blob: 0adcf3fc0d4cd3f499ae562dbd91ffbcfb03a21b [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/common_types.h"
#include "webrtc/common_video/include/frame_callback.h"
#include "webrtc/config.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/transport.h"
namespace webrtc {
class VideoDecoder;
class VideoReceiveStream {
// TODO(mflodman) Move all these settings to VideoDecoder and move the
// declaration to common_types.h.
struct 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;
DecoderSpecificSettings decoder_specific;
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;
// 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;
int current_payload_type = -1;
int total_bitrate_bps = 0;
int discarded_packets = 0;
int width = 0;
int height = 0;
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;
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&) = default;
Config() = delete;
Config(Config&&) = default;
explicit Config(Transport* rtcp_send_transport)
: rtcp_send_transport(rtcp_send_transport) {}
Config& operator=(Config&&) = default;
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 {
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-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 FecConfig for description.
FecConfig fec;
// RTX settings for incoming video payloads that may be received. RTX is
// disabled if there's no config present.
struct Rtx {
// SSRCs to use for the RTX streams.
uint32_t ssrc = 0;
// Payload type to use for the RTX stream.
int payload_type = 0;
// Map from video RTP payload type -> RTX config.
typedef std::map<int, Rtx> RtxMap;
RtxMap rtx;
// If set to true, the RTX payload type mapping supplied in |rtx| will be
// used when restoring RTX packets. Without it, RTX packets will always be
// restored to the last non-RTX packet payload type received.
bool use_rtx_payload_mapping_on_restore = false;
// RTP header extensions used for the received stream.
std::vector<RtpExtension> extensions;
} rtp;
// Transport for outgoing packets (RTCP).
Transport* rtcp_send_transport = nullptr;
// VideoRenderer will be called for each decoded frame. 'nullptr' disables
// rendering of this stream.
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;
// Called for each decoded frame. E.g. used when adding effects to the
// decoded
// stream. 'nullptr' disables the callback.
// TODO(tommi): This seems to be only used by a test or two. Consider
// removing it (and use an appropriate alternative in the tests) as well
// as the associated code in VideoStreamDecoder.
I420FrameCallback* pre_render_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;
virtual ~VideoReceiveStream() {}
} // namespace webrtc