| /* |
| * 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 WEBRTC_VIDEO_RECEIVE_STREAM_H_ |
| #define WEBRTC_VIDEO_RECEIVE_STREAM_H_ |
| |
| #include <limits> |
| #include <map> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/base/platform_file.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/common_video/include/frame_callback.h" |
| #include "webrtc/config.h" |
| #include "webrtc/media/base/videosinkinterface.h" |
| #include "webrtc/transport.h" |
| |
| namespace webrtc { |
| |
| class VideoDecoder; |
| |
| class VideoReceiveStream { |
| public: |
| // TODO(mflodman) Move all these settings to VideoDecoder and move the |
| // declaration to common_types.h. |
| struct Decoder { |
| std::string ToString() const; |
| |
| // The actual decoder instance. |
| VideoDecoder* decoder = nullptr; |
| |
| // Received RTP packets with this payload type will be sent to this decoder |
| // instance. |
| int payload_type = 0; |
| |
| // Name of the decoded payload (such as VP8). Maps back to the depacketizer |
| // used to unpack incoming packets. |
| std::string payload_name; |
| |
| DecoderSpecificSettings decoder_specific; |
| }; |
| |
| struct Stats { |
| std::string ToString(int64_t time_ms) const; |
| |
| int network_frame_rate = 0; |
| int decode_frame_rate = 0; |
| int render_frame_rate = 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; |
| int min_playout_delay_ms = 0; |
| int render_delay_ms = 10; |
| |
| int current_payload_type = -1; |
| |
| int total_bitrate_bps = 0; |
| int discarded_packets = 0; |
| |
| int width = 0; |
| int height = 0; |
| |
| int sync_offset_ms = std::numeric_limits<int>::max(); |
| |
| uint32_t ssrc = 0; |
| std::string c_name; |
| StreamDataCounters rtp_stats; |
| RtcpPacketTypeCounter rtcp_packet_type_counts; |
| RtcpStatistics rtcp_stats; |
| }; |
| |
| 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&) = default; |
| |
| public: |
| Config() = delete; |
| Config(Config&&) = default; |
| explicit Config(Transport* rtcp_send_transport) |
| : rtcp_send_transport(rtcp_send_transport) {} |
| |
| Config& operator=(Config&&) = default; |
| Config& operator=(const Config&) = delete; |
| |
| // 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; |
| |
| // Receive-stream specific RTP settings. |
| struct 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-alvestrand-rmcat-remb for information. |
| bool remb = false; |
| |
| // See draft-holmer-rmcat-transport-wide-cc-extensions for details. |
| bool transport_cc = false; |
| |
| // See NackConfig for description. |
| NackConfig nack; |
| |
| // See UlpfecConfig for description. |
| UlpfecConfig ulpfec; |
| |
| // RTX settings for incoming video payloads that may be received. RTX is |
| // disabled if there's no config present. |
| struct Rtx { |
| // SSRCs to use for the RTX streams. |
| uint32_t ssrc = 0; |
| |
| // Payload type to use for the RTX stream. |
| int payload_type = 0; |
| }; |
| |
| // Map from video RTP payload type -> RTX config. |
| typedef std::map<int, Rtx> RtxMap; |
| RtxMap rtx; |
| |
| // If set to true, the RTX payload type mapping supplied in |rtx| will be |
| // used when restoring RTX packets. Without it, RTX packets will always be |
| // restored to the last non-RTX packet payload type received. |
| bool use_rtx_payload_mapping_on_restore = false; |
| |
| // RTP header extensions used for the received stream. |
| std::vector<RtpExtension> extensions; |
| } rtp; |
| |
| // Transport for outgoing packets (RTCP). |
| Transport* rtcp_send_transport = nullptr; |
| |
| // Must not be 'nullptr' when the stream is started. |
| 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. |
| // Only valid if 'renderer' is set. |
| int render_delay_ms = 10; |
| |
| // If set, pass frames on to the renderer as soon as they are |
| // available. |
| bool disable_prerenderer_smoothing = false; |
| |
| // 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; |
| |
| // Called for each incoming video frame, i.e. in encoded state. E.g. used |
| // when |
| // saving the stream to a file. 'nullptr' disables the callback. |
| EncodedFrameObserver* pre_decode_callback = nullptr; |
| |
| // Called for each decoded frame. E.g. used when adding effects to the |
| // decoded |
| // stream. 'nullptr' disables the callback. |
| // TODO(tommi): This seems to be only used by a test or two. Consider |
| // removing it (and use an appropriate alternative in the tests) as well |
| // as the associated code in VideoStreamDecoder. |
| I420FrameCallback* pre_render_callback = nullptr; |
| |
| // 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; |
| }; |
| |
| // 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; |
| |
| // Takes ownership of the file, is responsible for closing it later. |
| // Calling this method will close and finalize any current log. |
| // Giving rtc::kInvalidPlatformFileValue disables logging. |
| // If a frame to be written would make the log too large the write fails and |
| // the log is closed and finalized. A |byte_limit| of 0 means no limit. |
| virtual void EnableEncodedFrameRecording(rtc::PlatformFile file, |
| size_t byte_limit) = 0; |
| inline void DisableEncodedFrameRecording() { |
| EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0); |
| } |
| |
| protected: |
| virtual ~VideoReceiveStream() {} |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ |