Adding upstream version 86.0.1.upstream/86.0.1upstream

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

1 files changed, 214 insertions, 0 deletions

diff --git a/third_party/libwebrtc/webrtc/call/call.h b/third_party/libwebrtc/webrtc/call/call.h
new file mode 100644
index 0000000000..84b2f73ef0
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/call/call.h
@@ -0,0 +1,214 @@
+/*
+ *  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_CALL_H_
+#define CALL_CALL_H_
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/rtcerror.h"
+#include "call/audio_receive_stream.h"
+#include "call/audio_send_stream.h"
+#include "call/audio_state.h"
+#include "call/flexfec_receive_stream.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "common_types.h"  // NOLINT(build/include)
+#include "rtc_base/bitrateallocationstrategy.h"
+#include "rtc_base/networkroute.h"
+#include "rtc_base/platform_file.h"
+#include "rtc_base/socket.h"
+
+namespace webrtc {
+
+class AudioProcessing;
+class RtcEventLog;
+
+enum class MediaType {
+  ANY,
+  AUDIO,
+  VIDEO,
+  DATA
+};
+
+// Like std::min, but considers non-positive values to be unset.
+// TODO(zstein): Remove once all callers use rtc::Optional.
+template <typename T>
+static T MinPositive(T a, T b) {
+  if (a <= 0) {
+    return b;
+  }
+  if (b <= 0) {
+    return a;
+  }
+  return std::min(a, b);
+}
+
+class PacketReceiver {
+ public:
+  enum DeliveryStatus {
+    DELIVERY_OK,
+    DELIVERY_UNKNOWN_SSRC,
+    DELIVERY_PACKET_ERROR,
+  };
+
+  virtual DeliveryStatus DeliverPacket(MediaType media_type,
+                                       const uint8_t* packet,
+                                       size_t length,
+                                       const PacketTime& packet_time) = 0;
+
+ protected:
+  virtual ~PacketReceiver() {}
+};
+
+// A Call instance can contain several send and/or receive streams. All streams
+// are assumed to have the same remote endpoint and will share bitrate estimates
+// etc.
+class Call {
+ public:
+  struct Config {
+    explicit Config(RtcEventLog* event_log) : event_log(event_log) {
+      RTC_DCHECK(event_log);
+    }
+
+    static constexpr int kDefaultStartBitrateBps = 300000;
+
+    // Bitrate config used until valid bitrate estimates are calculated. Also
+    // used to cap total bitrate used. This comes from the remote connection.
+    struct BitrateConfig {
+      int min_bitrate_bps = 0;
+      int start_bitrate_bps = kDefaultStartBitrateBps;
+      int max_bitrate_bps = -1;
+    } bitrate_config;
+
+    // The local client's bitrate preferences. The actual configuration used
+    // is a combination of this and |bitrate_config|. The combination is
+    // currently more complicated than a simple mask operation (see
+    // SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
+    // start <= max holds for set parameters.
+    struct BitrateConfigMask {
+      rtc::Optional<int> min_bitrate_bps;
+      rtc::Optional<int> start_bitrate_bps;
+      rtc::Optional<int> max_bitrate_bps;
+    };
+
+    // AudioState which is possibly shared between multiple calls.
+    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+    rtc::scoped_refptr<AudioState> audio_state;
+
+    // Audio Processing Module to be used in this call.
+    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+    AudioProcessing* audio_processing = nullptr;
+
+    // RtcEventLog to use for this call. Required.
+    // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
+    RtcEventLog* event_log = nullptr;
+  };
+
+  struct Stats {
+    std::string ToString(int64_t time_ms) const;
+
+    int send_bandwidth_bps = 0;       // Estimated available send bandwidth.
+    int max_padding_bitrate_bps = 0;  // Cumulative configured max padding.
+    int recv_bandwidth_bps = 0;       // Estimated available receive bandwidth.
+    int64_t pacer_delay_ms = 0;
+    int64_t rtt_ms = -1;
+  };
+
+  static Call* Create(const Call::Config& config);
+
+  // Allows mocking |transport_send| for testing.
+  static Call* Create(
+      const Call::Config& config,
+      std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
+
+  virtual AudioSendStream* CreateAudioSendStream(
+      const AudioSendStream::Config& config) = 0;
+  virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
+
+  virtual AudioReceiveStream* CreateAudioReceiveStream(
+      const AudioReceiveStream::Config& config) = 0;
+  virtual void DestroyAudioReceiveStream(
+      AudioReceiveStream* receive_stream) = 0;
+
+  virtual VideoSendStream* CreateVideoSendStream(
+      VideoSendStream::Config config,
+      VideoEncoderConfig encoder_config) = 0;
+  virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
+
+  virtual VideoReceiveStream* CreateVideoReceiveStream(
+      VideoReceiveStream::Config configuration) = 0;
+  virtual void DestroyVideoReceiveStream(
+      VideoReceiveStream* receive_stream) = 0;
+
+  // In order for a created VideoReceiveStream to be aware that it is
+  // protected by a FlexfecReceiveStream, the latter should be created before
+  // the former.
+  virtual FlexfecReceiveStream* CreateFlexfecReceiveStream(
+      const FlexfecReceiveStream::Config& config) = 0;
+  virtual void DestroyFlexfecReceiveStream(
+      FlexfecReceiveStream* receive_stream) = 0;
+
+  // All received RTP and RTCP packets for the call should be inserted to this
+  // PacketReceiver. The PacketReceiver pointer is valid as long as the
+  // Call instance exists.
+  virtual PacketReceiver* Receiver() = 0;
+
+  // Returns the call statistics, such as estimated send and receive bandwidth,
+  // pacing delay, etc.
+  virtual Stats GetStats() const = 0;
+
+  // The greater min and smaller max set by this and SetBitrateConfigMask will
+  // be used. The latest non-negative start value from either call will be used.
+  // Specifying a start bitrate (>0) will reset the current bitrate estimate.
+  // This is due to how the 'x-google-start-bitrate' flag is currently
+  // implemented. Passing -1 leaves the start bitrate unchanged. Behavior is not
+  // guaranteed for other negative values or 0.
+  virtual void SetBitrateConfig(
+      const Config::BitrateConfig& bitrate_config) = 0;
+
+  // The greater min and smaller max set by this and SetBitrateConfig will be
+  // used. The latest non-negative start value form either call will be used.
+  // Specifying a start bitrate will reset the current bitrate estimate.
+  // Assumes 0 <= min <= start <= max holds for set parameters.
+  virtual void SetBitrateConfigMask(
+      const Config::BitrateConfigMask& bitrate_mask) = 0;
+
+  virtual void SetBitrateAllocationStrategy(
+      std::unique_ptr<rtc::BitrateAllocationStrategy>
+          bitrate_allocation_strategy) = 0;
+
+  // TODO(skvlad): When the unbundled case with multiple streams for the same
+  // media type going over different networks is supported, track the state
+  // for each stream separately. Right now it's global per media type.
+  virtual void SignalChannelNetworkState(MediaType media,
+                                         NetworkState state) = 0;
+
+  virtual void OnTransportOverheadChanged(
+      MediaType media,
+      int transport_overhead_per_packet) = 0;
+
+  virtual void OnNetworkRouteChanged(
+      const std::string& transport_name,
+      const rtc::NetworkRoute& network_route) = 0;
+
+  virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
+
+  virtual VoiceEngine* voice_engine() = 0;
+
+  virtual ~Call() {}
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_CALL_H_