Added an empty AudioTransportProxy to AudioState.

webrtc/audio/audio_transport_proxy.cc

Index: webrtc/audio/audio_transport_proxy.cc
diff --git a/webrtc/audio/audio_transport_proxy.cc b/webrtc/audio/audio_transport_proxy.cc
new file mode 100644
index 0000000000000000000000000000000000000000..f5655f68887bdfa0ae54915185af0a82d2117a99
--- /dev/null
+++ b/webrtc/audio/audio_transport_proxy.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2016 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/audio/audio_transport_proxy.h"
+
+namespace webrtc {
+
+int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples,
+                                              const size_t nBytesPerSample,
+                                              const size_t nChannels,
+                                              const uint32_t samplesPerSec,
+                                              void* audioSamples,
+                                              size_t& nSamplesOut,
+                                              int64_t* elapsed_time_ms,
+                                              int64_t* ntp_time_ms) {
+  RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
+  RTC_DCHECK_GE(nChannels, 1u);
+  RTC_DCHECK_LE(nChannels, 2u);
+  RTC_DCHECK_GE(
+      samplesPerSec,
+      static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
+  RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
+  RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
+                sizeof(AudioFrame::data_));
+
+  // Pass call through to original audio transport instance.
+  return voe_audio_transport_->NeedMorePlayData(
+      nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples,
+      nSamplesOut, elapsed_time_ms, ntp_time_ms);
+}
+
+void AudioTransportProxy::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) {
+  RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 8 * sizeof(int16_t));
+  RTC_DCHECK_GE(number_of_channels, 1u);
+  RTC_DCHECK_LE(number_of_channels, 2u);
+  RTC_DCHECK_GE(static_cast<int>(sample_rate),
+                AudioProcessing::NativeRate::kSampleRate8kHz);
+  RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate);
+  RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
+                sizeof(AudioFrame::data_));
+  voe_audio_transport_->PullRenderData(
+      bits_per_sample, sample_rate, number_of_channels, number_of_frames,
+      audio_data, elapsed_time_ms, ntp_time_ms);
+}
+
+}  // namespace webrtc