Adding upstream version 110.0.1.upstream/110.0.1upstream

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

1 files changed, 336 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/video_receive_stream.h b/third_party/libwebrtc/call/video_receive_stream.h
new file mode 100644
index 0000000000..bcc6555c31
--- /dev/null
+++ b/third_party/libwebrtc/call/video_receive_stream.h
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+/*
+ *  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 <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/call/transport.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "api/crypto/crypto_options.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.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/receive_stream.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 VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
+ public:
+  // 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 VideoReceiveStreamInterface. The
+    // VideoReceiveStreamInterface client should not interpret the attribute.
+    std::function<void(const RecordableEncodedFrame&)> callback;
+    // Memento of when a keyframe request was last sent. The
+    // VideoReceiveStreamInterface 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(SdpVideoFormat video_format, int payload_type);
+    Decoder();
+    Decoder(const Decoder&);
+    ~Decoder();
+
+    bool operator==(const Decoder& other) const;
+
+    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 {
+    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;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcvideoreceiverstats-jitterbufferdelay
+    double jitter_buffer_delay_seconds = 0;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcvideoreceiverstats-jitterbufferemittedcount
+    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.
+    // https://www.w3.org/TR/webrtc-stats/#dom-rtcvideoreceiverstats-framesdropped
+    uint32_t frames_dropped = 0;
+    uint32_t frames_decoded = 0;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsdiscarded
+    uint64_t packets_discarded = 0;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime
+    TimeDelta total_decode_time = TimeDelta::Zero();
+    // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
+    TimeDelta total_processing_delay = TimeDelta::Zero();
+    // TODO(bugs.webrtc.org/13986): standardize
+    TimeDelta total_assembly_time = TimeDelta::Zero();
+    uint32_t frames_assembled_from_multiple_packets = 0;
+    // Total inter frame delay in seconds.
+    // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalinterframedelay
+    double total_inter_frame_delay = 0;
+    // Total squared inter frame delay in seconds^2.
+    // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalsqauredinterframedelay
+    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;
+
+    // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
+    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;
+
+    // Mozilla modification: Init these.
+    uint32_t rtcp_sender_packets_sent = 0;
+    uint32_t rtcp_sender_octets_sent = 0;
+    int64_t rtcp_sender_ntp_timestamp_ms = 0;
+    int64_t rtcp_sender_remote_ntp_timestamp_ms = 0;
+
+    // Timing frame info: all important timestamps for a full lifetime of a
+    // single 'timing frame'.
+    absl::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&&);
+    Config(Transport* rtcp_send_transport,
+           VideoDecoderFactory* decoder_factory = nullptr);
+    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;
+
+    // Ownership stays with WebrtcVideoEngine (delegated from PeerConnection).
+    VideoDecoderFactory* decoder_factory = nullptr;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp : public ReceiveStreamRtpConfig {
+      Rtp();
+      Rtp(const Rtp&);
+      ~Rtp();
+      std::string ToString() const;
+
+      // See NackConfig for description.
+      NackConfig nack;
+
+      // 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;
+
+      // How to request keyframes from a remote sender. Applies only if lntf is
+      // disabled.
+      KeyFrameReqMethod keyframe_method = KeyFrameReqMethod::kPliRtcp;
+
+      // See draft-alvestrand-rmcat-remb for information.
+      bool remb = false;
+
+      bool tmmbr = false;
+
+      // See LntfConfig for description.
+      LntfConfig lntf;
+
+      // 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 handlers 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;
+
+      RtcpEventObserver* rtcp_event_observer = nullptr;
+    } 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;
+
+    // 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;
+  };
+
+  // TODO(pbos): Add info on currently-received codec to Stats.
+  virtual Stats GetStats() 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;
+
+  // 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 void SetRtcpMode(RtcpMode mode) = 0;
+
+  // Sets or clears a flexfec RTP sink. This affects `rtp.packet_sink_` and
+  // `rtp.protected_by_flexfec` parts of the configuration. Must be called on
+  // the packet delivery thread.
+  // TODO(bugs.webrtc.org/11993): Packet delivery thread today means `worker
+  // thread` but will be `network thread`.
+  virtual void SetFlexFecProtection(RtpPacketSinkInterface* flexfec_sink) = 0;
+
+  // Turns on/off loss notifications. Must be called on the packet delivery
+  // thread.
+  virtual void SetLossNotificationEnabled(bool enabled) = 0;
+
+  // Modify `rtp.nack.rtp_history_ms` post construction. Setting this value
+  // to 0 disables nack.
+  // Must be called on the packet delivery thread.
+  virtual void SetNackHistory(TimeDelta history) = 0;
+
+  virtual void SetProtectionPayloadTypes(int red_payload_type,
+                                         int ulpfec_payload_type) = 0;
+
+  virtual void SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) = 0;
+
+  virtual void SetAssociatedPayloadTypes(
+      std::map<int, int> associated_payload_types) = 0;
+
+ protected:
+  virtual ~VideoReceiveStreamInterface() {}
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_VIDEO_RECEIVE_STREAM_H_