Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i…

webrtc/audio/audio_state.h

+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_STATE_H_
+#define WEBRTC_AUDIO_AUDIO_STATE_H_
+
+#include "webrtc/audio_state.h"
+#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+
+namespace webrtc {
+namespace internal {
+
+class AudioState final : public webrtc::AudioState,
+                         public webrtc::VoiceEngineObserver {
+ public:
+  explicit AudioState(const AudioState::Config& config);
+  ~AudioState() override;
+
+  VoiceEngine* voice_engine();
+  bool typing_noise_detected() const;
+
+ private:
+  // webrtc::VoiceEngineObserver implementation.
+  void CallbackOnError(int channel_id, int err_code) override;
+
+  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker process_thread_checker_;
+  const webrtc::AudioState::Config config_;
+
+  // We hold one interface pointer to the VoE to make sure it is kept alive.
+  ScopedVoEInterface<VoEBase> voe_base_;
+
+  // TODO(solenberg): I don't actually think it is worth adding a crit sect in
+  //                  this case, but reviewer may think differently...

[tommi, 2015/10/29 16:28:50]

are we reading/writing from/to this variable from many different threads?  Would
it make sense for it to be atomic?

[the sun, 2015/10/30 12:19:01]

Done.

+  // rtc::CriticalSection crit_sect_;
+  bool typing_noise_detected_ = false;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_STATE_H_