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Diffstat (limited to 'third_party/libwebrtc/audio/voip/voip_core.h')
| -rw-r--r-- | third_party/libwebrtc/audio/voip/voip_core.h | 174 |
1 files changed, 174 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/voip/voip_core.h b/third_party/libwebrtc/audio/voip/voip_core.h new file mode 100644 index 0000000000..6c3aec6fa2 --- /dev/null +++ b/third_party/libwebrtc/audio/voip/voip_core.h @@ -0,0 +1,174 @@ +/* + * 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 "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. + 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); + ~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_; + + 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_ |