[Cast] Cleanup: Remove TransportXXXXXSender, an unnecessary layer of indirection.

media/cast/transport/cast_transport_sender_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/transport/cast_transport_sender_impl.h"
 
 #include "base/single_thread_task_runner.h"
 #include "media/cast/transport/cast_transport_config.h"
 #include "media/cast/transport/cast_transport_defines.h"
 
@@ skipping 38 matching lines @@
                                                        net::IPEndPoint(),
                                                        remote_end_point,
                                                        status_callback)),
       logging_(),
       pacer_(clock,
              &logging_,
              external_transport ? external_transport : transport_.get(),
              transport_task_runner),
       rtcp_builder_(&pacer_),
       raw_events_callback_(raw_events_callback) {
+  DCHECK(clock_);
   if (!raw_events_callback_.is_null()) {
     DCHECK(raw_events_callback_interval > base::TimeDelta());
     event_subscriber_.reset(new SimpleEventSubscriber);
     logging_.AddRawEventSubscriber(event_subscriber_.get());
     raw_events_timer_.Start(FROM_HERE,
                             raw_events_callback_interval,
                             this,
                             &CastTransportSenderImpl::SendRawEvents);
   }
 }
 
 CastTransportSenderImpl::~CastTransportSenderImpl() {
   if (event_subscriber_.get())
     logging_.RemoveRawEventSubscriber(event_subscriber_.get());
 }
 
 void CastTransportSenderImpl::InitializeAudio(
     const CastTransportAudioConfig& config) {
-  pacer_.RegisterAudioSsrc(config.rtp.config.ssrc);
-  audio_sender_.reset(new TransportAudioSender(
-      config, clock_, transport_task_runner_, &pacer_));
-  if (audio_sender_->initialized())
+  LOG_IF(WARNING, config.rtp.config.aes_key.empty() ||
+                  config.rtp.config.aes_iv_mask.empty())
+      << "Unsafe to send audio with encryption DISABLED.";
+  if (!audio_encryptor_.Initialize(config.rtp.config.aes_key,
+                                   config.rtp.config.aes_iv_mask)) {
+    status_callback_.Run(TRANSPORT_AUDIO_UNINITIALIZED);
+    return;
+  }
+  audio_sender_.reset(new RtpSender(clock_, transport_task_runner_, &pacer_));
+  if (audio_sender_->InitializeAudio(config)) {
+    pacer_.RegisterAudioSsrc(config.rtp.config.ssrc);
     status_callback_.Run(TRANSPORT_AUDIO_INITIALIZED);
-  else
+  } else {
+    audio_sender_.reset();
     status_callback_.Run(TRANSPORT_AUDIO_UNINITIALIZED);
+  }
 }
 
 void CastTransportSenderImpl::InitializeVideo(
     const CastTransportVideoConfig& config) {
-  pacer_.RegisterVideoSsrc(config.rtp.config.ssrc);
-  video_sender_.reset(new TransportVideoSender(
-      config, clock_, transport_task_runner_, &pacer_));
-  if (video_sender_->initialized())
+  LOG_IF(WARNING, config.rtp.config.aes_key.empty() ||
+                  config.rtp.config.aes_iv_mask.empty())
+      << "Unsafe to send video with encryption DISABLED.";
+  if (!video_encryptor_.Initialize(config.rtp.config.aes_key,
+                                   config.rtp.config.aes_iv_mask)) {
+    status_callback_.Run(TRANSPORT_VIDEO_UNINITIALIZED);
+    return;
+  }
+  video_sender_.reset(new RtpSender(clock_, transport_task_runner_, &pacer_));
+  if (video_sender_->InitializeVideo(config)) {
+    pacer_.RegisterVideoSsrc(config.rtp.config.ssrc);
     status_callback_.Run(TRANSPORT_VIDEO_INITIALIZED);
-  else
+  } else {
+    video_sender_.reset();
     status_callback_.Run(TRANSPORT_VIDEO_UNINITIALIZED);
+  }
 }
 
 void CastTransportSenderImpl::SetPacketReceiver(
     const PacketReceiverCallback& packet_receiver) {
   transport_->StartReceiving(packet_receiver);
 }
 
+namespace {
+void EncryptAndSendFrame(const EncodedFrame& frame,
+                         TransportEncryptionHandler* encryptor,
+                         RtpSender* sender) {
+  if (encryptor->initialized()) {
+    EncodedFrame encrypted_frame;
+    frame.CopyMetadataTo(&encrypted_frame);
+    if (encryptor->Encrypt(frame.frame_id, frame.data, &encrypted_frame.data)) {
+      sender->SendFrame(encrypted_frame);
+    } else {
+      LOG(ERROR) << "Encryption failed.  Not sending frame with ID "
+                 << frame.frame_id;
+    }
+  } else {
+    sender->SendFrame(frame);
+  }
+}
+}  // namespace
+
 void CastTransportSenderImpl::InsertCodedAudioFrame(
     const EncodedFrame& audio_frame) {
   DCHECK(audio_sender_) << "Audio sender uninitialized";
-  audio_sender_->SendFrame(audio_frame);
+  EncryptAndSendFrame(audio_frame, &audio_encryptor_, audio_sender_.get());
 }
 
 void CastTransportSenderImpl::InsertCodedVideoFrame(
     const EncodedFrame& video_frame) {
   DCHECK(video_sender_) << "Video sender uninitialized";
-  video_sender_->SendFrame(video_frame);
+  EncryptAndSendFrame(video_frame, &video_encryptor_, video_sender_.get());
 }
 
 void CastTransportSenderImpl::SendRtcpFromRtpSender(
     uint32 packet_type_flags,
     uint32 ntp_seconds,
     uint32 ntp_fraction,
     uint32 rtp_timestamp,
     const RtcpDlrrReportBlock& dlrr,
     uint32 sending_ssrc,
     const std::string& c_name) {
@@ skipping 31 matching lines @@
   DCHECK(event_subscriber_.get());
   DCHECK(!raw_events_callback_.is_null());
   std::vector<PacketEvent> packet_events;
   event_subscriber_->GetPacketEventsAndReset(&packet_events);
   raw_events_callback_.Run(packet_events);
 }
 
 }  // namespace transport
 }  // namespace cast
 }  // namespace media