4332d09028
This is a reland of 68007e97ec9399125e4be9964af8b0338766cd91 Original change's description: > Reland "Remove WEBRTC_TRACE." > > This is a reland of 2209b90449473e1df3e0797b6271c7624b41907d > 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} > > Bug: webrtc:5118 > Change-Id: I2d93fd40fcaa251c363bdcfb1c04b834a3a7f0e9 > Reviewed-on: https://webrtc-review.googlesource.com/6000 > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20132} Bug: webrtc:5118 Change-Id: I3b46406899d043c3260fc3195b524138324f7313 Reviewed-on: https://webrtc-review.googlesource.com/6301 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20144}
82 lines
2.5 KiB
C++
82 lines
2.5 KiB
C++
/*
|
|
* Copyright (c) 2012 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.
|
|
*/
|
|
|
|
#if defined(WEBRTC_ANDROID)
|
|
#include "modules/audio_device/android/audio_device_template.h"
|
|
#include "modules/audio_device/android/audio_record_jni.h"
|
|
#include "modules/audio_device/android/audio_track_jni.h"
|
|
#endif
|
|
|
|
#include "modules/audio_coding/include/audio_coding_module.h"
|
|
#include "rtc_base/checks.h"
|
|
#include "voice_engine/channel_proxy.h"
|
|
#include "voice_engine/voice_engine_impl.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Counter to be ensure that we can add a correct ID in all static trace
|
|
// methods. It is not the nicest solution, especially not since we already
|
|
// have a counter in VoEBaseImpl. In other words, there is room for
|
|
// improvement here.
|
|
static int32_t gVoiceEngineInstanceCounter = 0;
|
|
|
|
VoiceEngine* GetVoiceEngine() {
|
|
VoiceEngineImpl* self = new VoiceEngineImpl();
|
|
if (self != NULL) {
|
|
self->AddRef(); // First reference. Released in VoiceEngine::Delete.
|
|
gVoiceEngineInstanceCounter++;
|
|
}
|
|
return self;
|
|
}
|
|
|
|
int VoiceEngineImpl::AddRef() {
|
|
return ++_ref_count;
|
|
}
|
|
|
|
// This implements the Release() method for all the inherited interfaces.
|
|
int VoiceEngineImpl::Release() {
|
|
int new_ref = --_ref_count;
|
|
assert(new_ref >= 0);
|
|
if (new_ref == 0) {
|
|
// Clear any pointers before starting destruction. Otherwise worker-
|
|
// threads will still have pointers to a partially destructed object.
|
|
// Example: AudioDeviceBuffer::RequestPlayoutData() can access a
|
|
// partially deconstructed |_ptrCbAudioTransport| during destruction
|
|
// if we don't call Terminate here.
|
|
Terminate();
|
|
delete this;
|
|
}
|
|
|
|
return new_ref;
|
|
}
|
|
|
|
std::unique_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
|
|
int channel_id) {
|
|
RTC_DCHECK(channel_id >= 0);
|
|
rtc::CritScope cs(crit_sec());
|
|
return std::unique_ptr<voe::ChannelProxy>(
|
|
new voe::ChannelProxy(channel_manager().GetChannel(channel_id)));
|
|
}
|
|
|
|
VoiceEngine* VoiceEngine::Create() {
|
|
return GetVoiceEngine();
|
|
}
|
|
|
|
bool VoiceEngine::Delete(VoiceEngine*& voiceEngine) {
|
|
if (voiceEngine == NULL)
|
|
return false;
|
|
|
|
VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
|
|
s->Release();
|
|
voiceEngine = NULL;
|
|
return true;
|
|
}
|
|
} // namespace webrtc
|