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 |