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}
135 lines
5.2 KiB
C++
135 lines
5.2 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_VOE_BASE_IMPL_H
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#define WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/voice_engine/shared_data.h"
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namespace webrtc {
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class ProcessThread;
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class VoEBaseImpl : public VoEBase,
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public AudioTransport,
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public AudioDeviceObserver {
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public:
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int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) override;
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int DeRegisterVoiceEngineObserver() override;
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int Init(AudioDeviceModule* external_adm = nullptr,
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AudioProcessing* audioproc = nullptr) override;
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AudioProcessing* audio_processing() override {
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return shared_->audio_processing();
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}
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int Terminate() override;
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int CreateChannel() override;
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int CreateChannel(const Config& config) override;
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int DeleteChannel(int channel) override;
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int StartReceive(int channel) override;
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int StartPlayout(int channel) override;
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int StartSend(int channel) override;
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int StopReceive(int channel) override;
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int StopPlayout(int channel) override;
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int StopSend(int channel) override;
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int GetVersion(char version[1024]) override;
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int LastError() override;
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AudioTransport* audio_transport() override { return this; }
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int AssociateSendChannel(int channel, int accociate_send_channel) override;
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// AudioTransport
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int32_t RecordedDataIsAvailable(const void* audioSamples, size_t nSamples,
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size_t nBytesPerSample, uint8_t nChannels,
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uint32_t samplesPerSec, uint32_t totalDelayMS,
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int32_t clockDrift, uint32_t micLevel,
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bool keyPressed,
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uint32_t& newMicLevel) override;
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int32_t NeedMorePlayData(size_t nSamples, size_t nBytesPerSample,
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uint8_t nChannels, uint32_t samplesPerSec,
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void* audioSamples, size_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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int OnDataAvailable(const int voe_channels[], int number_of_voe_channels,
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const int16_t* audio_data, int sample_rate,
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int number_of_channels, size_t number_of_frames,
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int audio_delay_milliseconds, int volume,
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bool key_pressed, bool need_audio_processing) override;
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void OnData(int voe_channel, const void* audio_data, int bits_per_sample,
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int sample_rate, int number_of_channels,
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size_t number_of_frames) override;
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void PushCaptureData(int voe_channel, const void* audio_data,
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int bits_per_sample, int sample_rate,
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int number_of_channels,
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size_t number_of_frames) override;
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void PullRenderData(int bits_per_sample, int sample_rate,
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int number_of_channels, size_t number_of_frames,
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void* audio_data, int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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// AudioDeviceObserver
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void OnErrorIsReported(ErrorCode error) override;
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void OnWarningIsReported(WarningCode warning) override;
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protected:
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VoEBaseImpl(voe::SharedData* shared);
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~VoEBaseImpl() override;
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private:
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int32_t StartPlayout();
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int32_t StopPlayout();
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int32_t StartSend();
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int32_t StopSend();
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int32_t TerminateInternal();
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// Helper function to process the recorded data with AudioProcessing Module,
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// demultiplex the data to specific voe channels, encode and send to the
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// network. When |number_of_VoE_channels| is 0, it will demultiplex the
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// data to all the existing VoE channels.
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// It returns new AGC microphone volume or 0 if no volume changes
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// should be done.
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int ProcessRecordedDataWithAPM(
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const int voe_channels[], int number_of_voe_channels,
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const void* audio_data, uint32_t sample_rate, uint8_t number_of_channels,
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size_t number_of_frames, uint32_t audio_delay_milliseconds,
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int32_t clock_drift, uint32_t volume, bool key_pressed);
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void GetPlayoutData(int sample_rate, int number_of_channels,
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size_t number_of_frames, bool feed_data_to_apm,
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void* audio_data, int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms);
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int32_t AddVoEVersion(char* str) const;
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// Initialize channel by setting Engine Information then initializing
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// channel.
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int InitializeChannel(voe::ChannelOwner* channel_owner);
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#ifdef WEBRTC_EXTERNAL_TRANSPORT
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int32_t AddExternalTransportBuild(char* str) const;
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#endif
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VoiceEngineObserver* voiceEngineObserverPtr_;
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CriticalSectionWrapper& callbackCritSect_;
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AudioFrame audioFrame_;
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voe::SharedData* shared_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
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