blob: 1bd8e946cce5c6a41930c90bd29ded258872fae6 [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 <limits>
#include <map>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/recordable_encoded_frame.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_timing.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/rtp_config.h"
#include "common_video/frame_counts.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class RtpPacketSinkInterface;
class VideoDecoderFactory;
class VideoReceiveStream {
// Class for handling moving in/out recording state.
struct RecordingState {
RecordingState() = default;
explicit RecordingState(
std::function<void(const RecordableEncodedFrame&)> callback)
: callback(std::move(callback)) {}
// Callback stored from the VideoReceiveStream. The VideoReceiveStream
// client should not interpret the attribute.
std::function<void(const RecordableEncodedFrame&)> callback;
// Memento of when a keyframe request was last sent. The VideoReceiveStream
// client should not interpret the attribute.
absl::optional<int64_t> last_keyframe_request_ms;
// TODO(mflodman) Move all these settings to VideoDecoder and move the
// declaration to common_types.h.
struct Decoder {
Decoder(const Decoder&);
std::string ToString() const;
SdpVideoFormat video_format;
// Received RTP packets with this payload type will be sent to this decoder
// instance.
int payload_type = 0;
struct Stats {
std::string ToString(int64_t time_ms) const;
int network_frame_rate = 0;
int decode_frame_rate = 0;
int render_frame_rate = 0;
uint32_t frames_rendered = 0;
// Decoder stats.
std::string decoder_implementation_name = "unknown";
FrameCounts frame_counts;
int decode_ms = 0;
int max_decode_ms = 0;
int current_delay_ms = 0;
int target_delay_ms = 0;
int jitter_buffer_ms = 0;
double jitter_buffer_delay_seconds = 0;
uint64_t jitter_buffer_emitted_count = 0;
int min_playout_delay_ms = 0;
int render_delay_ms = 10;
int64_t interframe_delay_max_ms = -1;
// Frames dropped due to decoding failures or if the system is too slow.
uint32_t frames_dropped = 0;
uint32_t frames_decoded = 0;
uint64_t total_decode_time_ms = 0;
// Total inter frame delay in seconds.
double total_inter_frame_delay = 0;
// Total squared inter frame delay in seconds^2.
double total_squared_inter_frame_delay = 0;
int64_t first_frame_received_to_decoded_ms = -1;
absl::optional<uint64_t> qp_sum;
int current_payload_type = -1;
int total_bitrate_bps = 0;
int width = 0;
int height = 0;
uint32_t freeze_count = 0;
uint32_t pause_count = 0;
uint32_t total_freezes_duration_ms = 0;
uint32_t total_pauses_duration_ms = 0;
uint32_t total_frames_duration_ms = 0;
double sum_squared_frame_durations = 0.0;
VideoContentType content_type = VideoContentType::UNSPECIFIED;
absl::optional<int64_t> estimated_playout_ntp_timestamp_ms;
int sync_offset_ms = std::numeric_limits<int>::max();
uint32_t ssrc = 0;
std::string c_name;
RtpReceiveStats rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
struct Config {
// 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&);
Config() = delete;
explicit Config(Transport* rtcp_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;
// Decoders for every payload that we can receive.
std::vector<Decoder> decoders;
// Ownership stays with WebrtcVideoEngine (delegated from PeerConnection).
VideoDecoderFactory* decoder_factory = nullptr;
// Receive-stream specific RTP settings.
struct Rtp {
Rtp(const Rtp&);
std::string ToString() const;
// Synchronization source (stream identifier) to be received.
uint32_t remote_ssrc = 0;
// Sender SSRC used for sending RTCP (such as receiver reports).
uint32_t local_ssrc = 0;
// See RtcpMode for description.
RtcpMode rtcp_mode = RtcpMode::kCompound;
// Extended RTCP settings.
struct RtcpXr {
// True if RTCP Receiver Reference Time Report Block extension
// (RFC 3611) should be enabled.
bool receiver_reference_time_report = false;
} rtcp_xr;
// See draft-holmer-rmcat-transport-wide-cc-extensions for details.
bool transport_cc = false;
// See LntfConfig for description.
LntfConfig lntf;
// See NackConfig for description.
NackConfig nack;
// Payload types for ULPFEC and RED, respectively.
int ulpfec_payload_type = -1;
int red_payload_type = -1;
// SSRC for retransmissions.
uint32_t rtx_ssrc = 0;
// Set if the stream is protected using FlexFEC.
bool protected_by_flexfec = false;
// Optional callback sink to support additional packet handlsers such as
// FlexFec.
RtpPacketSinkInterface* packet_sink_ = nullptr;
// Map from rtx payload type -> media payload type.
// For RTX to be enabled, both an SSRC and this mapping are needed.
std::map<int, int> rtx_associated_payload_types;
// Payload types that should be depacketized using raw depacketizer
// (payload header will not be parsed and must not be present, additional
// meta data is expected to be present in generic frame descriptor
// RTP header extension).
std::set<int> raw_payload_types;
// RTP header extensions used for the received stream.
std::vector<RtpExtension> extensions;
} rtp;
// Transport for outgoing packets (RTCP).
Transport* rtcp_send_transport = nullptr;
// Must always be set.
rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
// Expected delay needed by the renderer, i.e. the frame will be delivered
// this many milliseconds, if possible, earlier than the ideal render time.
int render_delay_ms = 10;
// If false, pass frames on to the renderer as soon as they are
// available.
bool enable_prerenderer_smoothing = true;
// Identifier for an A/V synchronization group. Empty string to disable.
// TODO(pbos): Synchronize streams in a sync group, not just video streams
// to one of the audio streams.
std::string sync_group;
// 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;
// An optional custom frame decryptor that allows the entire frame to be
// decrypted in whatever way the caller choses. This is not required by
// default.
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor;
// Per PeerConnection cryptography options.
CryptoOptions crypto_options;
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
// 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;
// TODO(pbos): Add info on currently-received codec to Stats.
virtual Stats GetStats() const = 0;
virtual std::vector<RtpSource> GetSources() const = 0;
// Sets a base minimum for the playout delay. Base minimum delay sets lower
// bound on minimum delay value determining lower bound on playout delay.
// Returns true if value was successfully set, false overwise.
virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
// Returns current value of base minimum delay in milliseconds.
virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
// Allows a FrameDecryptor to be attached to a VideoReceiveStream after
// creation without resetting the decoder state.
virtual void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) = 0;
// Allows a frame transformer to be attached to a VideoReceiveStream after
// creation without resetting the decoder state.
virtual void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0;
// Sets and returns recording state. The old state is moved out
// of the video receive stream and returned to the caller, and |state|
// is moved in. If the state's callback is set, it will be called with
// recordable encoded frames as they arrive.
// If |generate_key_frame| is true, the method will generate a key frame.
// When the function returns, it's guaranteed that all old callouts
// to the returned callback has ceased.
// Note: the client should not interpret the returned state's attributes, but
// instead treat it as opaque data.
virtual RecordingState SetAndGetRecordingState(RecordingState state,
bool generate_key_frame) = 0;
// Cause eventual generation of a key frame from the sender.
virtual void GenerateKeyFrame() = 0;
virtual ~VideoReceiveStream() {}
class DEPRECATED_VideoReceiveStream : public VideoReceiveStream {
// RtpDemuxer only forwards a given RTP packet to one sink. However, some
// sinks, such as FlexFEC, might wish to be informed of all of the packets
// a given sink receives (or any set of sinks). They may do so by registering
// themselves as secondary sinks.
virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0;
virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0;
} // namespace webrtc