Adding upstream version 86.0.1.upstream/86.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 270 insertions, 0 deletions

diff --git a/third_party/libwebrtc/webrtc/call/video_receive_stream.h b/third_party/libwebrtc/webrtc/call/video_receive_stream.h
new file mode 100644
index 0000000000..c315ecfcce
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/call/video_receive_stream.h
@@ -0,0 +1,270 @@
+/*
+ *  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_RECEIVE_STREAM_H_
+#define CALL_VIDEO_RECEIVE_STREAM_H_
+
+#include <limits>
+#include <map>
+#include <string>
+#include <vector>
+
+#include "api/call/transport.h"
+#include "api/rtpparameters.h"
+#include "call/rtp_config.h"
+#include "common_types.h"  // NOLINT(build/include)
+#include "common_video/include/frame_callback.h"
+#include "media/base/videosinkinterface.h"
+#include "rtc_base/platform_file.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class RtpPacketSinkInterface;
+class VideoDecoder;
+
+class VideoReceiveStream {
+ public:
+  // TODO(mflodman) Move all these settings to VideoDecoder and move the
+  // declaration to common_types.h.
+  struct Decoder {
+    Decoder();
+    Decoder(const Decoder&);
+    ~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 {
+    Stats();
+    ~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;
+
+    uint32_t rtcp_sender_packets_sent;
+    uint32_t rtcp_sender_octets_sent;
+    NtpTime rtcp_sender_ntp_timestamp;
+
+    // Timing frame info: all important timestamps for a full lifetime of a
+    // single 'timing frame'.
+    rtc::Optional<webrtc::TimingFrameInfo> timing_frame_info;
+  };
+
+  struct Config {
+   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&);
+
+   public:
+    Config() = delete;
+    Config(Config&&);
+    explicit Config(Transport* rtcp_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;
+
+    // Decoders for every payload that we can receive.
+    std::vector<Decoder> decoders;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp {
+      Rtp();
+      Rtp(const Rtp&);
+      ~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;
+
+      bool tmmbr = false;
+
+      // See draft-holmer-rmcat-transport-wide-cc-extensions for details.
+      bool transport_cc = false;
+
+      // TODO(jesup) - there should be a kKeyFrameReqNone
+      KeyFrameRequestMethod keyframe_method = kKeyFrameReqPliRtcp;
+
+      // 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;
+
+      // 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;
+
+    // Called when a RTCP bye or timeout occurs. 'nullptr' disables the
+    // callback.
+    RtcpEventObserver* rtcp_event_observer = nullptr;
+  };
+
+  // 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;
+
+ protected:
+  virtual ~VideoReceiveStream() {}
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_VIDEO_RECEIVE_STREAM_H_