blob: e54a51d901b495cbe5f072f7d8fd56345a194311 [file] [log] [blame]
* 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 <stdint.h>
#include <map>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.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 "call/rtp_config.h"
#include "common_video/frame_counts.h"
#include "common_video/include/quality_limitation_reason.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "video/config/video_encoder_config.h"
namespace webrtc {
class FrameEncryptorInterface;
class VideoSendStream {
// Multiple StreamStats objects are present if simulcast is used (multiple
// kMedia streams) or if RTX or FlexFEC is negotiated. Multiple SVC layers, on
// the other hand, does not cause additional StreamStats.
struct StreamStats {
enum class StreamType {
// A media stream is an RTP stream for audio or video. Retransmissions and
// FEC is either sent over the same SSRC or negotiated to be sent over
// separate SSRCs, in which case separate StreamStats objects exist with
// references to this media stream's SSRC.
// RTX streams are streams dedicated to retransmissions. They have a
// dependency on a single kMedia stream: `referenced_media_ssrc`.
// FlexFEC streams are streams dedicated to FlexFEC. They have a
// dependency on a single kMedia stream: `referenced_media_ssrc`.
std::string ToString() const;
StreamType type = StreamType::kMedia;
// If `type` is kRtx or kFlexfec this value is present. The referenced SSRC
// is the kMedia stream that this stream is performing retransmissions or
// FEC for. If `type` is kMedia, this value is null.
absl::optional<uint32_t> referenced_media_ssrc;
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;
// `avg_delay_ms` and `max_delay_ms` are only used in tests. Consider
// deleting.
int avg_delay_ms = 0;
int max_delay_ms = 0;
StreamDataCounters rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
absl::optional<ReportBlockData> report_block_data;
double encode_frame_rate = 0.0;
int frames_encoded = 0;
absl::optional<uint64_t> qp_sum;
uint64_t total_encode_time_ms = 0;
uint64_t total_encoded_bytes_target = 0;
uint32_t huge_frames_sent = 0;
struct Stats {
std::string ToString(int64_t time_ms) const;
std::string encoder_implementation_name = "unknown";
double 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;
uint64_t total_encode_time_ms = 0;
uint64_t total_encoded_bytes_target = 0;
uint32_t frames = 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_congestion_window = 0;
uint32_t frames_dropped_by_encoder = 0;
// 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;
QualityLimitationReason quality_limitation_reason =
std::map<QualityLimitationReason, int64_t> quality_limitation_durations_ms;
uint32_t quality_limitation_resolution_changes = 0;
// 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 =
uint32_t frames_sent = 0;
uint32_t huge_frames_sent = 0;
absl::optional<bool> power_efficient_encoder;
struct Config {
Config() = delete;
explicit Config(Transport* send_transport);
Config& operator=(Config&&);
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;
RtpConfig rtp;
VideoStreamEncoderSettings encoder_settings;
// Time interval between RTCP report for video
int rtcp_report_interval_ms = 1000;
// Transport for outgoing packets.
Transport* send_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;
// 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;
// An optional encoder selector provided by the user.
// Overrides VideoEncoderFactory::GetEncoderSelector().
// Owned by RtpSenderBase.
VideoEncoderFactory::EncoderSelectorInterface* encoder_selector = nullptr;
// Per PeerConnection cryptography options.
CryptoOptions crypto_options;
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
// 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(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;
// Accessor for determining if the stream is active. This is an inexpensive
// call that must be made on the same thread as `Start()` and `Stop()` methods
// are called on and will return `true` iff activity has been started either
// via `Start()` or `UpdateActiveSimulcastLayers()`. If activity is either
// stopped or is in the process of being stopped as a result of a call to
// either `Stop()` or `UpdateActiveSimulcastLayers()` where all layers were
// deactivated, the return value will be `false`.
virtual bool started() = 0;
// If the resource is overusing, the VideoSendStream will try to reduce
// resolution or frame rate until no resource is overusing.
// TODO( When the ResourceAdaptationProcessor
// is moved to Call this method could be deleted altogether in favor of
// Call-level APIs only.
virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
virtual std::vector<rtc::scoped_refptr<Resource>>
GetAdaptationResources() = 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 void ReconfigureVideoEncoder(VideoEncoderConfig config,
SetParametersCallback callback) = 0;
virtual Stats GetStats() = 0;
virtual void GenerateKeyFrame(const std::vector<std::string>& rids) = 0;
virtual ~VideoSendStream() {}
} // namespace webrtc