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This should help debugging when adaptation is or is not happening unexpectedly. Log spam is prevented by not logging if the same result happened to the same resource already and we haven't adapted since then. Bug: webrtc:11616 Change-Id: Ia6c5cc35061d252f1c66f2f2bf3b94d2485498d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176221 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31378}
228 lines
8.9 KiB
C++
228 lines
8.9 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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}
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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::kFrameMarkingUri[];
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constexpr char RtpExtension::kGenericFrameDescriptorUri00[];
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constexpr char RtpExtension::kDependencyDescriptorUri[];
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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::kFrameMarkingUri ||
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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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}
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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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}
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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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