| /* |
| * 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_RTP_PARAMETERS_H_ |
| #define API_RTP_PARAMETERS_H_ |
| |
| #include <stdint.h> |
| #include <string> |
| #include <unordered_map> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/media_types.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, |
| LNTF, // "goog-lntf" |
| 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 resolution, possibly causing down-scaling. |
| MAINTAIN_FRAMERATE, |
| // On over-use, request lower frame rate, possibly causing frame drops. |
| MAINTAIN_RESOLUTION, |
| // Try to strike a "pleasing" balance between frame rate or resolution. |
| BALANCED, |
| }; |
| |
| RTC_EXPORT 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[]; |
| |
| // Header extension for RTP timestamp offset, see RFC 5450 for details: |
| // http://tools.ietf.org/html/rfc5450 |
| static const char kTimestampOffsetUri[]; |
| |
| // Header extension for absolute send time, see url for details: |
| // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time |
| static const char kAbsSendTimeUri[]; |
| |
| // Header extension for absolute capture time, see url for details: |
| // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time |
| static const char kAbsoluteCaptureTimeUri[]; |
| |
| // 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[]; |
| |
| // Header extension for video content type. E.g. default or screenshare. |
| static const char kVideoContentTypeUri[]; |
| |
| // Header extension for video timing. |
| static const char kVideoTimingUri[]; |
| |
| // Header extension for video frame marking. |
| static const char kFrameMarkingUri[]; |
| |
| // Experimental codec agnostic frame descriptor. |
| static const char kGenericFrameDescriptorUri00[]; |
| static const char kGenericFrameDescriptorUri01[]; |
| // TODO(bugs.webrtc.org/10243): Remove once dependencies have been updated. |
| static const char kGenericFrameDescriptorUri[]; |
| |
| // 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 char kTransportSequenceNumberV2Uri[]; |
| |
| static const char kPlayoutDelayUri[]; |
| |
| // 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[]; |
| |
| // 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[]; |
| |
| // 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 char kRepairedRidUri[]; |
| |
| // 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 RTC_EXPORT 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. |
| // Screencast support is experimental. |
| absl::optional<int> num_temporal_layers; |
| |
| // For video, scale the resolution down by this factor. |
| 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. |
| 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 RTC_EXPORT 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_RTP_PARAMETERS_H_ |