1 files changed, 48 insertions, 5 deletions

diff --git a/chromium/third_party/webrtc/video_engine/include/vie_rtp_rtcp.h b/chromium/third_party/webrtc/video_engine/include/vie_rtp_rtcp.h
index 9d899ad7f39..972ca58486e 100644
--- a/chromium/third_party/webrtc/video_engine/include/vie_rtp_rtcp.h
+++ b/chromium/third_party/webrtc/video_engine/include/vie_rtp_rtcp.h
@@ -27,6 +27,7 @@
 namespace webrtc {
 
 class VideoEngine;
+struct ReceiveBandwidthEstimatorStats;
 
 // This enumerator sets the RTCP mode.
 enum ViERTCPMode {
@@ -48,11 +49,6 @@ enum StreamType {
   kViEStreamTypeRtx = 1  // Retransmission media stream
 };
 
-enum BandwidthEstimationMode {
-  kViEMultiStreamEstimation,
-  kViESingleStreamEstimation
-};
-
 // This class declares an abstract interface for a user defined observer. It is
 // up to the VideoEngine user to implement a derived class which implements the
 // observer class. The observer is registered using RegisterRTPObserver() and
@@ -138,6 +134,15 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   virtual int SetRtxSendPayloadType(const int video_channel,
                                     const uint8_t payload_type) = 0;
 
+  // This enables sending redundant payloads when padding up the bitrate instead
+  // of sending dummy padding packets. This feature is off by default and will
+  // only have an effect if RTX is also enabled.
+  // TODO(holmer): Remove default implementation once this has been implemented
+  // in libjingle.
+  virtual int SetPadWithRedundantPayloads(int video_channel, bool enable) {
+    return 0;
+  }
+
   virtual int SetRtxReceivePayloadType(const int video_channel,
                                        const uint8_t payload_type) = 0;
 
@@ -265,6 +270,22 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   virtual int SetTransmissionSmoothingStatus(int video_channel,
                                              bool enable) = 0;
 
+  // Sets a minimal bitrate which will be padded to when the encoder doesn't
+  // produce enough bitrate.
+  // TODO(pbos): Remove default implementation when libjingle's
+  // FakeWebRtcVideoEngine is updated.
+  virtual int SetMinTransmitBitrate(int video_channel,
+                                    int min_transmit_bitrate_kbps) {
+    return -1;
+  };
+
+  // Set a constant amount to deduct from received bitrate estimates before
+  // using it to allocate capacity among outgoing video streams.
+  virtual int SetReservedTransmitBitrate(
+      int video_channel, unsigned int reserved_transmit_bitrate_bps) {
+    return 0;
+  }
+
   // This function returns our locally created statistics of the received RTP
   // stream.
   virtual int GetReceiveChannelRtcpStatistics(const int video_channel,
@@ -359,6 +380,14 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   virtual int DeregisterReceiveChannelRtpStatisticsCallback(
       int video_channel, StreamDataCountersCallback* callback) = 0;
 
+
+  // Gets sent and received RTCP packet types.
+  // TODO(asapersson): Remove default implementation.
+  virtual int GetRtcpPacketTypeCounters(
+      int video_channel,
+      RtcpPacketTypeCounter* packets_sent,
+      RtcpPacketTypeCounter* packets_received) const { return -1; }
+
   // The function gets bandwidth usage statistics from the sent RTP streams in
   // bits/s.
   virtual int GetBandwidthUsage(const int video_channel,
@@ -389,6 +418,20 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
       const int video_channel,
       unsigned int* estimated_bandwidth) const = 0;
 
+  // This function gets the receive-side bandwidth esitmator statistics.
+  // TODO(jiayl): remove the default impl when libjingle's FakeWebRtcVideoEngine
+  // is updated.
+  virtual int GetReceiveBandwidthEstimatorStats(
+      const int video_channel,
+      ReceiveBandwidthEstimatorStats* output) const { return -1; }
+
+  // This function gets the PacedSender queuing delay for the last sent frame.
+  // TODO(jiayl): remove the default impl when libjingle is updated.
+  virtual int GetPacerQueuingDelayMs(
+      const int video_channel, int* delay_ms) const {
+    return -1;
+  }
+
   // This function enables capturing of RTP packets to a binary file on a
   // specific channel and for a given direction. The file can later be
   // replayed using e.g. RTP Tools rtpplay since the binary file format is