ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
85 lines
2.9 KiB
C++
85 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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#ifndef WEBRTC_VOICE_ENGINE_SHARED_DATA_H
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#define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/utility/include/process_thread.h"
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#include "webrtc/voice_engine/channel_manager.h"
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#include "webrtc/voice_engine/statistics.h"
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#include "webrtc/voice_engine/voice_engine_defines.h"
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class ProcessThread;
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namespace webrtc {
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class Config;
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class CriticalSectionWrapper;
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namespace voe {
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class TransmitMixer;
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class OutputMixer;
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class SharedData
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{
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public:
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// Public accessors.
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uint32_t instance_id() const { return _instanceId; }
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Statistics& statistics() { return _engineStatistics; }
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ChannelManager& channel_manager() { return _channelManager; }
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AudioDeviceModule* audio_device() { return _audioDevicePtr; }
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void set_audio_device(AudioDeviceModule* audio_device);
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AudioProcessing* audio_processing() { return audioproc_.get(); }
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void set_audio_processing(AudioProcessing* audio_processing);
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TransmitMixer* transmit_mixer() { return _transmitMixerPtr; }
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OutputMixer* output_mixer() { return _outputMixerPtr; }
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CriticalSectionWrapper* crit_sec() { return _apiCritPtr; }
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ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); }
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AudioDeviceModule::AudioLayer audio_device_layer() const {
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return _audioDeviceLayer;
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}
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void set_audio_device_layer(AudioDeviceModule::AudioLayer layer) {
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_audioDeviceLayer = layer;
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}
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int NumOfSendingChannels();
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int NumOfPlayingChannels();
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// Convenience methods for calling statistics().SetLastError().
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void SetLastError(int32_t error) const;
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void SetLastError(int32_t error, TraceLevel level) const;
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void SetLastError(int32_t error, TraceLevel level,
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const char* msg) const;
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protected:
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const uint32_t _instanceId;
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CriticalSectionWrapper* _apiCritPtr;
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ChannelManager _channelManager;
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Statistics _engineStatistics;
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AudioDeviceModule* _audioDevicePtr;
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OutputMixer* _outputMixerPtr;
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TransmitMixer* _transmitMixerPtr;
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rtc::scoped_ptr<AudioProcessing> audioproc_;
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rtc::scoped_ptr<ProcessThread> _moduleProcessThreadPtr;
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AudioDeviceModule::AudioLayer _audioDeviceLayer;
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SharedData(const Config& config);
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virtual ~SharedData();
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};
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} // namespace voe
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_SHARED_DATA_H
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