Add PeerConnectionInterface::UpdateCallBitrate with call tests.

webrtc/call/call.cc

 /*
  *  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.
  */
 
@@ skipping 124 matching lines @@
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override;
 
   // Implements RecoveredPacketReceiver.
   bool OnRecoveredPacket(const uint8_t* packet, size_t length) override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
 
+  void SetBitrateConfigMask(
+      const webrtc::Call::Config::BitrateConfigMask& bitrate_config) override;
+
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
   void OnTransportOverheadChanged(MediaType media,
                                   int transport_overhead_per_packet) override;
 
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
 
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
@@ skipping 26 matching lines @@
                                                   size_t length,
                                                   const PacketTime& packet_time)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
   void UpdateSendHistograms(int64_t first_sent_packet_ms)
       EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
   void UpdateHistograms();
   void UpdateAggregateNetworkState();
 
+  // These values only get set if there is actually something to change.
+  struct BitrateUpdate {
+    rtc::Optional<int> min_bitrate_bps;
+    rtc::Optional<int> start_bitrate_bps;
+    rtc::Optional<int> max_bitrate_bps;
+  };
+
+  // Applies update to the BitrateConfig cached in |config_| and updates the
+  // congestion controller if update is not empty.
+  void UpdateCurrentBitrateConfig(const BitrateUpdate& update);
+
   Clock* const clock_;
 
   const int num_cpu_cores_;
   const std::unique_ptr<ProcessThread> module_process_thread_;
   const std::unique_ptr<ProcessThread> pacer_thread_;
   const std::unique_ptr<CallStats> call_stats_;
   const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
@@ skipping 74 matching lines @@
   std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
   ReceiveSideCongestionController receive_side_cc_;
   const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
   const int64_t start_ms_;
   // TODO(perkj): |worker_queue_| is supposed to replace
   // |module_process_thread_|.
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue worker_queue_;
 
+  // The config mask set by SetBitrateConfigMask.
+  // 0 <= min <= start <= max
+  Config::BitrateConfigMask bitrate_config_mask_;
+
+  // The config set by SetBitrateConfig.
+  // min >= 0, start != 0, max == -1 || max > 0
+  Config::BitrateConfig base_bitrate_config_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 std::string Call::Stats::ToString(int64_t time_ms) const {
   std::stringstream ss;
   ss << "Call stats: " << time_ms << ", {";
   ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
   ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
   ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
@@ skipping 35 matching lines @@
       received_audio_bytes_per_second_counter_(clock_, nullptr, true),
       received_video_bytes_per_second_counter_(clock_, nullptr, true),
       received_rtcp_bytes_per_second_counter_(clock_, nullptr, true),
       min_allocated_send_bitrate_bps_(0),
       configured_max_padding_bitrate_bps_(0),
       estimated_send_bitrate_kbps_counter_(clock_, nullptr, true),
       pacer_bitrate_kbps_counter_(clock_, nullptr, true),
       receive_side_cc_(clock_, transport_send->packet_router()),
       video_send_delay_stats_(new SendDelayStats(clock_)),
       start_ms_(clock_->TimeInMilliseconds()),
-      worker_queue_("call_worker_queue") {
+      worker_queue_("call_worker_queue"),
+      base_bitrate_config_(config.bitrate_config) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(config.event_log != nullptr);
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   Trace::CreateTrace();
@@ skipping 504 matching lines @@
     rtc::CritScope cs(&bitrate_crit_);
     stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_;
   }
   return stats;
 }
 
 void Call::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+
   RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
-  if (bitrate_config.max_bitrate_bps != -1)
+  RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
+  if (bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
-  if (config_.bitrate_config.min_bitrate_bps ==
-          bitrate_config.min_bitrate_bps &&
-      (bitrate_config.start_bitrate_bps <= 0 ||
-       config_.bitrate_config.start_bitrate_bps ==
-           bitrate_config.start_bitrate_bps) &&
-      config_.bitrate_config.max_bitrate_bps ==
-          bitrate_config.max_bitrate_bps) {
-    // Nothing new to set, early abort to avoid encoder reconfigurations.
+  }
+
+  BitrateUpdate update;
+
+  const int min = bitrate_config.min_bitrate_bps;
+  if (min != base_bitrate_config_.min_bitrate_bps) {
+    base_bitrate_config_.min_bitrate_bps = min;
+    if (!bitrate_config_mask_.min_bitrate_bps ||
+        min > *bitrate_config_mask_.min_bitrate_bps) {
+      update.min_bitrate_bps = rtc::Optional<int>(min);
+      LOG(LS_INFO) << "WebRTC.Call.SetBitrateConfig: "
+                   << "Planning to update min to " << min;
+    }
+  }
+
+  const int start = bitrate_config.start_bitrate_bps;
+  if (start != -1 && start != base_bitrate_config_.start_bitrate_bps) {
+    base_bitrate_config_.start_bitrate_bps = start;
+    update.start_bitrate_bps = rtc::Optional<int>(start);
+    LOG(LS_INFO) << "WebRTC.Call.SetBitrateConfig: "
+                 << "Planning to update start to " << start;
+  }
+  const int max = bitrate_config.max_bitrate_bps;
+  if (max != -1 && max != base_bitrate_config_.max_bitrate_bps) {
+    base_bitrate_config_.max_bitrate_bps = max;
+    if (!bitrate_config_mask_.max_bitrate_bps ||
+        max < *bitrate_config_mask_.max_bitrate_bps) {
+      update.max_bitrate_bps = rtc::Optional<int>(max);
+      LOG(LS_INFO) << "WebRTC.Call.SetBitrateConfig: "
+                   << "Planning to update max to " << max;
+    }
+  }
+
+  UpdateCurrentBitrateConfig(update);
+}
+
+void Call::SetBitrateConfigMask(
+    const webrtc::Call::Config::BitrateConfigMask& mask) {
+  TRACE_EVENT0("webrtc", "Call::SetBitrateConfigMask");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+
+  bitrate_config_mask_ = mask;
+  BitrateUpdate update;
+
+  if (mask.min_bitrate_bps &&
+      *mask.min_bitrate_bps > base_bitrate_config_.min_bitrate_bps) {
+    update.min_bitrate_bps = mask.min_bitrate_bps;
+  }
+  update.start_bitrate_bps = mask.start_bitrate_bps;
+  if (mask.max_bitrate_bps &&
+      (base_bitrate_config_.max_bitrate_bps == -1 ||
+       *mask.max_bitrate_bps < base_bitrate_config_.max_bitrate_bps)) {
+    update.max_bitrate_bps = mask.max_bitrate_bps;
+  }
+
+  UpdateCurrentBitrateConfig(update);
+}
+
+void Call::UpdateCurrentBitrateConfig(const BitrateUpdate& update) {
+  if (!update.min_bitrate_bps && !update.start_bitrate_bps &&
+      !update.max_bitrate_bps) {
+    // Nothing to update.
+    LOG(LS_INFO) << "WebRTC.Call.UpdateCurrentBitrateConfig: "
+                 << "nothing to update";

[Taylor Brandstetter, 2017/05/11 23:14:08]

This could be spammy if it happens for every SDP offer/answer; can you change it
to LS_VERBOSE?

[Zach Stein, 2017/05/23 20:58:53]

Done.

     return;
   }
-  config_.bitrate_config.min_bitrate_bps = bitrate_config.min_bitrate_bps;
-  // Start bitrate of -1 means we should keep the old bitrate, which there is
-  // no point in remembering for the future.
-  if (bitrate_config.start_bitrate_bps > 0)
-    config_.bitrate_config.start_bitrate_bps = bitrate_config.start_bitrate_bps;
-  config_.bitrate_config.max_bitrate_bps = bitrate_config.max_bitrate_bps;
-  RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
+
+  Config::BitrateConfig updated;
+  const Config::BitrateConfig& current = config_.bitrate_config;
+  updated.min_bitrate_bps =
+      update.min_bitrate_bps.value_or(current.min_bitrate_bps);
+  updated.start_bitrate_bps =
+      update.start_bitrate_bps.value_or(current.start_bitrate_bps);
+  updated.max_bitrate_bps =
+      update.max_bitrate_bps.value_or(current.max_bitrate_bps);
+
+  // TODO(zstein): This clamp logic is different than send_side_cc's (moves min
+  // instead of max, allows start to be greather than min) - check that it is a
+  // sensible difference.
+
+  // Clamp min under max and start under max and above min.
+  if (updated.max_bitrate_bps != -1) {
+    updated.min_bitrate_bps =
+        std::min(updated.min_bitrate_bps, updated.max_bitrate_bps);
+  }
+  updated.start_bitrate_bps =
+      cricket::MinPositive(updated.start_bitrate_bps, updated.max_bitrate_bps);
+  updated.start_bitrate_bps =
+      std::max(updated.start_bitrate_bps, updated.min_bitrate_bps);
+
+  int effective_start =
+      update.start_bitrate_bps ? updated.start_bitrate_bps : -1;
+  LOG(INFO) << "WebRTC.Call.UpdateCurrentBitrateConfig: "
+            << "calling SetBweBitrates with args (" << updated.min_bitrate_bps
+            << ", " << effective_start << ", " << updated.max_bitrate_bps
+            << ")";
   transport_send_->send_side_cc()->SetBweBitrates(
-      bitrate_config.min_bitrate_bps, bitrate_config.start_bitrate_bps,
-      bitrate_config.max_bitrate_bps);
+      updated.min_bitrate_bps, effective_start, updated.max_bitrate_bps);
+  config_.bitrate_config = updated;
 }
 
 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   switch (media) {
     case MediaType::AUDIO:
       audio_network_state_ = state;
       break;
     case MediaType::VIDEO:
       video_network_state_ = state;
@@ skipping 390 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc