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/*
* 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 <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 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 uri == webrtc::RtpExtension::kAudioLevelUri ||
uri == webrtc::RtpExtension::kTimestampOffsetUri ||
#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::kAbsoluteCaptureTimeUri ||
uri == webrtc::RtpExtension::kVideoRotationUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri ||
uri == webrtc::RtpExtension::kPlayoutDelayUri ||
uri == webrtc::RtpExtension::kVideoContentTypeUri ||
uri == webrtc::RtpExtension::kMidUri ||
uri == webrtc::RtpExtension::kRidUri ||
uri == webrtc::RtpExtension::kRepairedRidUri ||
uri == webrtc::RtpExtension::kVideoLayersAllocationUri;
}
const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
const std::vector<RtpExtension>& extensions,
absl::string_view uri) {
for (const auto& extension : extensions) {
if (extension.uri == uri) {
return &extension;
}
}
return nullptr;
}
std::vector<RtpExtension> RtpExtension::FilterDuplicateNonEncrypted(
const std::vector<RtpExtension>& extensions) {
std::vector<RtpExtension> filtered;
for (auto extension = extensions.begin(); extension != extensions.end();
++extension) {
if (extension->encrypt) {
filtered.push_back(*extension);
continue;
}
// Only add non-encrypted extension if no encrypted with the same URI
// is also present...
if (std::any_of(extension + 1, extensions.end(),
[&](const RtpExtension& check) {
return extension->uri == check.uri;
})) {
continue;
}
// ...and has not been added before.
if (!FindHeaderExtensionByUri(filtered, extension->uri)) {
filtered.push_back(*extension);
}
}
return filtered;
}
} // namespace webrtc
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