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Diffstat (limited to 'third_party/libwebrtc/call/video_receive_stream.h')
| -rw-r--r-- | third_party/libwebrtc/call/video_receive_stream.h | 344 |
1 files changed, 344 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/video_receive_stream.h b/third_party/libwebrtc/call/video_receive_stream.h new file mode 100644 index 0000000000..0dc7dee71d --- /dev/null +++ b/third_party/libwebrtc/call/video_receive_stream.h @@ -0,0 +1,344 @@ +/* + * 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 <cstdint> +#include <limits> +#include <map> +#include <set> +#include <string> +#include <utility> +#include <vector> + +#include "api/call/transport.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "api/crypto/crypto_options.h" +#include "api/rtp_headers.h" +#include "api/rtp_parameters.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/receive_stream.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" +#include "rtc_base/checks.h" + +namespace webrtc { + +class RtpPacketSinkInterface; +class VideoDecoderFactory; + +class VideoReceiveStreamInterface : public MediaReceiveStreamInterface { + 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 VideoReceiveStreamInterface. The + // VideoReceiveStreamInterface client should not interpret the attribute. + std::function<void(const RecordableEncodedFrame&)> callback; + // Memento of when a keyframe request was last sent. The + // VideoReceiveStreamInterface 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(SdpVideoFormat video_format, int payload_type); + Decoder(); + Decoder(const Decoder&); + ~Decoder(); + + bool operator==(const Decoder& other) const; + + 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. + absl::optional<std::string> decoder_implementation_name; + absl::optional<bool> power_efficient_decoder; + 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-rtcinboundrtpstreamstats-jitterbufferdelay + TimeDelta jitter_buffer_delay = TimeDelta::Zero(); + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbuffertargetdelay + TimeDelta jitter_buffer_target_delay = TimeDelta::Zero(); + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferemittedcount + uint64_t jitter_buffer_emitted_count = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay + TimeDelta jitter_buffer_minimum_delay = TimeDelta::Zero(); + 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-rtcreceivedrtpstreamstats-packetsdiscarded + uint64_t packets_discarded = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime + TimeDelta total_decode_time = TimeDelta::Zero(); + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay + TimeDelta total_processing_delay = TimeDelta::Zero(); + // TODO(bugs.webrtc.org/13986): standardize + TimeDelta total_assembly_time = TimeDelta::Zero(); + uint32_t frames_assembled_from_multiple_packets = 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; + + 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; + absl::optional<RtpReceiveStats> rtx_rtp_stats; + + // Mozilla modification: Init these. + uint32_t rtcp_sender_packets_sent = 0; + uint32_t rtcp_sender_octets_sent = 0; + int64_t rtcp_sender_ntp_timestamp_ms = 0; + int64_t rtcp_sender_remote_ntp_timestamp_ms = 0; + + // 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&&); + Config(Transport* rtcp_send_transport, + VideoDecoderFactory* decoder_factory = nullptr); + 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 : public ReceiveStreamRtpConfig { + Rtp(); + Rtp(const Rtp&); + ~Rtp(); + std::string ToString() const; + + // See NackConfig for description. + NackConfig nack; + + // 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; + + // How to request keyframes from a remote sender. Applies only if lntf is + // disabled. + KeyFrameReqMethod keyframe_method = KeyFrameReqMethod::kPliRtcp; + + // See draft-alvestrand-rmcat-remb for information. + bool remb = false; + + bool tmmbr = false; + + // See LntfConfig for description. + LntfConfig lntf; + + // 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 handlers 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; + + RtcpEventObserver* rtcp_event_observer = nullptr; + } 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; + + // 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; + }; + + // TODO(pbos): Add info on currently-received codec to Stats. + virtual Stats GetStats() 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; + + // 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; + + virtual void SetRtcpMode(RtcpMode mode) = 0; + + // Sets or clears a flexfec RTP sink. This affects `rtp.packet_sink_` and + // `rtp.protected_by_flexfec` parts of the configuration. Must be called on + // the packet delivery thread. + // TODO(bugs.webrtc.org/11993): Packet delivery thread today means `worker + // thread` but will be `network thread`. + virtual void SetFlexFecProtection(RtpPacketSinkInterface* flexfec_sink) = 0; + + // Turns on/off loss notifications. Must be called on the packet delivery + // thread. + virtual void SetLossNotificationEnabled(bool enabled) = 0; + + // Modify `rtp.nack.rtp_history_ms` post construction. Setting this value + // to 0 disables nack. + // Must be called on the packet delivery thread. + virtual void SetNackHistory(TimeDelta history) = 0; + + virtual void SetProtectionPayloadTypes(int red_payload_type, + int ulpfec_payload_type) = 0; + + virtual void SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) = 0; + + virtual void SetAssociatedPayloadTypes( + std::map<int, int> associated_payload_types) = 0; + + virtual void UpdateRtxSsrc(uint32_t ssrc) = 0; + + protected: + virtual ~VideoReceiveStreamInterface() {} +}; + +} // namespace webrtc + +#endif // CALL_VIDEO_RECEIVE_STREAM_H_ |