211 lines
6.6 KiB
C++
211 lines
6.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_RECEIVE_STREAM_H_
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#define WEBRTC_VIDEO_RECEIVE_STREAM_H_
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#include <limits>
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#include <map>
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#include <string>
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#include <vector>
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#include "webrtc/common_types.h"
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#include "webrtc/common_video/include/frame_callback.h"
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#include "webrtc/config.h"
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#include "webrtc/media/base/videosinkinterface.h"
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#include "webrtc/transport.h"
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namespace webrtc {
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class VideoDecoder;
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class VideoReceiveStream {
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public:
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// TODO(mflodman) Move all these settings to VideoDecoder and move the
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// declaration to common_types.h.
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struct Decoder {
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std::string ToString() const;
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// The actual decoder instance.
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VideoDecoder* decoder = nullptr;
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// Received RTP packets with this payload type will be sent to this decoder
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// instance.
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int payload_type = 0;
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// Name of the decoded payload (such as VP8). Maps back to the depacketizer
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// used to unpack incoming packets.
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std::string payload_name;
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};
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struct Stats {
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int network_frame_rate = 0;
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int decode_frame_rate = 0;
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int render_frame_rate = 0;
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// Decoder stats.
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std::string decoder_implementation_name = "unknown";
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FrameCounts frame_counts;
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int decode_ms = 0;
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int max_decode_ms = 0;
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int current_delay_ms = 0;
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int target_delay_ms = 0;
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int jitter_buffer_ms = 0;
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int min_playout_delay_ms = 0;
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int render_delay_ms = 10;
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int current_payload_type = -1;
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int total_bitrate_bps = 0;
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int discarded_packets = 0;
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int sync_offset_ms = std::numeric_limits<int>::max();
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uint32_t ssrc = 0;
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std::string c_name;
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StreamDataCounters rtp_stats;
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RtcpPacketTypeCounter rtcp_packet_type_counts;
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RtcpStatistics rtcp_stats;
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};
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struct Config {
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private:
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// Access to the copy constructor is private to force use of the Copy()
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// method for those exceptional cases where we do use it.
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Config(const Config&) = default;
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public:
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Config() = delete;
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Config(Config&&) = default;
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explicit Config(Transport* rtcp_send_transport)
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: rtcp_send_transport(rtcp_send_transport) {}
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Config& operator=(Config&&) = default;
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Config& operator=(const Config&) = delete;
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// Mostly used by tests. Avoid creating copies if you can.
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Config Copy() const { return Config(*this); }
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std::string ToString() const;
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// Decoders for every payload that we can receive.
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std::vector<Decoder> decoders;
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// Receive-stream specific RTP settings.
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struct Rtp {
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std::string ToString() const;
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// Synchronization source (stream identifier) to be received.
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uint32_t remote_ssrc = 0;
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// Sender SSRC used for sending RTCP (such as receiver reports).
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uint32_t local_ssrc = 0;
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// See RtcpMode for description.
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RtcpMode rtcp_mode = RtcpMode::kCompound;
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// Extended RTCP settings.
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struct RtcpXr {
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// True if RTCP Receiver Reference Time Report Block extension
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// (RFC 3611) should be enabled.
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bool receiver_reference_time_report = false;
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} rtcp_xr;
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// See draft-alvestrand-rmcat-remb for information.
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bool remb = false;
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// See draft-holmer-rmcat-transport-wide-cc-extensions for details.
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bool transport_cc = false;
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// See NackConfig for description.
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NackConfig nack;
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// See FecConfig for description.
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FecConfig fec;
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// RTX settings for incoming video payloads that may be received. RTX is
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// disabled if there's no config present.
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struct Rtx {
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// SSRCs to use for the RTX streams.
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uint32_t ssrc = 0;
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// Payload type to use for the RTX stream.
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int payload_type = 0;
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};
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// Map from video RTP payload type -> RTX config.
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typedef std::map<int, Rtx> RtxMap;
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RtxMap rtx;
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// If set to true, the RTX payload type mapping supplied in |rtx| will be
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// used when restoring RTX packets. Without it, RTX packets will always be
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// restored to the last non-RTX packet payload type received.
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bool use_rtx_payload_mapping_on_restore = false;
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// RTP header extensions used for the received stream.
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std::vector<RtpExtension> extensions;
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} rtp;
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// Transport for outgoing packets (RTCP).
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Transport* rtcp_send_transport = nullptr;
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// VideoRenderer will be called for each decoded frame. 'nullptr' disables
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// rendering of this stream.
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rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
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// Expected delay needed by the renderer, i.e. the frame will be delivered
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// this many milliseconds, if possible, earlier than the ideal render time.
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// Only valid if 'renderer' is set.
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int render_delay_ms = 10;
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// If set, pass frames on to the renderer as soon as they are
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// available.
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bool disable_prerenderer_smoothing = false;
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// Identifier for an A/V synchronization group. Empty string to disable.
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// TODO(pbos): Synchronize streams in a sync group, not just video streams
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// to one of the audio streams.
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std::string sync_group;
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// Called for each incoming video frame, i.e. in encoded state. E.g. used
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// when
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// saving the stream to a file. 'nullptr' disables the callback.
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EncodedFrameObserver* pre_decode_callback = nullptr;
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// Called for each decoded frame. E.g. used when adding effects to the
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// decoded
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// stream. 'nullptr' disables the callback.
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// TODO(tommi): This seems to be only used by a test or two. Consider
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// removing it (and use an appropriate alternative in the tests) as well
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// as the associated code in VideoStreamDecoder.
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I420FrameCallback* pre_render_callback = nullptr;
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// Target delay in milliseconds. A positive value indicates this stream is
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// used for streaming instead of a real-time call.
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int target_delay_ms = 0;
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};
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// Starts stream activity.
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// When a stream is active, it can receive, process and deliver packets.
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virtual void Start() = 0;
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// Stops stream activity.
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// When a stream is stopped, it can't receive, process or deliver packets.
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virtual void Stop() = 0;
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// TODO(pbos): Add info on currently-received codec to Stats.
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virtual Stats GetStats() const = 0;
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protected:
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virtual ~VideoReceiveStream() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_
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