| /* |
| * Copyright (c) 2013 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 CALL_VIDEO_RECEIVE_STREAM_H_ |
| #define CALL_VIDEO_RECEIVE_STREAM_H_ |
| |
| #include <limits> |
| #include <map> |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/call/transport.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/crypto/frame_decryptor_interface.h" |
| #include "api/frame_transformer_interface.h" |
| #include "api/rtp_headers.h" |
| #include "api/rtp_parameters.h" |
| #include "api/transport/rtp/rtp_source.h" |
| #include "api/video/recordable_encoded_frame.h" |
| #include "api/video/video_content_type.h" |
| #include "api/video/video_frame.h" |
| #include "api/video/video_sink_interface.h" |
| #include "api/video/video_timing.h" |
| #include "api/video_codecs/sdp_video_format.h" |
| #include "call/rtp_config.h" |
| #include "common_video/frame_counts.h" |
| #include "modules/rtp_rtcp/include/rtcp_statistics.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| |
| namespace webrtc { |
| |
| class RtpPacketSinkInterface; |
| class VideoDecoderFactory; |
| |
| class VideoReceiveStream { |
| public: |
| // Class for handling moving in/out recording state. |
| struct RecordingState { |
| RecordingState() = default; |
| explicit RecordingState( |
| std::function<void(const RecordableEncodedFrame&)> callback) |
| : callback(std::move(callback)) {} |
| |
| // Callback stored from the VideoReceiveStream. The VideoReceiveStream |
| // client should not interpret the attribute. |
| std::function<void(const RecordableEncodedFrame&)> callback; |
| // Memento of when a keyframe request was last sent. The VideoReceiveStream |
| // client should not interpret the attribute. |
| absl::optional<int64_t> last_keyframe_request_ms; |
| }; |
| |
| // TODO(mflodman) Move all these settings to VideoDecoder and move the |
| // declaration to common_types.h. |
| struct Decoder { |
| Decoder(); |
| Decoder(const Decoder&); |
| ~Decoder(); |
| std::string ToString() const; |
| |
| SdpVideoFormat video_format; |
| |
| // Received RTP packets with this payload type will be sent to this decoder |
| // instance. |
| int payload_type = 0; |
| }; |
| |
| struct Stats { |
| Stats(); |
| ~Stats(); |
| std::string ToString(int64_t time_ms) const; |
| |
| int network_frame_rate = 0; |
| int decode_frame_rate = 0; |
| int render_frame_rate = 0; |
| uint32_t frames_rendered = 0; |
| |
| // Decoder stats. |
| std::string decoder_implementation_name = "unknown"; |
| FrameCounts frame_counts; |
| int decode_ms = 0; |
| int max_decode_ms = 0; |
| int current_delay_ms = 0; |
| int target_delay_ms = 0; |
| int jitter_buffer_ms = 0; |
| // https://w3c.github.io/webrtc-stats/#dom-rtcvideoreceiverstats-jitterbufferdelay |
| double jitter_buffer_delay_seconds = 0; |
| // https://w3c.github.io/webrtc-stats/#dom-rtcvideoreceiverstats-jitterbufferemittedcount |
| uint64_t jitter_buffer_emitted_count = 0; |
| int min_playout_delay_ms = 0; |
| int render_delay_ms = 10; |
| int64_t interframe_delay_max_ms = -1; |
| // Frames dropped due to decoding failures or if the system is too slow. |
| // https://www.w3.org/TR/webrtc-stats/#dom-rtcvideoreceiverstats-framesdropped |
| uint32_t frames_dropped = 0; |
| uint32_t frames_decoded = 0; |
| // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime |
| uint64_t total_decode_time_ms = 0; |
| // Total inter frame delay in seconds. |
| // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalinterframedelay |
| double total_inter_frame_delay = 0; |
| // Total squared inter frame delay in seconds^2. |
| // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalsqauredinterframedelay |
| double total_squared_inter_frame_delay = 0; |
| int64_t first_frame_received_to_decoded_ms = -1; |
| absl::optional<uint64_t> qp_sum; |
| |
| int current_payload_type = -1; |
| |
| int total_bitrate_bps = 0; |
| |
| int width = 0; |
| int height = 0; |
| |
| uint32_t freeze_count = 0; |
| uint32_t pause_count = 0; |
| uint32_t total_freezes_duration_ms = 0; |
| uint32_t total_pauses_duration_ms = 0; |
| uint32_t total_frames_duration_ms = 0; |
| double sum_squared_frame_durations = 0.0; |
| |
| VideoContentType content_type = VideoContentType::UNSPECIFIED; |
| |
| // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp |
| absl::optional<int64_t> estimated_playout_ntp_timestamp_ms; |
| int sync_offset_ms = std::numeric_limits<int>::max(); |
| |
| uint32_t ssrc = 0; |
| std::string c_name; |
| RtpReceiveStats rtp_stats; |
| RtcpPacketTypeCounter rtcp_packet_type_counts; |
| |
| // Timing frame info: all important timestamps for a full lifetime of a |
| // single 'timing frame'. |
| absl::optional<webrtc::TimingFrameInfo> timing_frame_info; |
| }; |
| |
| struct Config { |
| private: |
| // Access to the copy constructor is private to force use of the Copy() |
| // method for those exceptional cases where we do use it. |
| Config(const Config&); |
| |
| public: |
| Config() = delete; |
| Config(Config&&); |
| explicit Config(Transport* rtcp_send_transport); |
| Config& operator=(Config&&); |
| Config& operator=(const Config&) = delete; |
| ~Config(); |
| |
| // Mostly used by tests. Avoid creating copies if you can. |
| Config Copy() const { return Config(*this); } |
| |
| std::string ToString() const; |
| |
| // Decoders for every payload that we can receive. |
| std::vector<Decoder> decoders; |
| |
| // Ownership stays with WebrtcVideoEngine (delegated from PeerConnection). |
| VideoDecoderFactory* decoder_factory = nullptr; |
| |
| // Receive-stream specific RTP settings. |
| struct Rtp { |
| Rtp(); |
| Rtp(const Rtp&); |
| ~Rtp(); |
| std::string ToString() const; |
| |
| // Synchronization source (stream identifier) to be received. |
| uint32_t remote_ssrc = 0; |
| |
| // Sender SSRC used for sending RTCP (such as receiver reports). |
| uint32_t local_ssrc = 0; |
| |
| // See RtcpMode for description. |
| RtcpMode rtcp_mode = RtcpMode::kCompound; |
| |
| // Extended RTCP settings. |
| struct RtcpXr { |
| // True if RTCP Receiver Reference Time Report Block extension |
| // (RFC 3611) should be enabled. |
| bool receiver_reference_time_report = false; |
| } rtcp_xr; |
| |
| // See draft-holmer-rmcat-transport-wide-cc-extensions for details. |
| bool transport_cc = false; |
| |
| // See LntfConfig for description. |
| LntfConfig lntf; |
| |
| // See NackConfig for description. |
| NackConfig nack; |
| |
| // Payload types for ULPFEC and RED, respectively. |
| int ulpfec_payload_type = -1; |
| int red_payload_type = -1; |
| |
| // SSRC for retransmissions. |
| uint32_t rtx_ssrc = 0; |
| |
| // Set if the stream is protected using FlexFEC. |
| bool protected_by_flexfec = false; |
| |
| // Optional callback sink to support additional packet handlsers such as |
| // FlexFec. |
| RtpPacketSinkInterface* packet_sink_ = nullptr; |
| |
| // Map from rtx payload type -> media payload type. |
| // For RTX to be enabled, both an SSRC and this mapping are needed. |
| std::map<int, int> rtx_associated_payload_types; |
| |
| // Payload types that should be depacketized using raw depacketizer |
| // (payload header will not be parsed and must not be present, additional |
| // meta data is expected to be present in generic frame descriptor |
| // RTP header extension). |
| std::set<int> raw_payload_types; |
| |
| // RTP header extensions used for the received stream. |
| std::vector<RtpExtension> extensions; |
| } rtp; |
| |
| // Transport for outgoing packets (RTCP). |
| Transport* rtcp_send_transport = nullptr; |
| |
| // Must always be set. |
| rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr; |
| |
| // Expected delay needed by the renderer, i.e. the frame will be delivered |
| // this many milliseconds, if possible, earlier than the ideal render time. |
| int render_delay_ms = 10; |
| |
| // If false, pass frames on to the renderer as soon as they are |
| // available. |
| bool enable_prerenderer_smoothing = true; |
| |
| // Identifier for an A/V synchronization group. Empty string to disable. |
| // TODO(pbos): Synchronize streams in a sync group, not just video streams |
| // to one of the audio streams. |
| std::string sync_group; |
| |
| // Target delay in milliseconds. A positive value indicates this stream is |
| // used for streaming instead of a real-time call. |
| int target_delay_ms = 0; |
| |
| // An optional custom frame decryptor that allows the entire frame to be |
| // decrypted in whatever way the caller choses. This is not required by |
| // default. |
| rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor; |
| |
| // Per PeerConnection cryptography options. |
| CryptoOptions crypto_options; |
| |
| rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer; |
| }; |
| |
| // Starts stream activity. |
| // When a stream is active, it can receive, process and deliver packets. |
| virtual void Start() = 0; |
| // Stops stream activity. |
| // When a stream is stopped, it can't receive, process or deliver packets. |
| virtual void Stop() = 0; |
| |
| // TODO(pbos): Add info on currently-received codec to Stats. |
| virtual Stats GetStats() const = 0; |
| |
| virtual std::vector<RtpSource> GetSources() const = 0; |
| |
| // Sets a base minimum for the playout delay. Base minimum delay sets lower |
| // bound on minimum delay value determining lower bound on playout delay. |
| // |
| // Returns true if value was successfully set, false overwise. |
| virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0; |
| |
| // Returns current value of base minimum delay in milliseconds. |
| virtual int GetBaseMinimumPlayoutDelayMs() const = 0; |
| |
| // Allows a FrameDecryptor to be attached to a VideoReceiveStream after |
| // creation without resetting the decoder state. |
| virtual void SetFrameDecryptor( |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) = 0; |
| |
| // Allows a frame transformer to be attached to a VideoReceiveStream after |
| // creation without resetting the decoder state. |
| virtual void SetDepacketizerToDecoderFrameTransformer( |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0; |
| |
| // Sets and returns recording state. The old state is moved out |
| // of the video receive stream and returned to the caller, and |state| |
| // is moved in. If the state's callback is set, it will be called with |
| // recordable encoded frames as they arrive. |
| // If |generate_key_frame| is true, the method will generate a key frame. |
| // When the function returns, it's guaranteed that all old callouts |
| // to the returned callback has ceased. |
| // Note: the client should not interpret the returned state's attributes, but |
| // instead treat it as opaque data. |
| virtual RecordingState SetAndGetRecordingState(RecordingState state, |
| bool generate_key_frame) = 0; |
| |
| // Cause eventual generation of a key frame from the sender. |
| virtual void GenerateKeyFrame() = 0; |
| |
| protected: |
| virtual ~VideoReceiveStream() {} |
| }; |
| |
| class DEPRECATED_VideoReceiveStream : public VideoReceiveStream { |
| public: |
| // RtpDemuxer only forwards a given RTP packet to one sink. However, some |
| // sinks, such as FlexFEC, might wish to be informed of all of the packets |
| // a given sink receives (or any set of sinks). They may do so by registering |
| // themselves as secondary sinks. |
| virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0; |
| virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_VIDEO_RECEIVE_STREAM_H_ |