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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.
*/
#ifndef CALL_VIDEO_SEND_STREAM_H_
#define CALL_VIDEO_SEND_STREAM_H_
#include <stdint.h>
#include <map>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/media_transport_interface.h"
#include "api/rtp_parameters.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_source_interface.h"
#include "api/video/video_stream_encoder_settings.h"
#include "api/video_codecs/video_encoder_config.h"
#include "call/rtp_config.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class FrameEncryptorInterface;
class VideoSendStream {
public:
struct StreamStats {
StreamStats();
~StreamStats();
std::string ToString() const;
FrameCounts frame_counts;
bool is_rtx = false;
bool is_flexfec = false;
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 {
Stats();
~Stats();
std::string ToString(int64_t time_ms) const;
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;
uint32_t frames_encoded = 0;
uint32_t frames_dropped_by_capturer = 0;
uint32_t frames_dropped_by_encoder_queue = 0;
uint32_t frames_dropped_by_rate_limiter = 0;
uint32_t frames_dropped_by_encoder = 0;
absl::optional<uint64_t> qp_sum;
// Bitrate the encoder is currently configured to use due to bandwidth
// limitations.
int target_media_bitrate_bps = 0;
// Bitrate the encoder is actually producing.
int media_bitrate_bps = 0;
bool suspended = false;
bool bw_limited_resolution = false;
bool cpu_limited_resolution = false;
bool bw_limited_framerate = false;
bool cpu_limited_framerate = false;
// Total number of times resolution as been requested to be changed due to
// CPU/quality adaptation.
int number_of_cpu_adapt_changes = 0;
int number_of_quality_adapt_changes = 0;
bool has_entered_low_resolution = false;
std::map<uint32_t, StreamStats> substreams;
webrtc::VideoContentType content_type =
webrtc::VideoContentType::UNSPECIFIED;
uint32_t huge_frames_sent = 0;
};
struct Config {
public:
Config() = delete;
Config(Config&&);
Config(Transport* send_transport, MediaTransportInterface* media_transport);
explicit Config(Transport* send_transport);
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;
VideoStreamEncoderSettings encoder_settings;
RtpConfig rtp;
// Time interval between RTCP report for video
int rtcp_report_interval_ms = 1000;
// Transport for outgoing packets.
Transport* send_transport = nullptr;
MediaTransportInterface* media_transport = 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;
// Enables periodic bandwidth probing in application-limited region.
bool periodic_alr_bandwidth_probing = false;
// Track ID as specified during track creation.
std::string track_id;
// An optional custom frame encryptor that allows the entire frame to be
// encrypted in whatever way the caller chooses. This is not required by
// default.
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor;
// Per PeerConnection cryptography options.
CryptoOptions crypto_options;
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&);
};
// Updates the sending state for all simulcast layers that the video send
// stream owns. This can mean updating the activity one or for multiple
// layers. The ordering of active layers is the order in which the
// rtp modules are stored in the VideoSendStream.
// Note: This starts stream activity if it is inactive and one of the layers
// is active. This stops stream activity if it is active and all layers are
// inactive.
virtual void UpdateActiveSimulcastLayers(
const std::vector<bool> active_layers) = 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;
virtual void SetSource(
rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
const DegradationPreference& degradation_preference) = 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(VideoEncoderConfig config) = 0;
virtual Stats GetStats() = 0;
protected:
virtual ~VideoSendStream() {}
};
} // namespace webrtc
#endif // CALL_VIDEO_SEND_STREAM_H_