blob: f199ca1f851dac7ae89b79ca9680bf22cc75e817 [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 <string>
#include <vector>
#include "webrtc/api/call/transport.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/call/rtp_config.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/include/frame_callback.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/rtc_base/platform_file.h"
namespace webrtc {
class RtpPacketSinkInterface;
class VideoDecoder;
class VideoReceiveStream {
// 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;
// The actual decoder instance.
VideoDecoder* decoder = nullptr;
// Received RTP packets with this payload type will be sent to this decoder
// instance.
int payload_type = 0;
// Name of the decoded payload (such as VP8). Maps back to the depacketizer
// used to unpack incoming packets.
std::string payload_name;
// This map contains the codec specific parameters from SDP, i.e. the "fmtp"
// parameters. It is the same as cricket::CodecParameterMap used in
// cricket::VideoCodec.
std::map<std::string, std::string> codec_params;
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;
int min_playout_delay_ms = 0;
int render_delay_ms = 10;
int64_t interframe_delay_max_ms = -1;
uint32_t frames_decoded = 0;
rtc::Optional<uint64_t> qp_sum;
int current_payload_type = -1;
int total_bitrate_bps = 0;
int discarded_packets = 0;
int width = 0;
int height = 0;
VideoContentType content_type = VideoContentType::UNSPECIFIED;
int sync_offset_ms = std::numeric_limits<int>::max();
uint32_t ssrc = 0;
std::string c_name;
StreamDataCounters rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
RtcpStatistics rtcp_stats;
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
rtc::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;
// 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;
// TODO(nisse): This remb setting is currently set but never
// applied. REMB logic is now the responsibility of
// PacketRouter, and it will generate REMB feedback if
// OnReceiveBitrateChanged is used, which depends on how the
// estimators belonging to the ReceiveSideCongestionController
// are configured. Decide if this setting should be deleted, and
// if it needs to be replaced by a setting in PacketRouter to
// disable REMB feedback.
// See draft-alvestrand-rmcat-remb for information.
bool remb = false;
// See draft-holmer-rmcat-transport-wide-cc-extensions for details.
bool transport_cc = false;
// See NackConfig for description.
NackConfig nack;
// See UlpfecConfig for description.
// TODO(nisse): UlpfecConfig includes the field red_rtx_payload_type,
// which duplicates info in the rtx_associated_payload_types mapping. So
// delete the use of UlpfecConfig here, and replace by the values which
// make sense in this context, likely those are ulpfec_payload_type_ and
// red_payload_type_.
UlpfecConfig ulpfec;
// SSRC for retransmissions.
uint32_t rtx_ssrc = 0;
// Set if the stream is protected using FlexFEC.
bool protected_by_flexfec = false;
// 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;
// TODO(nisse): This is a temporary accessor function to enable
// reversing and renaming of the rtx_payload_types mapping.
void AddRtxBinding(int rtx_payload_type, int media_payload_type) {
rtx_associated_payload_types[rtx_payload_type] = media_payload_type;
// RTP header extensions used for the received stream.
std::vector<RtpExtension> extensions;
} rtp;
// Transport for outgoing packets (RTCP).
Transport* rtcp_send_transport = nullptr;
// Must not be 'nullptr' when the stream is started.
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.
// Only valid if 'renderer' is set.
int render_delay_ms = 10;
// If set, pass frames on to the renderer as soon as they are
// available.
bool disable_prerenderer_smoothing = false;
// 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;
// Called for each incoming video frame, i.e. in encoded state. E.g. used
// when
// saving the stream to a file. 'nullptr' disables the callback.
EncodedFrameObserver* pre_decode_callback = nullptr;
// 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;
// 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;
// Takes ownership of the file, is responsible for closing it later.
// Calling this method will close and finalize any current log.
// Giving rtc::kInvalidPlatformFileValue disables logging.
// If a frame to be written would make the log too large the write fails and
// the log is closed and finalized. A |byte_limit| of 0 means no limit.
virtual void EnableEncodedFrameRecording(rtc::PlatformFile file,
size_t byte_limit) = 0;
inline void DisableEncodedFrameRecording() {
EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0);
// 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;
virtual ~VideoReceiveStream() {}
} // namespace webrtc