194 lines
6.4 KiB
C++
194 lines
6.4 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_SEND_STREAM_H_
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#define WEBRTC_VIDEO_SEND_STREAM_H_
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#include <map>
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#include <string>
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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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#include "webrtc/media/base/videosinkinterface.h"
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namespace webrtc {
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class LoadObserver;
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class VideoEncoder;
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// Class to deliver captured frame to the video send stream.
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class VideoCaptureInput {
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public:
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// These methods do not lock internally and must be called sequentially.
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// If your application switches input sources synchronization must be done
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// externally to make sure that any old frames are not delivered concurrently.
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virtual void IncomingCapturedFrame(const VideoFrame& video_frame) = 0;
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protected:
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virtual ~VideoCaptureInput() {}
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};
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class VideoSendStream {
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public:
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struct StreamStats {
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FrameCounts frame_counts;
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int width = 0;
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int height = 0;
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// TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
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int total_bitrate_bps = 0;
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int retransmit_bitrate_bps = 0;
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int avg_delay_ms = 0;
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int max_delay_ms = 0;
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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 Stats {
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std::string encoder_implementation_name = "unknown";
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int input_frame_rate = 0;
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int encode_frame_rate = 0;
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int avg_encode_time_ms = 0;
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int encode_usage_percent = 0;
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int target_media_bitrate_bps = 0;
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int media_bitrate_bps = 0;
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bool suspended = false;
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bool bw_limited_resolution = false;
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std::map<uint32_t, StreamStats> substreams;
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};
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struct Config {
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Config() = delete;
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explicit Config(Transport* send_transport)
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: send_transport(send_transport) {}
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std::string ToString() const;
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struct EncoderSettings {
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std::string ToString() const;
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std::string payload_name;
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int payload_type = -1;
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// TODO(sophiechang): Delete this field when no one is using internal
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// sources anymore.
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bool internal_source = false;
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// Allow 100% encoder utilization. Used for HW encoders where CPU isn't
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// expected to be the limiting factor, but a chip could be running at
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// 30fps (for example) exactly.
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bool full_overuse_time = false;
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// Uninitialized VideoEncoder instance to be used for encoding. Will be
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// initialized from inside the VideoSendStream.
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VideoEncoder* encoder = nullptr;
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} encoder_settings;
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static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
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struct Rtp {
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std::string ToString() const;
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std::vector<uint32_t> ssrcs;
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// See RtcpMode for description.
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RtcpMode rtcp_mode = RtcpMode::kCompound;
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// Max RTP packet size delivered to send transport from VideoEngine.
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size_t max_packet_size = kDefaultMaxPacketSize;
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// RTP header extensions to use for this send stream.
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std::vector<RtpExtension> extensions;
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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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// Settings for RTP retransmission payload format, see RFC 4588 for
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// details.
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struct Rtx {
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std::string ToString() const;
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// SSRCs to use for the RTX streams.
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std::vector<uint32_t> ssrcs;
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// Payload type to use for the RTX stream.
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int payload_type = -1;
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} rtx;
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// RTCP CNAME, see RFC 3550.
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std::string c_name;
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} rtp;
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// Transport for outgoing packets.
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Transport* send_transport = nullptr;
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// Callback for overuse and normal usage based on the jitter of incoming
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// captured frames. 'nullptr' disables the callback.
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LoadObserver* overuse_callback = nullptr;
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// Called for each I420 frame before encoding the frame. Can be used for
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// effects, snapshots etc. 'nullptr' disables the callback.
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rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;
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// Called for each encoded frame, e.g. used for file storage. 'nullptr'
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// disables the callback. Also measures timing and passes the time
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// spent on encoding. This timing will not fire if encoding takes longer
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// than the measuring window, since the sample data will have been dropped.
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EncodedFrameObserver* post_encode_callback = nullptr;
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// Renderer for local preview. The local renderer will be called even if
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// sending hasn't started. 'nullptr' disables local rendering.
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rtc::VideoSinkInterface<VideoFrame>* local_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 expected render time.
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// Only valid if |local_renderer| is set.
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int render_delay_ms = 0;
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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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// True if the stream should be suspended when the available bitrate fall
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// below the minimum configured bitrate. If this variable is false, the
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// stream may send at a rate higher than the estimated available bitrate.
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bool suspend_below_min_bitrate = false;
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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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// Gets interface used to insert captured frames. Valid as long as the
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// VideoSendStream is valid.
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virtual VideoCaptureInput* Input() = 0;
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// Set which streams to send. Must have at least as many SSRCs as configured
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// in the config. Encoder settings are passed on to the encoder instance along
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// with the VideoStream settings.
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virtual void ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
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virtual Stats GetStats() = 0;
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protected:
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virtual ~VideoSendStream() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_SEND_STREAM_H_
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