blob: b96be9e343138c352916a95b825da44f80232334 [file] [log] [blame]
/*
* 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.
*/
#ifndef VOICE_ENGINE_VOE_BASE_IMPL_H_
#define VOICE_ENGINE_VOE_BASE_IMPL_H_
#include "voice_engine/include/voe_base.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/criticalsection.h"
#include "voice_engine/shared_data.h"
namespace webrtc {
class ProcessThread;
class VoEBaseImpl : public VoEBase,
public AudioTransport {
public:
int Init(
AudioDeviceModule* audio_device,
AudioProcessing* audio_processing,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) override;
voe::TransmitMixer* transmit_mixer() override {
return shared_->transmit_mixer();
}
void Terminate() override;
int CreateChannel() override;
int CreateChannel(const ChannelConfig& config) override;
int DeleteChannel(int channel) override;
int StartPlayout(int channel) override;
int StartSend(int channel) override;
int StopPlayout(int channel) override;
int StopSend(int channel) override;
int SetPlayout(bool enabled) override;
int SetRecording(bool enabled) override;
AudioTransport* audio_transport() override { return this; }
// AudioTransport
int32_t RecordedDataIsAvailable(const void* audio_data,
const size_t number_of_frames,
const size_t bytes_per_sample,
const size_t number_of_channels,
const uint32_t sample_rate,
const uint32_t audio_delay_milliseconds,
const int32_t clock_drift,
const uint32_t volume,
const bool key_pressed,
uint32_t& new_mic_volume) override;
RTC_DEPRECATED int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override;
void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override;
RTC_DEPRECATED void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override;
protected:
VoEBaseImpl(voe::SharedData* shared);
~VoEBaseImpl() override;
private:
int32_t StartPlayout();
int32_t StopPlayout();
int32_t StartSend();
int32_t StopSend();
void TerminateInternal();
void GetPlayoutData(int sample_rate, size_t number_of_channels,
size_t number_of_frames, bool feed_data_to_apm,
void* audio_data, int64_t* elapsed_time_ms,
int64_t* ntp_time_ms);
// Initialize channel by setting Engine Information then initializing
// channel.
int InitializeChannel(voe::ChannelOwner* channel_owner);
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
AudioFrame audioFrame_;
voe::SharedData* shared_;
bool playout_enabled_ = true;
bool recording_enabled_ = true;
};
} // namespace webrtc
#endif // VOICE_ENGINE_VOE_BASE_IMPL_H_