Remove old methods in AudioTransport, make it pass a gn build

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) {}

[the sun, 2016/08/05 18:14:31]

I think this implementation file is unnecessary. Either just leave them as they
were or, ideally, make the methods pure virtual and force each implementor to
include an empty implementation. Note that the latter is more work as clients
need to be updated before submitting this CL.

[Max Morin WebRTC, 2016/08/08 15:28:57]

Ok, they are pure virtual now.

+
+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