Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 274 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/video_send_stream.h b/third_party/libwebrtc/call/video_send_stream.h
new file mode 100644
index 0000000000..de18fc7b92
--- /dev/null
+++ b/third_party/libwebrtc/call/video_send_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_SEND_STREAM_H_
+#define CALL_VIDEO_SEND_STREAM_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/adaptation/resource.h"
+#include "api/call/transport.h"
+#include "api/crypto/crypto_options.h"
+#include "api/frame_transformer_interface.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_sender_setparameters_callback.h"
+#include "api/scoped_refptr.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_source_interface.h"
+#include "api/video/video_stream_encoder_settings.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "call/rtp_config.h"
+#include "common_video/frame_counts.h"
+#include "common_video/include/quality_limitation_reason.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+
+class FrameEncryptorInterface;
+
+class VideoSendStream {
+ public:
+  // Multiple StreamStats objects are present if simulcast is used (multiple
+  // kMedia streams) or if RTX or FlexFEC is negotiated. Multiple SVC layers, on
+  // the other hand, does not cause additional StreamStats.
+  struct StreamStats {
+    enum class StreamType {
+      // A media stream is an RTP stream for audio or video. Retransmissions and
+      // FEC is either sent over the same SSRC or negotiated to be sent over
+      // separate SSRCs, in which case separate StreamStats objects exist with
+      // references to this media stream's SSRC.
+      kMedia,
+      // RTX streams are streams dedicated to retransmissions. They have a
+      // dependency on a single kMedia stream: `referenced_media_ssrc`.
+      kRtx,
+      // FlexFEC streams are streams dedicated to FlexFEC. They have a
+      // dependency on a single kMedia stream: `referenced_media_ssrc`.
+      kFlexfec,
+    };
+
+    StreamStats();
+    ~StreamStats();
+
+    std::string ToString() const;
+
+    StreamType type = StreamType::kMedia;
+    // If `type` is kRtx or kFlexfec this value is present. The referenced SSRC
+    // is the kMedia stream that this stream is performing retransmissions or
+    // FEC for. If `type` is kMedia, this value is null.
+    absl::optional<uint32_t> referenced_media_ssrc;
+    FrameCounts frame_counts;
+    int width = 0;
+    int height = 0;
+    // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
+    int total_bitrate_bps = 0;
+    int retransmit_bitrate_bps = 0;
+    // `avg_delay_ms` and `max_delay_ms` are only used in tests. Consider
+    // deleting.
+    int avg_delay_ms = 0;
+    int max_delay_ms = 0;
+    StreamDataCounters rtp_stats;
+    RtcpPacketTypeCounter rtcp_packet_type_counts;
+    // A snapshot of the most recent Report Block with additional data of
+    // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
+    absl::optional<ReportBlockData> report_block_data;
+    double encode_frame_rate = 0.0;
+    int frames_encoded = 0;
+    absl::optional<uint64_t> qp_sum;
+    uint64_t total_encode_time_ms = 0;
+    uint64_t total_encoded_bytes_target = 0;
+    uint32_t huge_frames_sent = 0;
+    absl::optional<ScalabilityMode> scalability_mode;
+  };
+
+  struct Stats {
+    Stats();
+    ~Stats();
+    std::string ToString(int64_t time_ms) const;
+    std::string encoder_implementation_name = "unknown";
+    double input_frame_rate = 0;
+    int encode_frame_rate = 0;
+    int avg_encode_time_ms = 0;
+    int encode_usage_percent = 0;
+    uint32_t frames_encoded = 0;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime
+    uint64_t total_encode_time_ms = 0;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
+    uint64_t total_encoded_bytes_target = 0;
+    uint32_t frames = 0;
+    uint32_t frames_dropped_by_capturer = 0;
+    uint32_t frames_dropped_by_encoder_queue = 0;
+    uint32_t frames_dropped_by_rate_limiter = 0;
+    uint32_t frames_dropped_by_congestion_window = 0;
+    uint32_t frames_dropped_by_encoder = 0;
+    // Bitrate the encoder is currently configured to use due to bandwidth
+    // limitations.
+    int target_media_bitrate_bps = 0;
+    // Bitrate the encoder is actually producing.
+    int media_bitrate_bps = 0;
+    bool suspended = false;
+    bool bw_limited_resolution = false;
+    bool cpu_limited_resolution = false;
+    bool bw_limited_framerate = false;
+    bool cpu_limited_framerate = false;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationreason
+    QualityLimitationReason quality_limitation_reason =
+        QualityLimitationReason::kNone;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations
+    std::map<QualityLimitationReason, int64_t> quality_limitation_durations_ms;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
+    uint32_t quality_limitation_resolution_changes = 0;
+    // Total number of times resolution as been requested to be changed due to
+    // CPU/quality adaptation.
+    int number_of_cpu_adapt_changes = 0;
+    int number_of_quality_adapt_changes = 0;
+    bool has_entered_low_resolution = false;
+    std::map<uint32_t, StreamStats> substreams;
+    webrtc::VideoContentType content_type =
+        webrtc::VideoContentType::UNSPECIFIED;
+    uint32_t frames_sent = 0;
+    uint32_t huge_frames_sent = 0;
+    absl::optional<bool> power_efficient_encoder;
+  };
+
+  struct Config {
+   public:
+    Config() = delete;
+    Config(Config&&);
+    explicit Config(Transport* 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;
+
+    RtpConfig rtp;
+
+    VideoStreamEncoderSettings encoder_settings;
+
+    // Time interval between RTCP report for video
+    int rtcp_report_interval_ms = 1000;
+
+    // Transport for outgoing packets.
+    Transport* send_transport = nullptr;
+
+    // Expected delay needed by the renderer, i.e. the frame will be delivered
+    // this many milliseconds, if possible, earlier than expected render time.
+    // Only valid if `local_renderer` is set.
+    int render_delay_ms = 0;
+
+    // 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;
+
+    // True if the stream should be suspended when the available bitrate fall
+    // below the minimum configured bitrate. If this variable is false, the
+    // stream may send at a rate higher than the estimated available bitrate.
+    bool suspend_below_min_bitrate = false;
+
+    // Enables periodic bandwidth probing in application-limited region.
+    bool periodic_alr_bandwidth_probing = false;
+
+    // An optional custom frame encryptor that allows the entire frame to be
+    // encrypted in whatever way the caller chooses. This is not required by
+    // default.
+    rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor;
+
+    // An optional encoder selector provided by the user.
+    // Overrides VideoEncoderFactory::GetEncoderSelector().
+    // Owned by RtpSenderBase.
+    VideoEncoderFactory::EncoderSelectorInterface* encoder_selector = nullptr;
+
+    // Per PeerConnection cryptography options.
+    CryptoOptions crypto_options;
+
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
+
+   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&);
+  };
+
+  // Updates the sending state for all simulcast layers that the video send
+  // stream owns. This can mean updating the activity one or for multiple
+  // layers. The ordering of active layers is the order in which the
+  // rtp modules are stored in the VideoSendStream.
+  // Note: This starts stream activity if it is inactive and one of the layers
+  // is active. This stops stream activity if it is active and all layers are
+  // inactive.
+  // `active_layers` should have the same size as the number of configured
+  // simulcast layers or one if only one rtp stream is used.
+  virtual void StartPerRtpStream(std::vector<bool> active_layers) = 0;
+
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  // Prefer to use StartPerRtpStream.
+  virtual void Start() = 0;
+
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+
+  // Accessor for determining if the stream is active. This is an inexpensive
+  // call that must be made on the same thread as `Start()` and `Stop()` methods
+  // are called on and will return `true` iff activity has been started either
+  // via `Start()` or `StartPerRtpStream()`. If activity is either
+  // stopped or is in the process of being stopped as a result of a call to
+  // either `Stop()` or `StartPerRtpStream()` where all layers were
+  // deactivated, the return value will be `false`.
+  virtual bool started() = 0;
+
+  // If the resource is overusing, the VideoSendStream will try to reduce
+  // resolution or frame rate until no resource is overusing.
+  // TODO(https://crbug.com/webrtc/11565): When the ResourceAdaptationProcessor
+  // is moved to Call this method could be deleted altogether in favor of
+  // Call-level APIs only.
+  virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
+  virtual std::vector<rtc::scoped_refptr<Resource>>
+  GetAdaptationResources() = 0;
+
+  virtual void SetSource(
+      rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
+      const DegradationPreference& degradation_preference) = 0;
+
+  // Set which streams to send. Must have at least as many SSRCs as configured
+  // in the config. Encoder settings are passed on to the encoder instance along
+  // with the VideoStream settings.
+  virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;
+
+  virtual void ReconfigureVideoEncoder(VideoEncoderConfig config,
+                                       SetParametersCallback callback) = 0;
+
+  virtual Stats GetStats() = 0;
+
+  virtual void GenerateKeyFrame(const std::vector<std::string>& rids) = 0;
+
+ protected:
+  virtual ~VideoSendStream() {}
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_VIDEO_SEND_STREAM_H_