Add PeerConnectionInterface::UpdateCallBitrate with call tests.

webrtc/call/call.h

 /*
  *  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 WEBRTC_CALL_CALL_H_
 #define WEBRTC_CALL_CALL_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/api/rtcerror.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/call/audio_receive_stream.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/flexfec_receive_stream.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/common_types.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioProcessing;
 class RtcEventLog;
 
 const char* Version();
@@ skipping 35 matching lines @@
     static const int kDefaultStartBitrateBps;
 
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
       int min_bitrate_bps = 0;
       int start_bitrate_bps = kDefaultStartBitrateBps;
       int max_bitrate_bps = -1;
     } bitrate_config;
 
+    // TODO(zstein): Consider using PeerConnectionInterface::BitrateParameters
+    // instead (and move BitrateParameters to its own file in api/).
+    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);
 
+  static Call* Create(

[Taylor_Brandstetter, 2017/04/26 15:46:11]

Could you add a comment that this Create method is provided so that
"transport_send" can be mocked/faked for testing?

[Zach Stein, 2017/05/04 22:32:43]

Done.

+      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(
@@ skipping 16 matching lines @@
 
   // 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;
 
-  // TODO(pbos): Like BitrateConfig above this is currently per-stream instead
-  // of maximum for entire Call. This should be fixed along with the above.
-  // Specifying a start bitrate (>0) will currently reset the current bitrate
-  // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
+  // The min and start values will only be used if they are not set by
+  // SetBitrateConfigMask. The minimum max set by the two calls 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.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
 
+  // The min and start values set here are preferred to values set by
+  // SetBitrateConfig. The minimum of the max set by the two calls will be used.
+  // Assumes 0 <= min <= start <= max holds for set parameters.
+  virtual RTCError SetBitrateConfigMask(
+      const Config::BitrateConfigMask& bitrate_mask) = 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 ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_CALL_H_