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

webrtc/audio/audio_state.cc

Index: webrtc/audio/audio_state.cc
diff --git a/webrtc/audio/audio_state.cc b/webrtc/audio/audio_state.cc
new file mode 100644
index 0000000000000000000000000000000000000000..e63f97af2d4fb9fd3123f37532f0553382715984
--- /dev/null
+++ b/webrtc/audio/audio_state.cc
@@ -0,0 +1,79 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/audio/audio_state.h"
+
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/voice_engine/include/voe_errors.h"
+
+namespace webrtc {
+namespace internal {
+
+AudioState::AudioState(const AudioState::Config& config)
+    : config_(config), voe_base_(config.voice_engine) {
+  process_thread_checker_.DetachFromThread();
+  // Only one AudioState should be created per VoiceEngine.
+  RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
+}
+
+AudioState::~AudioState() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  voe_base_->DeRegisterVoiceEngineObserver();
+}
+
+VoiceEngine* AudioState::voice_engine() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return config_.voice_engine;
+}
+
+bool AudioState::typing_noise_detected() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  rtc::CritScope lock(&crit_sect_);
+  return typing_noise_detected_;
+}
+
+// Reference count; implementation copied from rtc::RefCountedObject.
+int AudioState::AddRef() const {
+  return rtc::AtomicOps::Increment(&ref_count_);
+}
+
+// Reference count; implementation copied from rtc::RefCountedObject.
+int AudioState::Release() const {
+  int count = rtc::AtomicOps::Decrement(&ref_count_);
+  if (!count) {
+    delete this;
+  }
+  return count;
+}
+
+void AudioState::CallbackOnError(int channel_id, int err_code) {
+  RTC_DCHECK(process_thread_checker_.CalledOnValidThread());
+
+  // All call sites in VoE, as of this writing, specify -1 as channel_id.
+  RTC_DCHECK(channel_id == -1);
+  LOG(LS_INFO) << "VoiceEngine error " << err_code << " reported on channel "
+               << channel_id << ".";
+  if (err_code == VE_TYPING_NOISE_WARNING) {
+    rtc::CritScope lock(&crit_sect_);
+    typing_noise_detected_ = true;
+  } else if (err_code == VE_TYPING_NOISE_OFF_WARNING) {
+    rtc::CritScope lock(&crit_sect_);
+    typing_noise_detected_ = false;
+  }
+}
+}  // namespace internal
+
+rtc::scoped_refptr<AudioState> AudioState::Create(
+    const AudioState::Config& config) {
+  return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
+}
+}  // namespace webrtc