Refactor: simplify interface of SenderRtcpSession and CastTransport.

media/cast/net/cast_transport_impl.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/net/cast_transport_impl.h"
 
 #include <stddef.h>
 #include <algorithm>
 #include <string>
 #include <utility>
@@ skipping 38 matching lines @@
     const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner) {
   return std::unique_ptr<CastTransport>(
       new CastTransportImpl(clock, logging_flush_interval, std::move(client),
                             std::move(transport), transport_task_runner.get()));
 }
 
 PacketReceiverCallback CastTransport::PacketReceiverForTesting() {
   return PacketReceiverCallback();
 }
 
+class CastTransportImpl::RtcpClient : public RtcpObserver {
+ public:
+  RtcpClient(std::unique_ptr<RtcpObserver> observer,
+             uint32_t rtp_sender_ssrc,
+             EventMediaType media_type,
+             CastTransportImpl* cast_transport_impl)
+      : rtp_sender_ssrc_(rtp_sender_ssrc),
+        rtcp_observer_(std::move(observer)),
+        media_type_(media_type),
+        cast_transport_impl_(cast_transport_impl) {}
+
+  void OnReceivedCastMessage(const RtcpCastMessage& cast_message) override {
+    rtcp_observer_->OnReceivedCastMessage(cast_message);
+    cast_transport_impl_->OnReceivedCastMessage(rtp_sender_ssrc_, cast_message);
+  }
+
+  void OnReceivedRtt(base::TimeDelta round_trip_time) override {
+    rtcp_observer_->OnReceivedRtt(round_trip_time);
+  }
+
+  void OnReceivedReceiverLog(const RtcpReceiverLogMessage& log) override {
+    cast_transport_impl_->OnReceivedLogMessage(media_type_, log);
+  }
+
+  void OnReceivedPli() override { rtcp_observer_->OnReceivedPli(); }
+
+ private:
+  const uint32_t rtp_sender_ssrc_;
+  const std::unique_ptr<RtcpObserver> rtcp_observer_;
+  const EventMediaType media_type_;
+  CastTransportImpl* const cast_transport_impl_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtcpClient);
+};
+
 CastTransportImpl::CastTransportImpl(
     base::TickClock* clock,
     base::TimeDelta logging_flush_interval,
     std::unique_ptr<Client> client,
     std::unique_ptr<PacketTransport> transport,
     const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner)
     : clock_(clock),
       logging_flush_interval_(logging_flush_interval),
       transport_client_(std::move(client)),
       transport_(std::move(transport)),
@@ skipping 20 matching lines @@
   transport_->StartReceiving(
       base::Bind(&CastTransportImpl::OnReceivedPacket, base::Unretained(this)));
 }
 
 CastTransportImpl::~CastTransportImpl() {
   transport_->StopReceiving();
 }
 
 void CastTransportImpl::InitializeAudio(
     const CastTransportRtpConfig& config,
-    const RtcpCastMessageCallback& cast_message_cb,
-    const RtcpRttCallback& rtt_cb,
-    const RtcpPliCallback& pli_cb) {
+    std::unique_ptr<RtcpObserver> rtcp_observer) {
   LOG_IF(WARNING, config.aes_key.empty() || config.aes_iv_mask.empty())
       << "Unsafe to send audio with encryption DISABLED.";
   if (!audio_encryptor_.Initialize(config.aes_key, config.aes_iv_mask)) {
     transport_client_->OnStatusChanged(TRANSPORT_AUDIO_UNINITIALIZED);
     return;
   }
 
   audio_sender_.reset(new RtpSender(transport_task_runner_, &pacer_));
   if (audio_sender_->Initialize(config)) {
     // Audio packets have a higher priority.
     pacer_.RegisterAudioSsrc(config.ssrc);
     pacer_.RegisterPrioritySsrc(config.ssrc);
     transport_client_->OnStatusChanged(TRANSPORT_AUDIO_INITIALIZED);
   } else {
     audio_sender_.reset();
     transport_client_->OnStatusChanged(TRANSPORT_AUDIO_UNINITIALIZED);
     return;
   }
 
-  audio_rtcp_session_.reset(new SenderRtcpSession(
-      base::Bind(&CastTransportImpl::OnReceivedCastMessage,
-                 weak_factory_.GetWeakPtr(), config.ssrc, cast_message_cb),
-      rtt_cb, base::Bind(&CastTransportImpl::OnReceivedLogMessage,
-                         weak_factory_.GetWeakPtr(), AUDIO_EVENT),
-      pli_cb, clock_, &pacer_, config.ssrc, config.feedback_ssrc));
+  audio_rtcp_observer_.reset(
+      new RtcpClient(std::move(rtcp_observer), config.ssrc, AUDIO_EVENT, this));
+  audio_rtcp_session_.reset(
+      new SenderRtcpSession(clock_, &pacer_, audio_rtcp_observer_.get(),
+                            config.ssrc, config.feedback_ssrc));
   pacer_.RegisterAudioSsrc(config.ssrc);
   valid_sender_ssrcs_.insert(config.feedback_ssrc);
   transport_client_->OnStatusChanged(TRANSPORT_AUDIO_INITIALIZED);
 }
 
 void CastTransportImpl::InitializeVideo(
     const CastTransportRtpConfig& config,
-    const RtcpCastMessageCallback& cast_message_cb,
-    const RtcpRttCallback& rtt_cb,
-    const RtcpPliCallback& pli_cb) {
+    std::unique_ptr<RtcpObserver> rtcp_observer) {
   LOG_IF(WARNING, config.aes_key.empty() || config.aes_iv_mask.empty())
       << "Unsafe to send video with encryption DISABLED.";
   if (!video_encryptor_.Initialize(config.aes_key, config.aes_iv_mask)) {
     transport_client_->OnStatusChanged(TRANSPORT_VIDEO_UNINITIALIZED);
     return;
   }
 
   video_sender_.reset(new RtpSender(transport_task_runner_, &pacer_));
   if (!video_sender_->Initialize(config)) {
     video_sender_.reset();
     transport_client_->OnStatusChanged(TRANSPORT_VIDEO_UNINITIALIZED);
     return;
   }
 
-  video_rtcp_session_.reset(new SenderRtcpSession(
-      base::Bind(&CastTransportImpl::OnReceivedCastMessage,
-                 weak_factory_.GetWeakPtr(), config.ssrc, cast_message_cb),
-      rtt_cb, base::Bind(&CastTransportImpl::OnReceivedLogMessage,
-                         weak_factory_.GetWeakPtr(), VIDEO_EVENT),
-      pli_cb, clock_, &pacer_, config.ssrc, config.feedback_ssrc));
+  video_rtcp_observer_.reset(
+      new RtcpClient(std::move(rtcp_observer), config.ssrc, VIDEO_EVENT, this));
+  video_rtcp_session_.reset(
+      new SenderRtcpSession(clock_, &pacer_, video_rtcp_observer_.get(),
+                            config.ssrc, config.feedback_ssrc));
   pacer_.RegisterVideoSsrc(config.ssrc);
   valid_sender_ssrcs_.insert(config.feedback_ssrc);
   transport_client_->OnStatusChanged(TRANSPORT_VIDEO_INITIALIZED);
 }
 
 namespace {
 void EncryptAndSendFrame(const EncodedFrame& frame,
                          TransportEncryptionHandler* encryptor,
                          RtpSender* sender) {
   // TODO(miu): We probably shouldn't attempt to send an empty frame, but this
@@ skipping 175 matching lines @@
           VLOG(2) << "Received log message via RTCP that we did not expect: "
                   << event_log_message.type;
           break;
       }
     }
   }
 }
 
 void CastTransportImpl::OnReceivedCastMessage(
     uint32_t ssrc,
-    const RtcpCastMessageCallback& cast_message_cb,
     const RtcpCastMessage& cast_message) {
-  if (!cast_message_cb.is_null())
-    cast_message_cb.Run(cast_message);
 
   DedupInfo dedup_info;
   if (audio_sender_ && audio_sender_->ssrc() == ssrc) {
     const int64_t acked_bytes =
         audio_sender_->GetLastByteSentForFrame(cast_message.ack_frame_id);
     last_byte_acked_for_audio_ =
         std::max(acked_bytes, last_byte_acked_for_audio_);
   } else if (video_sender_ && video_sender_->ssrc() == ssrc) {
     dedup_info.resend_interval = video_rtcp_session_->current_round_trip_time();
 
@@ skipping 117 matching lines @@
                "calling CastTransportImpl::SendRtcpFromRtpReceiver.";
     return;
   }
   pacer_.SendRtcpPacket(rtcp_builder_at_rtp_receiver_->local_ssrc(),
                         rtcp_builder_at_rtp_receiver_->Finish());
   rtcp_builder_at_rtp_receiver_.reset();
 }
 
 }  // namespace cast
 }  // namespace media