Added an empty AudioTransportProxy to AudioState.

webrtc/audio/audio_transport_proxy.cc

+/*
+ *  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 {
+
+AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport,
+                                         AudioProcessing* apm,
+                                         AudioMixer* mixer)
+    : voe_audio_transport_(voe_audio_transport) {
+  RTC_DCHECK(voe_audio_transport);
+  RTC_DCHECK(apm);
+}
+
+AudioTransportProxy::~AudioTransportProxy() {}
+
+int32_t AudioTransportProxy::RecordedDataIsAvailable(
+    const void* audioSamples,
+    const size_t nSamples,
+    const size_t nBytesPerSample,
+    const size_t nChannels,
+    const uint32_t samplesPerSec,
+    const uint32_t totalDelayMS,
+    const int32_t clockDrift,
+    const uint32_t currentMicLevel,
+    const bool keyPressed,
+    uint32_t& newMicLevel) {
+  // Pass call through to original audio transport instance.
+  return voe_audio_transport_->RecordedDataIsAvailable(
+      audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
+      totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
+}
+
+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::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) {
+  // This is part of deprecated VoE interface operating on specific
+  // VoE channels. It should not be used.
+  RTC_NOTREACHED();
+}
+
+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