C++
297 lines
11 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 <tuple>
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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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RtpCodec::RtpCodec() = default;
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RtpCodec::RtpCodec(const RtpCodec&) = default;
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RtpCodec::~RtpCodec() = 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 char RtpExtension::kVideoFrameTrackingIdUri[];
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constexpr char RtpExtension::kCsrcAudioLevelsUri[];
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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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uri == webrtc::RtpExtension::kVideoFrameTrackingIdUri;
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}
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bool RtpExtension::IsEncryptionSupported(absl::string_view uri) {
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return
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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::kEncryptHeaderExtensionsUri;
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}
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// Returns whether a header extension with the given URI exists.
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// Note: This does not differentiate between encrypted and non-encrypted
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// extensions, so use with care!
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static bool HeaderExtensionWithUriExists(
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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 true;
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}
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}
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return false;
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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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Filter filter) {
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const webrtc::RtpExtension* fallback_extension = nullptr;
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for (const auto& extension : extensions) {
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if (extension.uri != uri) {
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continue;
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}
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switch (filter) {
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case kDiscardEncryptedExtension:
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// We only accept an unencrypted extension.
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if (!extension.encrypt) {
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return &extension;
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}
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break;
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case kPreferEncryptedExtension:
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// We prefer an encrypted extension but we can fall back to an
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// unencrypted extension.
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if (extension.encrypt) {
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return &extension;
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} else {
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fallback_extension = &extension;
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}
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break;
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case kRequireEncryptedExtension:
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// We only accept an encrypted extension.
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if (extension.encrypt) {
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return &extension;
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}
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break;
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}
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}
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// Returning fallback extension (if any)
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return fallback_extension;
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}
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const RtpExtension* RtpExtension::FindHeaderExtensionByUriAndEncryption(
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const std::vector<RtpExtension>& extensions,
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absl::string_view uri,
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bool encrypt) {
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for (const auto& extension : extensions) {
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if (extension.uri == uri && extension.encrypt == encrypt) {
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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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const std::vector<RtpExtension> RtpExtension::DeduplicateHeaderExtensions(
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const std::vector<RtpExtension>& extensions,
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Filter filter) {
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std::vector<RtpExtension> filtered;
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// If we do not discard encrypted extensions, add them first
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if (filter != kDiscardEncryptedExtension) {
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for (const auto& extension : extensions) {
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if (!extension.encrypt) {
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continue;
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}
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if (!HeaderExtensionWithUriExists(filtered, extension.uri)) {
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filtered.push_back(extension);
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}
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}
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}
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// If we do not require encrypted extensions, add missing, non-encrypted
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// extensions.
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if (filter != kRequireEncryptedExtension) {
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for (const auto& extension : extensions) {
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if (extension.encrypt) {
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continue;
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}
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if (!HeaderExtensionWithUriExists(filtered, extension.uri)) {
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filtered.push_back(extension);
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}
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}
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}
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// Sort the returned vector to make comparisons of header extensions reliable.
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// In order of priority, we sort by uri first, then encrypt and id last.
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std::sort(filtered.begin(), filtered.end(),
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[](const RtpExtension& a, const RtpExtension& b) {
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return std::tie(a.uri, a.encrypt, a.id) <
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std::tie(b.uri, b.encrypt, b.id);
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});
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return filtered;
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}
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} // namespace webrtc
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