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Bug: webrtc:10159
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Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
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Steve Anton
2019-01-11 09:05:09 -08:00
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/*
* Copyright 2015 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.
*/
#ifndef API_RTPPARAMETERS_H_
#define API_RTPPARAMETERS_H_
#include <stdint.h>
#include <string>
#include <unordered_map>
#include <vector>
#include "absl/types/optional.h"
#include "api/mediatypes.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// These structures are intended to mirror those defined by:
// http://draft.ortc.org/#rtcrtpdictionaries*
// Contains everything specified as of 2017 Jan 24.
//
// They are used when retrieving or modifying the parameters of an
// RtpSender/RtpReceiver, or retrieving capabilities.
//
// Note on conventions: Where ORTC may use "octet", "short" and "unsigned"
// types, we typically use "int", in keeping with our style guidelines. The
// parameter's actual valid range will be enforced when the parameters are set,
// rather than when the parameters struct is built. An exception is made for
// SSRCs, since they use the full unsigned 32-bit range, and aren't expected to
// be used for any numeric comparisons/operations.
//
// Additionally, where ORTC uses strings, we may use enums for things that have
// a fixed number of supported values. However, for things that can be extended
// (such as codecs, by providing an external encoder factory), a string
// identifier is used.
enum class FecMechanism {
RED,
RED_AND_ULPFEC,
FLEXFEC,
};
// Used in RtcpFeedback struct.
enum class RtcpFeedbackType {
CCM,
NACK,
REMB, // "goog-remb"
TRANSPORT_CC,
};
// Used in RtcpFeedback struct when type is NACK or CCM.
enum class RtcpFeedbackMessageType {
// Equivalent to {type: "nack", parameter: undefined} in ORTC.
GENERIC_NACK,
PLI, // Usable with NACK.
FIR, // Usable with CCM.
};
enum class DtxStatus {
DISABLED,
ENABLED,
};
// Based on the spec in
// https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
// These options are enforced on a best-effort basis. For instance, all of
// these options may suffer some frame drops in order to avoid queuing.
// TODO(sprang): Look into possibility of more strictly enforcing the
// maintain-framerate option.
// TODO(deadbeef): Default to "balanced", as the spec indicates?
enum class DegradationPreference {
// Don't take any actions based on over-utilization signals. Not part of the
// web API.
DISABLED,
// On over-use, request lower frame rate, possibly causing frame drops.
MAINTAIN_FRAMERATE,
// On over-use, request lower resolution, possibly causing down-scaling.
MAINTAIN_RESOLUTION,
// Try to strike a "pleasing" balance between frame rate or resolution.
BALANCED,
};
extern const double kDefaultBitratePriority;
struct RtcpFeedback {
RtcpFeedbackType type = RtcpFeedbackType::CCM;
// Equivalent to ORTC "parameter" field with slight differences:
// 1. It's an enum instead of a string.
// 2. Generic NACK feedback is represented by a GENERIC_NACK message type,
// rather than an unset "parameter" value.
absl::optional<RtcpFeedbackMessageType> message_type;
// Constructors for convenience.
RtcpFeedback();
explicit RtcpFeedback(RtcpFeedbackType type);
RtcpFeedback(RtcpFeedbackType type, RtcpFeedbackMessageType message_type);
RtcpFeedback(const RtcpFeedback&);
~RtcpFeedback();
bool operator==(const RtcpFeedback& o) const {
return type == o.type && message_type == o.message_type;
}
bool operator!=(const RtcpFeedback& o) const { return !(*this == o); }
};
// RtpCodecCapability is to RtpCodecParameters as RtpCapabilities is to
// RtpParameters. This represents the static capabilities of an endpoint's
// implementation of a codec.
struct RtpCodecCapability {
RtpCodecCapability();
~RtpCodecCapability();
// Build MIME "type/subtype" string from |name| and |kind|.
std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
// Used to identify the codec. Equivalent to MIME subtype.
std::string name;
// The media type of this codec. Equivalent to MIME top-level type.
cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
// Clock rate in Hertz. If unset, the codec is applicable to any clock rate.
absl::optional<int> clock_rate;
// Default payload type for this codec. Mainly needed for codecs that use
// that have statically assigned payload types.
absl::optional<int> preferred_payload_type;
// Maximum packetization time supported by an RtpReceiver for this codec.
// TODO(deadbeef): Not implemented.
absl::optional<int> max_ptime;
// Preferred packetization time for an RtpReceiver or RtpSender of this
// codec.
// TODO(deadbeef): Not implemented.
absl::optional<int> ptime;
// The number of audio channels supported. Unused for video codecs.
absl::optional<int> num_channels;
// Feedback mechanisms supported for this codec.
std::vector<RtcpFeedback> rtcp_feedback;
// Codec-specific parameters that must be signaled to the remote party.
//
// Corresponds to "a=fmtp" parameters in SDP.
//
// Contrary to ORTC, these parameters are named using all lowercase strings.
// This helps make the mapping to SDP simpler, if an application is using
// SDP. Boolean values are represented by the string "1".
std::unordered_map<std::string, std::string> parameters;
// Codec-specific parameters that may optionally be signaled to the remote
// party.
// TODO(deadbeef): Not implemented.
std::unordered_map<std::string, std::string> options;
// Maximum number of temporal layer extensions supported by this codec.
// For example, a value of 1 indicates that 2 total layers are supported.
// TODO(deadbeef): Not implemented.
int max_temporal_layer_extensions = 0;
// Maximum number of spatial layer extensions supported by this codec.
// For example, a value of 1 indicates that 2 total layers are supported.
// TODO(deadbeef): Not implemented.
int max_spatial_layer_extensions = 0;
// Whether the implementation can send/receive SVC layers with distinct
// SSRCs. Always false for audio codecs. True for video codecs that support
// scalable video coding with MRST.
// TODO(deadbeef): Not implemented.
bool svc_multi_stream_support = false;
bool operator==(const RtpCodecCapability& o) const {
return name == o.name && kind == o.kind && clock_rate == o.clock_rate &&
preferred_payload_type == o.preferred_payload_type &&
max_ptime == o.max_ptime && ptime == o.ptime &&
num_channels == o.num_channels && rtcp_feedback == o.rtcp_feedback &&
parameters == o.parameters && options == o.options &&
max_temporal_layer_extensions == o.max_temporal_layer_extensions &&
max_spatial_layer_extensions == o.max_spatial_layer_extensions &&
svc_multi_stream_support == o.svc_multi_stream_support;
}
bool operator!=(const RtpCodecCapability& o) const { return !(*this == o); }
};
// Used in RtpCapabilities; represents the capabilities/preferences of an
// implementation for a header extension.
//
// Just called "RtpHeaderExtension" in ORTC, but the "Capability" suffix was
// added here for consistency and to avoid confusion with
// RtpHeaderExtensionParameters.
//
// Note that ORTC includes a "kind" field, but we omit this because it's
// redundant; if you call "RtpReceiver::GetCapabilities(MEDIA_TYPE_AUDIO)",
// you know you're getting audio capabilities.
struct RtpHeaderExtensionCapability {
// URI of this extension, as defined in RFC8285.
std::string uri;
// Preferred value of ID that goes in the packet.
absl::optional<int> preferred_id;
// If true, it's preferred that the value in the header is encrypted.
// TODO(deadbeef): Not implemented.
bool preferred_encrypt = false;
// Constructors for convenience.
RtpHeaderExtensionCapability();
explicit RtpHeaderExtensionCapability(const std::string& uri);
RtpHeaderExtensionCapability(const std::string& uri, int preferred_id);
~RtpHeaderExtensionCapability();
bool operator==(const RtpHeaderExtensionCapability& o) const {
return uri == o.uri && preferred_id == o.preferred_id &&
preferred_encrypt == o.preferred_encrypt;
}
bool operator!=(const RtpHeaderExtensionCapability& o) const {
return !(*this == o);
}
};
// RTP header extension, see RFC8285.
struct RtpExtension {
RtpExtension();
RtpExtension(const std::string& uri, int id);
RtpExtension(const std::string& uri, int id, bool encrypt);
~RtpExtension();
std::string ToString() const;
bool operator==(const RtpExtension& rhs) const {
return uri == rhs.uri && id == rhs.id && encrypt == rhs.encrypt;
}
static bool IsSupportedForAudio(const std::string& uri);
static bool IsSupportedForVideo(const std::string& uri);
// Return "true" if the given RTP header extension URI may be encrypted.
static bool IsEncryptionSupported(const std::string& uri);
// Returns the named header extension if found among all extensions,
// nullptr otherwise.
static const RtpExtension* FindHeaderExtensionByUri(
const std::vector<RtpExtension>& extensions,
const std::string& uri);
// Return a list of RTP header extensions with the non-encrypted extensions
// removed if both the encrypted and non-encrypted extension is present for
// the same URI.
static std::vector<RtpExtension> FilterDuplicateNonEncrypted(
const std::vector<RtpExtension>& extensions);
// Header extension for audio levels, as defined in:
// http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
static const char kAudioLevelUri[];
static const int kAudioLevelDefaultId;
// Header extension for RTP timestamp offset, see RFC 5450 for details:
// http://tools.ietf.org/html/rfc5450
static const char kTimestampOffsetUri[];
static const int kTimestampOffsetDefaultId;
// Header extension for absolute send time, see url for details:
// http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
static const char kAbsSendTimeUri[];
static const int kAbsSendTimeDefaultId;
// Header extension for coordination of video orientation, see url for
// details:
// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf
static const char kVideoRotationUri[];
static const int kVideoRotationDefaultId;
// Header extension for video content type. E.g. default or screenshare.
static const char kVideoContentTypeUri[];
static const int kVideoContentTypeDefaultId;
// Header extension for video timing.
static const char kVideoTimingUri[];
static const int kVideoTimingDefaultId;
// Header extension for video frame marking.
static const char kFrameMarkingUri[];
static const int kFrameMarkingDefaultId;
// Experimental codec agnostic frame descriptor.
static const char kGenericFrameDescriptorUri[];
static const int kGenericFrameDescriptorDefaultId;
// Header extension for transport sequence number, see url for details:
// http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
static const char kTransportSequenceNumberUri[];
static const int kTransportSequenceNumberDefaultId;
static const char kPlayoutDelayUri[];
static const int kPlayoutDelayDefaultId;
// Header extension for identifying media section within a transport.
// https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-49#section-15
static const char kMidUri[];
static const int kMidDefaultId;
// Encryption of Header Extensions, see RFC 6904 for details:
// https://tools.ietf.org/html/rfc6904
static const char kEncryptHeaderExtensionsUri[];
// Header extension for color space information.
static const char kColorSpaceUri[];
static const int kColorSpaceDefaultId;
// Header extension for RIDs and Repaired RIDs
// https://tools.ietf.org/html/draft-ietf-avtext-rid-09
// https://tools.ietf.org/html/draft-ietf-mmusic-rid-15
static const char kRidUri[];
static const int kRidDefaultId;
static const char kRepairedRidUri[];
static const int kRepairedRidDefaultId;
// Inclusive min and max IDs for two-byte header extensions and one-byte
// header extensions, per RFC8285 Section 4.2-4.3.
static constexpr int kMinId = 1;
static constexpr int kMaxId = 255;
static constexpr int kMaxValueSize = 255;
static constexpr int kOneByteHeaderExtensionMaxId = 14;
static constexpr int kOneByteHeaderExtensionMaxValueSize = 16;
std::string uri;
int id = 0;
bool encrypt = false;
};
// TODO(deadbeef): This is missing the "encrypt" flag, which is unimplemented.
typedef RtpExtension RtpHeaderExtensionParameters;
struct RtpFecParameters {
// If unset, a value is chosen by the implementation.
// Works just like RtpEncodingParameters::ssrc.
absl::optional<uint32_t> ssrc;
FecMechanism mechanism = FecMechanism::RED;
// Constructors for convenience.
RtpFecParameters();
explicit RtpFecParameters(FecMechanism mechanism);
RtpFecParameters(FecMechanism mechanism, uint32_t ssrc);
RtpFecParameters(const RtpFecParameters&);
~RtpFecParameters();
bool operator==(const RtpFecParameters& o) const {
return ssrc == o.ssrc && mechanism == o.mechanism;
}
bool operator!=(const RtpFecParameters& o) const { return !(*this == o); }
};
struct RtpRtxParameters {
// If unset, a value is chosen by the implementation.
// Works just like RtpEncodingParameters::ssrc.
absl::optional<uint32_t> ssrc;
// Constructors for convenience.
RtpRtxParameters();
explicit RtpRtxParameters(uint32_t ssrc);
RtpRtxParameters(const RtpRtxParameters&);
~RtpRtxParameters();
bool operator==(const RtpRtxParameters& o) const { return ssrc == o.ssrc; }
bool operator!=(const RtpRtxParameters& o) const { return !(*this == o); }
};
struct RtpEncodingParameters {
RtpEncodingParameters();
RtpEncodingParameters(const RtpEncodingParameters&);
~RtpEncodingParameters();
// If unset, a value is chosen by the implementation.
//
// Note that the chosen value is NOT returned by GetParameters, because it
// may change due to an SSRC conflict, in which case the conflict is handled
// internally without any event. Another way of looking at this is that an
// unset SSRC acts as a "wildcard" SSRC.
absl::optional<uint32_t> ssrc;
// Can be used to reference a codec in the |codecs| member of the
// RtpParameters that contains this RtpEncodingParameters. If unset, the
// implementation will choose the first possible codec (if a sender), or
// prepare to receive any codec (for a receiver).
// TODO(deadbeef): Not implemented. Implementation of RtpSender will always
// choose the first codec from the list.
absl::optional<int> codec_payload_type;
// Specifies the FEC mechanism, if set.
// TODO(deadbeef): Not implemented. Current implementation will use whatever
// FEC codecs are available, including red+ulpfec.
absl::optional<RtpFecParameters> fec;
// Specifies the RTX parameters, if set.
// TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
absl::optional<RtpRtxParameters> rtx;
// Only used for audio. If set, determines whether or not discontinuous
// transmission will be used, if an available codec supports it. If not
// set, the implementation default setting will be used.
// TODO(deadbeef): Not implemented. Current implementation will use a CN
// codec as long as it's present.
absl::optional<DtxStatus> dtx;
// The relative bitrate priority of this encoding. Currently this is
// implemented for the entire rtp sender by using the value of the first
// encoding parameter.
// TODO(webrtc.bugs.org/8630): Implement this per encoding parameter.
// Currently there is logic for how bitrate is distributed per simulcast layer
// in the VideoBitrateAllocator. This must be updated to incorporate relative
// bitrate priority.
double bitrate_priority = kDefaultBitratePriority;
// The relative DiffServ Code Point priority for this encoding, allowing
// packets to be marked relatively higher or lower without affecting
// bandwidth allocations. See https://w3c.github.io/webrtc-dscp-exp/ . NB
// we follow chromium's translation of the allowed string enum values for
// this field to 1.0, 0.5, et cetera, similar to bitrate_priority above.
// TODO(http://crbug.com/webrtc/8630): Implement this per encoding parameter.
double network_priority = kDefaultBitratePriority;
// Indicates the preferred duration of media represented by a packet in
// milliseconds for this encoding. If set, this will take precedence over the
// ptime set in the RtpCodecParameters. This could happen if SDP negotiation
// creates a ptime for a specific codec, which is later changed in the
// RtpEncodingParameters by the application.
// TODO(bugs.webrtc.org/8819): Not implemented.
absl::optional<int> ptime;
// If set, this represents the Transport Independent Application Specific
// maximum bandwidth defined in RFC3890. If unset, there is no maximum
// bitrate. Currently this is implemented for the entire rtp sender by using
// the value of the first encoding parameter.
//
// Just called "maxBitrate" in ORTC spec.
//
// TODO(deadbeef): With ORTC RtpSenders, this currently sets the total
// bandwidth for the entire bandwidth estimator (audio and video). This is
// just always how "b=AS" was handled, but it's not correct and should be
// fixed.
absl::optional<int> max_bitrate_bps;
// Specifies the minimum bitrate in bps for video.
// TODO(asapersson): Not implemented for ORTC API.
absl::optional<int> min_bitrate_bps;
// Specifies the maximum framerate in fps for video.
// TODO(asapersson): Different framerates are not supported per simulcast
// layer. If set, the maximum |max_framerate| is currently used.
// Not supported for screencast.
absl::optional<int> max_framerate;
// Specifies the number of temporal layers for video (if the feature is
// supported by the codec implementation).
// TODO(asapersson): Different number of temporal layers are not supported
// per simulcast layer.
// Not supported for screencast.
absl::optional<int> num_temporal_layers;
// For video, scale the resolution down by this factor.
// TODO(deadbeef): Not implemented.
absl::optional<double> scale_resolution_down_by;
// Scale the framerate down by this factor.
// TODO(deadbeef): Not implemented.
absl::optional<double> scale_framerate_down_by;
// For an RtpSender, set to true to cause this encoding to be encoded and
// sent, and false for it not to be encoded and sent. This allows control
// across multiple encodings of a sender for turning simulcast layers on and
// off.
// TODO(webrtc.bugs.org/8807): Updating this parameter will trigger an encoder
// reset, but this isn't necessarily required.
bool active = true;
// Value to use for RID RTP header extension.
// Called "encodingId" in ORTC.
// TODO(deadbeef): Not implemented.
std::string rid;
// RIDs of encodings on which this layer depends.
// Called "dependencyEncodingIds" in ORTC spec.
// TODO(deadbeef): Not implemented.
std::vector<std::string> dependency_rids;
bool operator==(const RtpEncodingParameters& o) const {
return ssrc == o.ssrc && codec_payload_type == o.codec_payload_type &&
fec == o.fec && rtx == o.rtx && dtx == o.dtx &&
bitrate_priority == o.bitrate_priority &&
network_priority == o.network_priority && ptime == o.ptime &&
max_bitrate_bps == o.max_bitrate_bps &&
min_bitrate_bps == o.min_bitrate_bps &&
max_framerate == o.max_framerate &&
num_temporal_layers == o.num_temporal_layers &&
scale_resolution_down_by == o.scale_resolution_down_by &&
scale_framerate_down_by == o.scale_framerate_down_by &&
active == o.active && rid == o.rid &&
dependency_rids == o.dependency_rids;
}
bool operator!=(const RtpEncodingParameters& o) const {
return !(*this == o);
}
};
struct RtpCodecParameters {
RtpCodecParameters();
RtpCodecParameters(const RtpCodecParameters&);
~RtpCodecParameters();
// Build MIME "type/subtype" string from |name| and |kind|.
std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
// Used to identify the codec. Equivalent to MIME subtype.
std::string name;
// The media type of this codec. Equivalent to MIME top-level type.
cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
// Payload type used to identify this codec in RTP packets.
// This must always be present, and must be unique across all codecs using
// the same transport.
int payload_type = 0;
// If unset, the implementation default is used.
absl::optional<int> clock_rate;
// The number of audio channels used. Unset for video codecs. If unset for
// audio, the implementation default is used.
// TODO(deadbeef): The "implementation default" part isn't fully implemented.
// Only defaults to 1, even though some codecs (such as opus) should really
// default to 2.
absl::optional<int> num_channels;
// The maximum packetization time to be used by an RtpSender.
// If |ptime| is also set, this will be ignored.
// TODO(deadbeef): Not implemented.
absl::optional<int> max_ptime;
// The packetization time to be used by an RtpSender.
// If unset, will use any time up to max_ptime.
// TODO(deadbeef): Not implemented.
absl::optional<int> ptime;
// Feedback mechanisms to be used for this codec.
// TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
std::vector<RtcpFeedback> rtcp_feedback;
// Codec-specific parameters that must be signaled to the remote party.
//
// Corresponds to "a=fmtp" parameters in SDP.
//
// Contrary to ORTC, these parameters are named using all lowercase strings.
// This helps make the mapping to SDP simpler, if an application is using
// SDP. Boolean values are represented by the string "1".
std::unordered_map<std::string, std::string> parameters;
bool operator==(const RtpCodecParameters& o) const {
return name == o.name && kind == o.kind && payload_type == o.payload_type &&
clock_rate == o.clock_rate && num_channels == o.num_channels &&
max_ptime == o.max_ptime && ptime == o.ptime &&
rtcp_feedback == o.rtcp_feedback && parameters == o.parameters;
}
bool operator!=(const RtpCodecParameters& o) const { return !(*this == o); }
};
// RtpCapabilities is used to represent the static capabilities of an
// endpoint. An application can use these capabilities to construct an
// RtpParameters.
struct RtpCapabilities {
RtpCapabilities();
~RtpCapabilities();
// Supported codecs.
std::vector<RtpCodecCapability> codecs;
// Supported RTP header extensions.
std::vector<RtpHeaderExtensionCapability> header_extensions;
// Supported Forward Error Correction (FEC) mechanisms. Note that the RED,
// ulpfec and flexfec codecs used by these mechanisms will still appear in
// |codecs|.
std::vector<FecMechanism> fec;
bool operator==(const RtpCapabilities& o) const {
return codecs == o.codecs && header_extensions == o.header_extensions &&
fec == o.fec;
}
bool operator!=(const RtpCapabilities& o) const { return !(*this == o); }
};
struct RtcpParameters final {
RtcpParameters();
RtcpParameters(const RtcpParameters&);
~RtcpParameters();
// The SSRC to be used in the "SSRC of packet sender" field. If not set, one
// will be chosen by the implementation.
// TODO(deadbeef): Not implemented.
absl::optional<uint32_t> ssrc;
// The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
//
// If empty in the construction of the RtpTransport, one will be generated by
// the implementation, and returned in GetRtcpParameters. Multiple
// RtpTransports created by the same OrtcFactory will use the same generated
// CNAME.
//
// If empty when passed into SetParameters, the CNAME simply won't be
// modified.
std::string cname;
// Send reduced-size RTCP?
bool reduced_size = false;
// Send RTCP multiplexed on the RTP transport?
// Not used with PeerConnection senders/receivers
bool mux = true;
bool operator==(const RtcpParameters& o) const {
return ssrc == o.ssrc && cname == o.cname &&
reduced_size == o.reduced_size && mux == o.mux;
}
bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
};
struct RTC_EXPORT RtpParameters {
RtpParameters();
RtpParameters(const RtpParameters&);
~RtpParameters();
// Used when calling getParameters/setParameters with a PeerConnection
// RtpSender, to ensure that outdated parameters are not unintentionally
// applied successfully.
std::string transaction_id;
// Value to use for MID RTP header extension.
// Called "muxId" in ORTC.
// TODO(deadbeef): Not implemented.
std::string mid;
std::vector<RtpCodecParameters> codecs;
std::vector<RtpHeaderExtensionParameters> header_extensions;
std::vector<RtpEncodingParameters> encodings;
// Only available with a Peerconnection RtpSender.
// In ORTC, our API includes an additional "RtpTransport"
// abstraction on which RTCP parameters are set.
RtcpParameters rtcp;
// When bandwidth is constrained and the RtpSender needs to choose between
// degrading resolution or degrading framerate, degradationPreference
// indicates which is preferred. Only for video tracks.
DegradationPreference degradation_preference =
DegradationPreference::BALANCED;
bool operator==(const RtpParameters& o) const {
return mid == o.mid && codecs == o.codecs &&
header_extensions == o.header_extensions &&
encodings == o.encodings && rtcp == o.rtcp &&
degradation_preference == o.degradation_preference;
}
bool operator!=(const RtpParameters& o) const { return !(*this == o); }
};
} // namespace webrtc
#endif // API_RTPPARAMETERS_H_