GYP->GN of audio_device_tests. Remove dead code. Fix new warnings.

webrtc/modules/audio_device/audio_transport.cc

Index: webrtc/modules/audio_device/audio_transport.cc
diff --git a/webrtc/modules/audio_device/audio_transport.cc b/webrtc/modules/audio_device/audio_transport.cc
new file mode 100644
index 0000000000000000000000000000000000000000..e1733a821f97e2110d45c14e3d91e5ca55e2d733
--- /dev/null
+++ b/webrtc/modules/audio_device/audio_transport.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#include "webrtc/modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+// Some AudioTransport implementations do not care about these functions, so
+// we provide default implementations.
+void AudioTransport::PushCaptureData(int voe_channel,
+                                     const void* audio_data,
+                                     int bits_per_sample,
+                                     int sample_rate,
+                                     size_t number_of_channels,
+                                     size_t number_of_frames) {}
+
+void AudioTransport::PullRenderData(int bits_per_sample,
+                                    int sample_rate,
+                                    size_t number_of_channels,
+                                    size_t number_of_frames,
+                                    void* audio_data,
+                                    int64_t* elapsed_time_ms,
+                                    int64_t* ntp_time_ms) {}
+}  // namespace webrtc