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