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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/rtpparameters.h"
#include <algorithm>
#include <sstream>
#include <string>
#include "rtc_base/checks.h"
namespace webrtc {
RtcpFeedback::RtcpFeedback() {}
RtcpFeedback::RtcpFeedback(RtcpFeedbackType type) : type(type) {}
RtcpFeedback::RtcpFeedback(RtcpFeedbackType type,
RtcpFeedbackMessageType message_type)
: type(type), message_type(message_type) {}
RtcpFeedback::~RtcpFeedback() {}
RtpCodecCapability::RtpCodecCapability() {}
RtpCodecCapability::~RtpCodecCapability() {}
RtpHeaderExtensionCapability::RtpHeaderExtensionCapability() {}
RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
const std::string& uri)
: uri(uri) {}
RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
const std::string& uri,
int preferred_id)
: uri(uri), preferred_id(preferred_id) {}
RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() {}
RtpExtension::RtpExtension() {}
RtpExtension::RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
RtpExtension::RtpExtension(const std::string& uri, int id, bool encrypt)
: uri(uri), id(id), encrypt(encrypt) {}
RtpExtension::~RtpExtension() {}
RtpFecParameters::RtpFecParameters() {}
RtpFecParameters::RtpFecParameters(FecMechanism mechanism)
: mechanism(mechanism) {}
RtpFecParameters::RtpFecParameters(FecMechanism mechanism, uint32_t ssrc)
: ssrc(ssrc), mechanism(mechanism) {}
RtpFecParameters::~RtpFecParameters() {}
RtpRtxParameters::RtpRtxParameters() {}
RtpRtxParameters::RtpRtxParameters(uint32_t ssrc) : ssrc(ssrc) {}
RtpRtxParameters::~RtpRtxParameters() {}
RtpEncodingParameters::RtpEncodingParameters() {}
RtpEncodingParameters::~RtpEncodingParameters() {}
RtpCodecParameters::RtpCodecParameters() {}
RtpCodecParameters::~RtpCodecParameters() {}
RtpCapabilities::RtpCapabilities() {}
RtpCapabilities::~RtpCapabilities() {}
RtpParameters::RtpParameters() {}
RtpParameters::~RtpParameters() {}
std::string RtpExtension::ToString() const {
std::stringstream ss;
ss << "{uri: " << uri;
ss << ", id: " << id;
if (encrypt) {
ss << ", encrypt";
}
ss << '}';
return ss.str();
}
const char RtpExtension::kAudioLevelUri[] =
"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
const int RtpExtension::kAudioLevelDefaultId = 1;
const char RtpExtension::kTimestampOffsetUri[] =
"urn:ietf:params:rtp-hdrext:toffset";
const int RtpExtension::kTimestampOffsetDefaultId = 2;
const char RtpExtension::kAbsSendTimeUri[] =
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
const int RtpExtension::kAbsSendTimeDefaultId = 3;
const char RtpExtension::kVideoRotationUri[] = "urn:3gpp:video-orientation";
const int RtpExtension::kVideoRotationDefaultId = 4;
const char RtpExtension::kTransportSequenceNumberUri[] =
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
// This extension allows applications to adaptively limit the playout delay
// on frames as per the current needs. For example, a gaming application
// has very different needs on end-to-end delay compared to a video-conference
// application.
const char RtpExtension::kPlayoutDelayUri[] =
"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
const int RtpExtension::kPlayoutDelayDefaultId = 6;
const char RtpExtension::kVideoContentTypeUri[] =
"http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
const int RtpExtension::kVideoContentTypeDefaultId = 7;
const char RtpExtension::kVideoTimingUri[] =
"http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
const int RtpExtension::kVideoTimingDefaultId = 8;
const char RtpExtension::kEncryptHeaderExtensionsUri[] =
"urn:ietf:params:rtp-hdrext:encrypt";
const char* RtpExtension::kRtpStreamIdUri =
"urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
const int RtpExtension::kRtpStreamIdDefaultId = 9;
const char* RtpExtension::kRepairedRtpStreamIdUri =
"urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
const int RtpExtension::kRepairedRtpStreamIdDefaultId = 10;
const char* RtpExtension::kMIdUri =
"urn:ietf:params:rtp-hdrext:sdes:mid";
const int RtpExtension::kMIdDefaultId = 11;
const char* RtpExtension::kCsrcAudioLevelUri =
"urn:ietf:params:rtp-hdrext:csrc-audio-level";
const int RtpExtension::kCsrcAudioLevelDefaultId = 12;
const int RtpExtension::kMinId = 1;
const int RtpExtension::kMaxId = 14;
bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
return uri == webrtc::RtpExtension::kAudioLevelUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
uri == webrtc::RtpExtension::kRtpStreamIdUri ||
uri == webrtc::RtpExtension::kRepairedRtpStreamIdUri ||
uri == webrtc::RtpExtension::kMIdUri ||
uri == webrtc::RtpExtension::kCsrcAudioLevelUri;
}
bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
uri == webrtc::RtpExtension::kAbsSendTimeUri ||
uri == webrtc::RtpExtension::kVideoRotationUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
uri == webrtc::RtpExtension::kPlayoutDelayUri ||
uri == webrtc::RtpExtension::kVideoContentTypeUri ||
uri == webrtc::RtpExtension::kVideoTimingUri ||
uri == webrtc::RtpExtension::kRtpStreamIdUri ||
uri == webrtc::RtpExtension::kRepairedRtpStreamIdUri ||
uri == webrtc::RtpExtension::kMIdUri;
}
bool RtpExtension::IsEncryptionSupported(const std::string& 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::kVideoRotationUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
uri == webrtc::RtpExtension::kPlayoutDelayUri ||
uri == webrtc::RtpExtension::kVideoContentTypeUri;
}
const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
const std::vector<RtpExtension>& extensions,
const std::string& 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::find_if(extension + 1, extensions.end(),
[extension](const RtpExtension& check) {
return extension->uri == check.uri;
}) != extensions.end()) {
continue;
}
// ...and has not been added before.
if (!FindHeaderExtensionByUri(filtered, extension->uri)) {
filtered.push_back(*extension);
}
}
return filtered;
}
} // namespace webrtc
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