247 lines
9.4 KiB
C
247 lines
9.4 KiB
C
|
/*
|
||
|
|
* 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/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/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 "api/video_codecs/video_encoder_config.h"
|
||
|
|
#include "call/rtp_config.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"
|
||
|
|
|
||
|
|
namespace webrtc {
|
||
|
|
|
||
|
|
class FrameEncryptorInterface;
|
||
|
|
|
||
|
|
class VideoSendStream {
|
||
|
|
public:
|
||
|
|
// 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.
|
||
|
|
kMedia,
|
||
|
|
// RTX streams are streams dedicated to retransmissions. They have a
|
||
|
|
// dependency on a single kMedia stream: |referenced_media_ssrc|.
|
||
|
|
kRtx,
|
||
|
|
// FlexFEC streams are streams dedicated to FlexFEC. They have a
|
||
|
|
// dependency on a single kMedia stream: |referenced_media_ssrc|.
|
||
|
|
kFlexfec,
|
||
|
|
};
|
||
|
|
|
||
|
|
StreamStats();
|
||
|
|
~StreamStats();
|
||
|
|
|
||
|
|
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;
|
||
|
|
int avg_delay_ms = 0;
|
||
|
|
int max_delay_ms = 0;
|
||
|
|
uint64_t total_packet_send_delay_ms = 0;
|
||
|
|
StreamDataCounters rtp_stats;
|
||
|
|
RtcpPacketTypeCounter rtcp_packet_type_counts;
|
||
|
|
RtcpStatistics rtcp_stats;
|
||
|
|
// 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 {
|
||
|
|
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;
|
||
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime
|
||
|
|
uint64_t total_encode_time_ms = 0;
|
||
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
|
||
|
|
uint64_t total_encoded_bytes_target = 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;
|
||
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationreason
|
||
|
|
QualityLimitationReason quality_limitation_reason =
|
||
|
|
QualityLimitationReason::kNone;
|
||
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations
|
||
|
|
std::map<QualityLimitationReason, int64_t> quality_limitation_durations_ms;
|
||
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
|
||
|
|
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 =
|
||
|
|
webrtc::VideoContentType::UNSPECIFIED;
|
||
|
|
uint32_t frames_sent = 0;
|
||
|
|
uint32_t huge_frames_sent = 0;
|
||
|
|
};
|
||
|
|
|
||
|
|
struct Config {
|
||
|
|
public:
|
||
|
|
Config() = delete;
|
||
|
|
Config(Config&&);
|
||
|
|
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;
|
||
|
|
|
||
|
|
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;
|
||
|
|
|
||
|
|
// Per PeerConnection cryptography options.
|
||
|
|
CryptoOptions crypto_options;
|
||
|
|
|
||
|
|
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
|
||
|
|
|
||
|
|
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;
|
||
|
|
|
||
|
|
// If the resource is overusing, the VideoSendStream will try to reduce
|
||
|
|
// resolution or frame rate until no resource is overusing.
|
||
|
|
// TODO(https://crbug.com/webrtc/11565): 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 Stats GetStats() = 0;
|
||
|
|
|
||
|
|
protected:
|
||
|
|
virtual ~VideoSendStream() {}
|
||
|
|
};
|
||
|
|
|
||
|
|
} // namespace webrtc
|
||
|
|
|
||
|
|
#endif // CALL_VIDEO_SEND_STREAM_H_
|