729b9109ca
This reverts commit 2209b90449473e1df3e0797b6271c7624b41907d. Reason for revert: breaks Chromium Original change's description: > Remove WEBRTC_TRACE. > > Bug: webrtc:5118 > Change-Id: I0af0f8845ee016fa61d7cecc526e2a672ec8732d > Reviewed-on: https://webrtc-review.googlesource.com/5382 > Reviewed-by: Niels Moller <nisse@webrtc.org> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20114} TBR=solenberg@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org Change-Id: Ie54fc05c1d7895c088cba410ed87a7c9a0701427 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:5118 Reviewed-on: https://webrtc-review.googlesource.com/5980 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20115}
96 lines
2.9 KiB
C++
96 lines
2.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#if defined(WEBRTC_ANDROID)
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#include "modules/audio_device/android/audio_device_template.h"
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#include "modules/audio_device/android/audio_record_jni.h"
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#include "modules/audio_device/android/audio_track_jni.h"
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#endif
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "rtc_base/checks.h"
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#include "system_wrappers/include/trace.h"
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#include "voice_engine/channel_proxy.h"
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#include "voice_engine/voice_engine_impl.h"
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namespace webrtc {
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// Counter to be ensure that we can add a correct ID in all static trace
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// methods. It is not the nicest solution, especially not since we already
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// have a counter in VoEBaseImpl. In other words, there is room for
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// improvement here.
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static int32_t gVoiceEngineInstanceCounter = 0;
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VoiceEngine* GetVoiceEngine() {
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VoiceEngineImpl* self = new VoiceEngineImpl();
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if (self != NULL) {
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self->AddRef(); // First reference. Released in VoiceEngine::Delete.
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gVoiceEngineInstanceCounter++;
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}
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return self;
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}
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int VoiceEngineImpl::AddRef() {
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return ++_ref_count;
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}
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// This implements the Release() method for all the inherited interfaces.
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int VoiceEngineImpl::Release() {
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int new_ref = --_ref_count;
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assert(new_ref >= 0);
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if (new_ref == 0) {
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WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
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"VoiceEngineImpl self deleting (voiceEngine=0x%p)", this);
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// Clear any pointers before starting destruction. Otherwise worker-
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// threads will still have pointers to a partially destructed object.
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// Example: AudioDeviceBuffer::RequestPlayoutData() can access a
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// partially deconstructed |_ptrCbAudioTransport| during destruction
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// if we don't call Terminate here.
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Terminate();
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delete this;
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}
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return new_ref;
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}
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std::unique_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
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int channel_id) {
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RTC_DCHECK(channel_id >= 0);
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rtc::CritScope cs(crit_sec());
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return std::unique_ptr<voe::ChannelProxy>(
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new voe::ChannelProxy(channel_manager().GetChannel(channel_id)));
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}
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VoiceEngine* VoiceEngine::Create() {
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return GetVoiceEngine();
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}
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bool VoiceEngine::Delete(VoiceEngine*& voiceEngine) {
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if (voiceEngine == NULL)
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return false;
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VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
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// Release the reference that was added in GetVoiceEngine.
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int ref = s->Release();
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voiceEngine = NULL;
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if (ref != 0) {
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WEBRTC_TRACE(
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kTraceWarning, kTraceVoice, -1,
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"VoiceEngine::Delete did not release the very last reference. "
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"%d references remain.",
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ref);
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}
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return true;
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}
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} // namespace webrtc
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