From 7fbfaa49d2de75d67e729c96f9ee503778f50141 Mon Sep 17 00:00:00 2001
From: "Piotr (Peter) Slatala" <psla@webrtc.org>
Date: Mon, 18 Mar 2019 10:31:54 -0700
Subject: [PATCH] PeerConnection::SetBitrate now also configures media
 transport.

(so far SetBitrate did not do anything for media transport)

Bug: webrtc:9719
Change-Id: I48e669341ffe6c9e4697ff9146c314be7796a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27169}
---
 call/call.cc                    | 36 +++++++++++++++++++++++++++++++++
 call/call.h                     |  3 +++
 call/degraded_call.h            |  2 ++
 media/engine/fake_webrtc_call.h |  3 +++
 pc/peer_connection.cc           |  2 +-
 5 files changed, 45 insertions(+), 1 deletion(-)

diff --git a/call/call.cc b/call/call.cc
index bf42ee3e6f..8dcd785b8c 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -244,6 +244,8 @@ class Call final : public webrtc::Call,
   // at least once after that.
   void MediaTransportChange(MediaTransportInterface* media_transport) override;
 
+  void SetClientBitratePreferences(const BitrateSettings& preferences) override;
+
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type,
                              const uint8_t* packet,
@@ -275,6 +277,9 @@ class Call final : public webrtc::Call,
   Clock* const clock_;
   TaskQueueFactory* const task_queue_factory_;
 
+  // Caching the last SetBitrate for media transport.
+  absl::optional<MediaTransportTargetRateConstraints> last_set_bitrate_
+      RTC_GUARDED_BY(&target_observer_crit_);
   const int num_cpu_cores_;
   const std::unique_ptr<ProcessThread> module_process_thread_;
   const std::unique_ptr<CallStats> call_stats_;
@@ -580,6 +585,37 @@ void Call::MediaTransportChange(MediaTransportInterface* media_transport) {
       constraints.min_bitrate =
           DataRate::bps(config_.bitrate_config.min_bitrate_bps);
     }
+
+    // User called ::SetBitrate on peer connection before
+    // media transport was created.
+    if (last_set_bitrate_) {
+      media_transport_->SetTargetBitrateLimits(*last_set_bitrate_);
+    } else {
+      media_transport_->SetTargetBitrateLimits(constraints);
+    }
+  }
+}
+
+void Call::SetClientBitratePreferences(const BitrateSettings& preferences) {
+  GetTransportControllerSend()->SetClientBitratePreferences(preferences);
+  // Can the client code invoke 'SetBitrate' before media transport is created?
+  // It's probably possible :/
+  MediaTransportTargetRateConstraints constraints;
+  if (preferences.start_bitrate_bps.has_value()) {
+    constraints.starting_bitrate =
+        webrtc::DataRate::bps(*preferences.start_bitrate_bps);
+  }
+  if (preferences.max_bitrate_bps.has_value()) {
+    constraints.max_bitrate =
+        webrtc::DataRate::bps(*preferences.max_bitrate_bps);
+  }
+  if (preferences.min_bitrate_bps.has_value()) {
+    constraints.min_bitrate =
+        webrtc::DataRate::bps(*preferences.min_bitrate_bps);
+  }
+  rtc::CritScope lock(&target_observer_crit_);
+  last_set_bitrate_ = constraints;
+  if (media_transport_) {
     media_transport_->SetTargetBitrateLimits(constraints);
   }
 }
diff --git a/call/call.h b/call/call.h
index 3eceb492a8..90977dab2c 100644
--- a/call/call.h
+++ b/call/call.h
@@ -124,6 +124,9 @@ class Call {
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
+  virtual void SetClientBitratePreferences(
+      const BitrateSettings& preferences) = 0;
+
   virtual ~Call() {}
 };
 
diff --git a/call/degraded_call.h b/call/degraded_call.h
index 89eafdb9dc..86b1c9a9b7 100644
--- a/call/degraded_call.h
+++ b/call/degraded_call.h
@@ -133,6 +133,8 @@ class DegradedCall : public Call, private Transport, private PacketReceiver {
   const std::unique_ptr<Call> call_;
 
   void MediaTransportChange(MediaTransportInterface* media_transport) override;
+  void SetClientBitratePreferences(
+      const webrtc::BitrateSettings& preferences) override {}
   const absl::optional<BuiltInNetworkBehaviorConfig> send_config_;
   const std::unique_ptr<ProcessThread> send_process_thread_;
   SimulatedNetwork* send_simulated_network_;
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 71a54d66bf..b20b7ed771 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -303,6 +303,9 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
   void MediaTransportChange(
       webrtc::MediaTransportInterface* media_transport_interface) override;
 
+  void SetClientBitratePreferences(
+      const webrtc::BitrateSettings& preferences) override {}
+
  private:
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index bcba7636af..15bea89869 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -3611,7 +3611,7 @@ RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
   }
 
   RTC_DCHECK(call_.get());
-  call_->GetTransportControllerSend()->SetClientBitratePreferences(bitrate);
+  call_->SetClientBitratePreferences(bitrate);
 
   return RTCError::OK();
 }