191 lines
7.2 KiB
C
191 lines
7.2 KiB
C
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/*
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* libjingle
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* Copyright 2004--2011, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_SESSION_PHONE_WEBRTCVOE_H_
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#define TALK_SESSION_PHONE_WEBRTCVOE_H_
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#include "talk/base/common.h"
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#include "talk/session/phone/webrtccommon.h"
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#ifdef WEBRTC_RELATIVE_PATH
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#include "common_types.h"
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#include "modules/audio_device/main/interface/audio_device.h"
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#include "voice_engine/main/interface/voe_audio_processing.h"
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#include "voice_engine/main/interface/voe_base.h"
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#include "voice_engine/main/interface/voe_codec.h"
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#include "voice_engine/main/interface/voe_dtmf.h"
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#include "voice_engine/main/interface/voe_errors.h"
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#include "voice_engine/main/interface/voe_file.h"
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#include "voice_engine/main/interface/voe_hardware.h"
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#include "voice_engine/main/interface/voe_neteq_stats.h"
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#include "voice_engine/main/interface/voe_network.h"
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#include "voice_engine/main/interface/voe_rtp_rtcp.h"
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#include "voice_engine/main/interface/voe_video_sync.h"
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#include "voice_engine/main/interface/voe_volume_control.h"
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#else
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#include "third_party/webrtc/files/include/audio_device.h"
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#include "third_party/webrtc/files/include/common_types.h"
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#include "third_party/webrtc/files/include/voe_audio_processing.h"
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#include "third_party/webrtc/files/include/voe_base.h"
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#include "third_party/webrtc/files/include/voe_codec.h"
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#include "third_party/webrtc/files/include/voe_dtmf.h"
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#include "third_party/webrtc/files/include/voe_errors.h"
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#include "third_party/webrtc/files/include/voe_file.h"
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#include "third_party/webrtc/files/include/voe_hardware.h"
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#include "third_party/webrtc/files/include/voe_neteq_stats.h"
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#include "third_party/webrtc/files/include/voe_network.h"
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#include "third_party/webrtc/files/include/voe_rtp_rtcp.h"
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#include "third_party/webrtc/files/include/voe_video_sync.h"
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#include "third_party/webrtc/files/include/voe_volume_control.h"
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#endif // WEBRTC_RELATIVE_PATH
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namespace cricket {
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// automatically handles lifetime of WebRtc VoiceEngine
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class scoped_voe_engine {
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public:
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explicit scoped_voe_engine(webrtc::VoiceEngine* e) : ptr(e) {}
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// VERIFY, to ensure that there are no leaks at shutdown
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~scoped_voe_engine() { if (ptr) VERIFY(webrtc::VoiceEngine::Delete(ptr)); }
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// Releases the current pointer.
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void reset() {
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if (ptr) {
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VERIFY(webrtc::VoiceEngine::Delete(ptr));
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ptr = NULL;
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}
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}
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webrtc::VoiceEngine* get() const { return ptr; }
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private:
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webrtc::VoiceEngine* ptr;
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};
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// scoped_ptr class to handle obtaining and releasing WebRTC interface pointers
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template<class T>
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class scoped_voe_ptr {
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public:
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explicit scoped_voe_ptr(const scoped_voe_engine& e)
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: ptr(T::GetInterface(e.get())) {}
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explicit scoped_voe_ptr(T* p) : ptr(p) {}
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~scoped_voe_ptr() { if (ptr) ptr->Release(); }
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T* operator->() const { return ptr; }
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T* get() const { return ptr; }
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// Releases the current pointer.
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void reset() {
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if (ptr) {
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ptr->Release();
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ptr = NULL;
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}
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}
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private:
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T* ptr;
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};
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// Utility class for aggregating the various WebRTC interface.
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// Fake implementations can also be injected for testing.
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class VoEWrapper {
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public:
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VoEWrapper()
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: engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
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base_(engine_), codec_(engine_), dtmf_(engine_), file_(engine_),
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hw_(engine_), neteq_(engine_), network_(engine_), rtp_(engine_),
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sync_(engine_), volume_(engine_) {
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}
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VoEWrapper(webrtc::VoEAudioProcessing* processing,
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webrtc::VoEBase* base,
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webrtc::VoECodec* codec,
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webrtc::VoEDtmf* dtmf,
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webrtc::VoEFile* file,
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webrtc::VoEHardware* hw,
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webrtc::VoENetEqStats* neteq,
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webrtc::VoENetwork* network,
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webrtc::VoERTP_RTCP* rtp,
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webrtc::VoEVideoSync* sync,
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webrtc::VoEVolumeControl* volume)
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: engine_(NULL),
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processing_(processing),
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base_(base),
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codec_(codec),
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dtmf_(dtmf),
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file_(file),
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hw_(hw),
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neteq_(neteq),
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network_(network),
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rtp_(rtp),
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sync_(sync),
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volume_(volume) {
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}
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~VoEWrapper() {}
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webrtc::VoiceEngine* engine() const { return engine_.get(); }
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webrtc::VoEAudioProcessing* processing() const { return processing_.get(); }
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webrtc::VoEBase* base() const { return base_.get(); }
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webrtc::VoECodec* codec() const { return codec_.get(); }
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webrtc::VoEDtmf* dtmf() const { return dtmf_.get(); }
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webrtc::VoEFile* file() const { return file_.get(); }
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webrtc::VoEHardware* hw() const { return hw_.get(); }
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webrtc::VoENetEqStats* neteq() const { return neteq_.get(); }
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webrtc::VoENetwork* network() const { return network_.get(); }
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webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
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webrtc::VoEVideoSync* sync() const { return sync_.get(); }
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webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
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int error() { return base_->LastError(); }
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private:
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scoped_voe_engine engine_;
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scoped_voe_ptr<webrtc::VoEAudioProcessing> processing_;
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scoped_voe_ptr<webrtc::VoEBase> base_;
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scoped_voe_ptr<webrtc::VoECodec> codec_;
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scoped_voe_ptr<webrtc::VoEDtmf> dtmf_;
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scoped_voe_ptr<webrtc::VoEFile> file_;
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scoped_voe_ptr<webrtc::VoEHardware> hw_;
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scoped_voe_ptr<webrtc::VoENetEqStats> neteq_;
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scoped_voe_ptr<webrtc::VoENetwork> network_;
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scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
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scoped_voe_ptr<webrtc::VoEVideoSync> sync_;
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scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
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};
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// Adds indirection to static WebRtc functions, allowing them to be mocked.
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class VoETraceWrapper {
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public:
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virtual ~VoETraceWrapper() {}
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virtual int SetTraceFilter(const unsigned int filter) {
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return webrtc::VoiceEngine::SetTraceFilter(filter);
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}
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virtual int SetTraceFile(const char* fileNameUTF8) {
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return webrtc::VoiceEngine::SetTraceFile(fileNameUTF8);
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}
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virtual int SetTraceCallback(webrtc::TraceCallback* callback) {
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return webrtc::VoiceEngine::SetTraceCallback(callback);
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}
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};
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}
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#endif // TALK_SESSION_PHONE_WEBRTCVOE_H_
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