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/*
* Copyright (c) 2020 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 AUDIO_VOIP_VOIP_CORE_H_
#define AUDIO_VOIP_VOIP_CORE_H_
#include <map>
#include <memory>
#include <queue>
#include <unordered_map>
#include <vector>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/voip/voip_base.h"
#include "api/voip/voip_codec.h"
#include "api/voip/voip_dtmf.h"
#include "api/voip/voip_engine.h"
#include "api/voip/voip_network.h"
#include "api/voip/voip_statistics.h"
#include "api/voip/voip_volume_control.h"
#include "audio/audio_transport_impl.h"
#include "audio/voip/audio_channel.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
// VoipCore is the implementatino of VoIP APIs listed in api/voip directory.
// It manages a vector of AudioChannel objects where each is mapped with a
// ChannelId (int) type. ChannelId is the primary key to locate a specific
// AudioChannel object to operate requested VoIP API from the caller.
//
// This class receives required audio components from caller at construction and
// owns the life cycle of them to orchestrate the proper destruction sequence.
class VoipCore : public VoipEngine,
public VoipBase,
public VoipNetwork,
public VoipCodec,
public VoipDtmf,
public VoipStatistics,
public VoipVolumeControl {
public:
// Construct VoipCore with provided arguments.
// ProcessThread implementation can be injected by |process_thread|
// (mainly for testing purpose) and when set to nullptr, default
// implementation will be used.
VoipCore(rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
std::unique_ptr<TaskQueueFactory> task_queue_factory,
rtc::scoped_refptr<AudioDeviceModule> audio_device_module,
rtc::scoped_refptr<AudioProcessing> audio_processing,
std::unique_ptr<ProcessThread> process_thread = nullptr);
~VoipCore() override = default;
// Implements VoipEngine interfaces.
VoipBase& Base() override { return *this; }
VoipNetwork& Network() override { return *this; }
VoipCodec& Codec() override { return *this; }
VoipDtmf& Dtmf() override { return *this; }
VoipStatistics& Statistics() override { return *this; }
VoipVolumeControl& VolumeControl() override { return *this; }
// Implements VoipBase interfaces.
ChannelId CreateChannel(Transport* transport,
absl::optional<uint32_t> local_ssrc) override;
VoipResult ReleaseChannel(ChannelId channel_id) override;
VoipResult StartSend(ChannelId channel_id) override;
VoipResult StopSend(ChannelId channel_id) override;
VoipResult StartPlayout(ChannelId channel_id) override;
VoipResult StopPlayout(ChannelId channel_id) override;
// Implements VoipNetwork interfaces.
VoipResult ReceivedRTPPacket(
ChannelId channel_id,
rtc::ArrayView<const uint8_t> rtp_packet) override;
VoipResult ReceivedRTCPPacket(
ChannelId channel_id,
rtc::ArrayView<const uint8_t> rtcp_packet) override;
// Implements VoipCodec interfaces.
VoipResult SetSendCodec(ChannelId channel_id,
int payload_type,
const SdpAudioFormat& encoder_format) override;
VoipResult SetReceiveCodecs(
ChannelId channel_id,
const std::map<int, SdpAudioFormat>& decoder_specs) override;
// Implements VoipDtmf interfaces.
VoipResult RegisterTelephoneEventType(ChannelId channel_id,
int rtp_payload_type,
int sample_rate_hz) override;
VoipResult SendDtmfEvent(ChannelId channel_id,
DtmfEvent dtmf_event,
int duration_ms) override;
// Implements VoipStatistics interfaces.
VoipResult GetIngressStatistics(ChannelId channel_id,
IngressStatistics& ingress_stats) override;
VoipResult GetChannelStatistics(ChannelId channe_id,
ChannelStatistics& channel_stats) override;
// Implements VoipVolumeControl interfaces.
VoipResult SetInputMuted(ChannelId channel_id, bool enable) override;
VoipResult GetInputVolumeInfo(ChannelId channel_id,
VolumeInfo& volume_info) override;
VoipResult GetOutputVolumeInfo(ChannelId channel_id,
VolumeInfo& volume_info) override;
private:
// Initialize ADM and default audio device if needed.
// Returns true if ADM is successfully initialized or already in such state
// (e.g called more than once). Returns false when ADM fails to initialize
// which would presumably render further processing useless. Note that such
// failure won't necessarily succeed in next initialization attempt as it
// would mean changing the ADM implementation. From Android N and onwards, the
// mobile app may not be able to gain microphone access when in background
// mode. Therefore it would be better to delay the logic as late as possible.
bool InitializeIfNeeded();
// Fetches the corresponding AudioChannel assigned with given |channel|.
// Returns nullptr if not found.
rtc::scoped_refptr<AudioChannel> GetChannel(ChannelId channel_id);
// Updates AudioTransportImpl with a new set of actively sending AudioSender
// (AudioEgress). This needs to be invoked whenever StartSend/StopSend is
// involved by caller. Returns false when the selected audio device fails to
// initialize where it can't expect to deliver any audio input sample.
bool UpdateAudioTransportWithSenders();
// Synchronization for these are handled internally.
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
std::unique_ptr<TaskQueueFactory> task_queue_factory_;
// Synchronization is handled internally by AudioProcessing.
// Must be placed before |audio_device_module_| for proper destruction.
rtc::scoped_refptr<AudioProcessing> audio_processing_;
// Synchronization is handled internally by AudioMixer.
// Must be placed before |audio_device_module_| for proper destruction.
rtc::scoped_refptr<AudioMixer> audio_mixer_;
// Synchronization is handled internally by AudioTransportImpl.
// Must be placed before |audio_device_module_| for proper destruction.
std::unique_ptr<AudioTransportImpl> audio_transport_;
// Synchronization is handled internally by AudioDeviceModule.
rtc::scoped_refptr<AudioDeviceModule> audio_device_module_;
// Synchronization is handled internally by ProcessThread.
// Must be placed before |channels_| for proper destruction.
std::unique_ptr<ProcessThread> process_thread_;
Mutex lock_;
// Member to track a next ChannelId for new AudioChannel.
int next_channel_id_ RTC_GUARDED_BY(lock_) = 0;
// Container to track currently active AudioChannel objects mapped by
// ChannelId.
std::unordered_map<ChannelId, rtc::scoped_refptr<AudioChannel>> channels_
RTC_GUARDED_BY(lock_);
// Boolean flag to ensure initialization only occurs once.
bool initialized_ RTC_GUARDED_BY(lock_) = false;
};
} // namespace webrtc
#endif // AUDIO_VOIP_VOIP_CORE_H_
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