[Cast] Repurpose CastInitializationStatus for variable frame size support.

media/cast/sender/audio_sender_unittest.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 <stdint.h>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/base/media.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/net/cast_transport_config.h"
 #include "media/cast/net/cast_transport_sender_impl.h"
 #include "media/cast/sender/audio_sender.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/utility/audio_utility.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
 
+namespace {
+
+void SaveOperationalStatus(OperationalStatus* out_status,
+                           OperationalStatus in_status) {
+  DVLOG(1) << "OperationalStatus transitioning from " << *out_status << " to "
+           << in_status;
+  *out_status = in_status;
+}
+
+}  // namespace
+
 class TestPacketSender : public PacketSender {
  public:
   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
 
   bool SendPacket(PacketRef packet, const base::Closure& cb) override {
     if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
       ++number_of_rtcp_packets_;
     } else {
       // Check that at least one RTCP packet was sent before the first RTP
       // packet.  This confirms that the receiver will have the necessary lip
@@ skipping 45 matching lines @@
         testing_clock_,
         net::IPEndPoint(),
         dummy_endpoint,
         make_scoped_ptr(new base::DictionaryValue),
         base::Bind(&UpdateCastTransportStatus),
         BulkRawEventsCallback(),
         base::TimeDelta(),
         task_runner_,
         PacketReceiverCallback(),
         &transport_));
+    OperationalStatus operational_status = STATUS_UNINITIALIZED;
     audio_sender_.reset(new AudioSender(
-        cast_environment_, audio_config_, transport_sender_.get()));
+        cast_environment_,
+        audio_config_,
+        base::Bind(&SaveOperationalStatus, &operational_status),
+        transport_sender_.get()));
     task_runner_->RunTasks();
+    CHECK_EQ(STATUS_INITIALIZED, operational_status);
   }
 
   ~AudioSenderTest() override {}
 
   static void UpdateCastTransportStatus(CastTransportStatus status) {
     EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
   }
 
   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
   TestPacketSender transport_;
@@ skipping 33 matching lines @@
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2);
   testing_clock_->Advance(max_rtcp_timeout);
   task_runner_->RunTasks();
   EXPECT_LE(1, transport_.number_of_rtp_packets());
   EXPECT_LE(1, transport_.number_of_rtcp_packets());
 }
 
 }  // namespace cast
 }  // namespace media