blob: f506c4031c3eacf6e923f3cea67d6f2f9586c4ee [file] [log] [blame]
/*
* 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 WEBRTC_API_RTPPARAMETERS_H_
#define WEBRTC_API_RTPPARAMETERS_H_
#include <string>
#include <unordered_map>
#include <vector>
#include "webrtc/api/mediatypes.h"
#include "webrtc/base/optional.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 {
ACK,
CCM,
NACK,
REMB, // "goog-remb"
TRANSPORT_CC,
};
// Used in RtcpFeedback struct when type is ACK, 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,
};
enum class DegradationPreference {
MAINTAIN_FRAMERATE,
MAINTAIN_RESOLUTION,
BALANCED,
};
enum class PriorityType { VERY_LOW, LOW, MEDIUM, HIGH };
struct RtcpFeedback {
RtcpFeedbackType type = RtcpFeedbackType::ACK;
// 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.
rtc::Optional<RtcpFeedbackMessageType> message_type;
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 {
// 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.
rtc::Optional<int> clock_rate;
// Default payload type for this codec. Mainly needed for codecs that use
// that have statically assigned payload types.
rtc::Optional<int> preferred_payload_type;
// Maximum packetization time supported by an RtpReceiver for this codec.
// TODO(deadbeef): Not implemented.
rtc::Optional<int> max_ptime;
// Preferred packetization time for an RtpReceiver or RtpSender of this
// codec.
// TODO(deadbeef): Not implemented.
rtc::Optional<int> ptime;
// The number of audio channels supported. Unused for video codecs.
rtc::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.
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 RFC5285.
std::string uri;
// Preferred value of ID that goes in the packet.
rtc::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;
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);
}
};
// Used in RtpParameters; represents a specific configuration of a header
// extension.
struct RtpHeaderExtensionParameters {
// URI of this extension, as defined in RFC5285.
std::string uri;
// ID value that goes in the packet.
int id = 0;
// If true, the value in the header is encrypted.
// TODO(deadbeef): Not implemented.
bool encrypt = false;
bool operator==(const RtpHeaderExtensionParameters& o) const {
return uri == o.uri && id == o.id && encrypt == o.encrypt;
}
bool operator!=(const RtpHeaderExtensionParameters& o) const {
return !(*this == o);
}
};
struct RtpFecParameters {
// If unset, a value is chosen by the implementation.
rtc::Optional<uint32_t> ssrc;
FecMechanism mechanism = FecMechanism::RED;
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.
rtc::Optional<uint32_t> ssrc;
bool operator==(const RtpRtxParameters& o) const { return ssrc == o.ssrc; }
bool operator!=(const RtpRtxParameters& o) const { return !(*this == o); }
};
struct RtpEncodingParameters {
// If unset, a value is chosen by the implementation.
rtc::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.
// TODO(deadbeef): Not implemented.
rtc::Optional<int> codec_payload_type;
// Specifies the FEC mechanism, if set.
// TODO(deadbeef): Not implemented.
rtc::Optional<RtpFecParameters> fec;
// Specifies the RTX parameters, if set.
// TODO(deadbeef): Not implemented.
rtc::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.
rtc::Optional<DtxStatus> dtx;
// The relative priority of this encoding.
// TODO(deadbeef): Not implemented.
rtc::Optional<PriorityType> priority;
// If set, this represents the Transport Independent Application Specific
// maximum bandwidth defined in RFC3890. If unset, there is no maximum
// bitrate.
// Just called "maxBitrate" in ORTC spec.
rtc::Optional<int> max_bitrate_bps;
// TODO(deadbeef): Not implemented.
rtc::Optional<int> max_framerate;
// For video, scale the resolution down by this factor.
// TODO(deadbeef): Not implemented.
double scale_resolution_down_by = 1.0;
// Scale the framerate down by this factor.
// TODO(deadbeef): Not implemented.
double scale_framerate_down_by = 1.0;
// For an RtpSender, set to true to cause this encoding to be sent, and false
// for it not to be sent. For an RtpReceiver, set to true to cause the
// encoding to be decoded, and false for it to be ignored.
// TODO(deadbeef): RtpReceiver part is not implemented.
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 &&
priority == o.priority && max_bitrate_bps == o.max_bitrate_bps &&
max_framerate == o.max_framerate &&
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 {
// 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.
rtc::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 is unimplemented.
rtc::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.
rtc::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.
rtc::Optional<int> ptime;
// Feedback mechanisms to be used for this codec.
// TODO(deadbeef): Not implemented.
std::vector<RtcpFeedback> rtcp_feedback;
// Codec-specific parameters that must be signaled to the remote party.
// Corresponds to "a=fmtp" parameters in SDP.
// TODO(deadbeef): Not implemented.
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 {
// Supported codecs.
std::vector<RtpCodecCapability> codecs;
// Supported RTP header extensions.
std::vector<RtpHeaderExtensionCapability> header_extensions;
// Supported Forward Error Correction (FEC) mechanisms.
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); }
};
// Note that unlike in ORTC, an RtcpParameters is not included in
// RtpParameters, because our API will include an additional "RtpTransport"
// abstraction on which RTCP parameters are set.
struct RtpParameters {
// Used when calling getParameters/setParameters with a PeerConnection
// RtpSender, to ensure that outdated parameters are not unintentionally
// applied successfully.
// TODO(deadbeef): Not implemented.
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;
// TODO(deadbeef): Not implemented.
std::vector<RtpHeaderExtensionParameters> header_extensions;
std::vector<RtpEncodingParameters> encodings;
// TODO(deadbeef): Not implemented.
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 &&
degradation_preference == o.degradation_preference;
}
bool operator!=(const RtpParameters& o) const { return !(*this == o); }
};
} // namespace webrtc
#endif // WEBRTC_API_RTPPARAMETERS_H_