599 lines
22 KiB
C
599 lines
22 KiB
C
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/*
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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_CHANNEL_H_
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#define WEBRTC_VOICE_ENGINE_CHANNEL_H_
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#include <memory>
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#include "webrtc/audio_sink.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
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#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
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#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
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#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
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#include "webrtc/modules/audio_processing/rms_level.h"
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#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/utility/include/file_player.h"
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#include "webrtc/modules/utility/include/file_recorder.h"
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#include "webrtc/voice_engine/include/voe_audio_processing.h"
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#include "webrtc/voice_engine/include/voe_network.h"
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#include "webrtc/voice_engine/level_indicator.h"
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#include "webrtc/voice_engine/network_predictor.h"
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#include "webrtc/voice_engine/shared_data.h"
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#include "webrtc/voice_engine/voice_engine_defines.h"
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namespace rtc {
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class TimestampWrapAroundHandler;
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}
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namespace webrtc {
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class AudioDeviceModule;
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class Config;
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class FileWrapper;
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class PacketRouter;
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class ProcessThread;
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class ReceiveStatistics;
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class RemoteNtpTimeEstimator;
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class RtcEventLog;
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class RTPPayloadRegistry;
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class RtpReceiver;
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class RTPReceiverAudio;
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class RtpRtcp;
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class TelephoneEventHandler;
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class VoEMediaProcess;
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class VoERTPObserver;
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class VoiceEngineObserver;
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struct CallStatistics;
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struct ReportBlock;
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struct SenderInfo;
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namespace voe {
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class OutputMixer;
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class RtpPacketSenderProxy;
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class Statistics;
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class StatisticsProxy;
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class TransportFeedbackProxy;
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class TransmitMixer;
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class TransportSequenceNumberProxy;
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class VoERtcpObserver;
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// Helper class to simplify locking scheme for members that are accessed from
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// multiple threads.
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// Example: a member can be set on thread T1 and read by an internal audio
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// thread T2. Accessing the member via this class ensures that we are
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// safe and also avoid TSan v2 warnings.
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class ChannelState {
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public:
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struct State {
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State()
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: rx_apm_is_enabled(false),
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input_external_media(false),
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output_file_playing(false),
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input_file_playing(false),
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playing(false),
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sending(false),
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receiving(false) {}
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bool rx_apm_is_enabled;
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bool input_external_media;
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bool output_file_playing;
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bool input_file_playing;
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bool playing;
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bool sending;
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bool receiving;
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};
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ChannelState() {}
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virtual ~ChannelState() {}
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void Reset() {
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rtc::CritScope lock(&lock_);
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state_ = State();
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}
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State Get() const {
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rtc::CritScope lock(&lock_);
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return state_;
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}
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void SetRxApmIsEnabled(bool enable) {
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rtc::CritScope lock(&lock_);
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state_.rx_apm_is_enabled = enable;
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}
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void SetInputExternalMedia(bool enable) {
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rtc::CritScope lock(&lock_);
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state_.input_external_media = enable;
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}
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void SetOutputFilePlaying(bool enable) {
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rtc::CritScope lock(&lock_);
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state_.output_file_playing = enable;
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}
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void SetInputFilePlaying(bool enable) {
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rtc::CritScope lock(&lock_);
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state_.input_file_playing = enable;
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}
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void SetPlaying(bool enable) {
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rtc::CritScope lock(&lock_);
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state_.playing = enable;
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}
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void SetSending(bool enable) {
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rtc::CritScope lock(&lock_);
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state_.sending = enable;
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}
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void SetReceiving(bool enable) {
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rtc::CritScope lock(&lock_);
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state_.receiving = enable;
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}
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private:
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rtc::CriticalSection lock_;
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State state_;
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};
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class Channel
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: public RtpData,
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public RtpFeedback,
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public FileCallback, // receiving notification from file player &
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// recorder
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public Transport,
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public AudioPacketizationCallback, // receive encoded packets from the
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// ACM
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public ACMVADCallback, // receive voice activity from the ACM
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public MixerParticipant // supplies output mixer with audio frames
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{
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public:
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friend class VoERtcpObserver;
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enum { KNumSocketThreads = 1 };
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enum { KNumberOfSocketBuffers = 8 };
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virtual ~Channel();
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static int32_t CreateChannel(Channel*& channel,
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int32_t channelId,
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uint32_t instanceId,
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RtcEventLog* const event_log,
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const Config& config);
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static int32_t CreateChannel(
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Channel*& channel,
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int32_t channelId,
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uint32_t instanceId,
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RtcEventLog* const event_log,
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const Config& config,
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const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
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Channel(int32_t channelId,
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uint32_t instanceId,
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RtcEventLog* const event_log,
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const Config& config,
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const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
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int32_t Init();
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int32_t SetEngineInformation(Statistics& engineStatistics,
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OutputMixer& outputMixer,
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TransmitMixer& transmitMixer,
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ProcessThread& moduleProcessThread,
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AudioDeviceModule& audioDeviceModule,
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VoiceEngineObserver* voiceEngineObserver,
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rtc::CriticalSection* callbackCritSect);
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int32_t UpdateLocalTimeStamp();
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void SetSink(std::unique_ptr<AudioSinkInterface> sink);
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// TODO(ossu): Don't use! It's only here to confirm that the decoder factory
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// passed into AudioReceiveStream is the same as the one set when creating the
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// ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can
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// go.
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const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
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// API methods
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// VoEBase
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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 StartReceiving();
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int32_t StopReceiving();
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int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
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int32_t DeRegisterVoiceEngineObserver();
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// VoECodec
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int32_t GetSendCodec(CodecInst& codec);
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int32_t GetRecCodec(CodecInst& codec);
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int32_t SetSendCodec(const CodecInst& codec);
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void SetBitRate(int bitrate_bps);
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int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX);
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int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX);
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int32_t SetRecPayloadType(const CodecInst& codec);
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int32_t GetRecPayloadType(CodecInst& codec);
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int32_t SetSendCNPayloadType(int type, PayloadFrequencies frequency);
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int SetOpusMaxPlaybackRate(int frequency_hz);
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int SetOpusDtx(bool enable_dtx);
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// VoENetwork
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int32_t RegisterExternalTransport(Transport* transport);
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int32_t DeRegisterExternalTransport();
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int32_t ReceivedRTPPacket(const uint8_t* received_packet,
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size_t length,
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const PacketTime& packet_time);
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int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
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// VoEFile
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int StartPlayingFileLocally(const char* fileName,
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bool loop,
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FileFormats format,
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int startPosition,
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float volumeScaling,
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int stopPosition,
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const CodecInst* codecInst);
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int StartPlayingFileLocally(InStream* stream,
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FileFormats format,
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int startPosition,
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float volumeScaling,
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int stopPosition,
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const CodecInst* codecInst);
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int StopPlayingFileLocally();
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int IsPlayingFileLocally() const;
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int RegisterFilePlayingToMixer();
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int StartPlayingFileAsMicrophone(const char* fileName,
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bool loop,
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FileFormats format,
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int startPosition,
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float volumeScaling,
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int stopPosition,
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const CodecInst* codecInst);
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int StartPlayingFileAsMicrophone(InStream* stream,
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FileFormats format,
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int startPosition,
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float volumeScaling,
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int stopPosition,
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const CodecInst* codecInst);
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int StopPlayingFileAsMicrophone();
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int IsPlayingFileAsMicrophone() const;
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int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
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int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
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int StopRecordingPlayout();
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void SetMixWithMicStatus(bool mix);
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// VoEExternalMediaProcessing
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int RegisterExternalMediaProcessing(ProcessingTypes type,
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VoEMediaProcess& processObject);
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int DeRegisterExternalMediaProcessing(ProcessingTypes type);
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int SetExternalMixing(bool enabled);
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// VoEVolumeControl
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int GetSpeechOutputLevel(uint32_t& level) const;
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int GetSpeechOutputLevelFullRange(uint32_t& level) const;
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int SetInputMute(bool enable);
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bool InputMute() const;
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int SetOutputVolumePan(float left, float right);
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int GetOutputVolumePan(float& left, float& right) const;
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int SetChannelOutputVolumeScaling(float scaling);
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int GetChannelOutputVolumeScaling(float& scaling) const;
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// VoENetEqStats
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int GetNetworkStatistics(NetworkStatistics& stats);
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void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
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// VoEVideoSync
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bool GetDelayEstimate(int* jitter_buffer_delay_ms,
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int* playout_buffer_delay_ms) const;
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uint32_t GetDelayEstimate() const;
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int LeastRequiredDelayMs() const;
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int SetMinimumPlayoutDelay(int delayMs);
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int GetPlayoutTimestamp(unsigned int& timestamp);
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int SetInitTimestamp(unsigned int timestamp);
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int SetInitSequenceNumber(short sequenceNumber);
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// VoEVideoSyncExtended
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int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
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// DTMF
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int SendTelephoneEventOutband(int event, int duration_ms);
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int SetSendTelephoneEventPayloadType(int payload_type);
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// VoEAudioProcessingImpl
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int UpdateRxVadDetection(AudioFrame& audioFrame);
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int RegisterRxVadObserver(VoERxVadCallback& observer);
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int DeRegisterRxVadObserver();
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int VoiceActivityIndicator(int& activity);
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#ifdef WEBRTC_VOICE_ENGINE_AGC
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int SetRxAgcStatus(bool enable, AgcModes mode);
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int GetRxAgcStatus(bool& enabled, AgcModes& mode);
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int SetRxAgcConfig(AgcConfig config);
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int GetRxAgcConfig(AgcConfig& config);
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#endif
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#ifdef WEBRTC_VOICE_ENGINE_NR
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int SetRxNsStatus(bool enable, NsModes mode);
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int GetRxNsStatus(bool& enabled, NsModes& mode);
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#endif
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// VoERTP_RTCP
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int SetLocalSSRC(unsigned int ssrc);
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int GetLocalSSRC(unsigned int& ssrc);
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int GetRemoteSSRC(unsigned int& ssrc);
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int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
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int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
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int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
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int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
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void EnableSendTransportSequenceNumber(int id);
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void EnableReceiveTransportSequenceNumber(int id);
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void RegisterSenderCongestionControlObjects(
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RtpPacketSender* rtp_packet_sender,
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TransportFeedbackObserver* transport_feedback_observer,
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PacketRouter* packet_router);
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void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
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void ResetCongestionControlObjects();
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void SetRTCPStatus(bool enable);
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int GetRTCPStatus(bool& enabled);
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int SetRTCP_CNAME(const char cName[256]);
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int GetRemoteRTCP_CNAME(char cName[256]);
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int GetRemoteRTCPData(unsigned int& NTPHigh,
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unsigned int& NTPLow,
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unsigned int& timestamp,
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unsigned int& playoutTimestamp,
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unsigned int* jitter,
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unsigned short* fractionLost);
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int SendApplicationDefinedRTCPPacket(unsigned char subType,
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unsigned int name,
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const char* data,
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unsigned short dataLengthInBytes);
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int GetRTPStatistics(unsigned int& averageJitterMs,
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unsigned int& maxJitterMs,
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unsigned int& discardedPackets);
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int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
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int GetRTPStatistics(CallStatistics& stats);
|
||
|
|
int SetCodecFECStatus(bool enable);
|
||
|
|
bool GetCodecFECStatus();
|
||
|
|
void SetNACKStatus(bool enable, int maxNumberOfPackets);
|
||
|
|
|
||
|
|
// From AudioPacketizationCallback in the ACM
|
||
|
|
int32_t SendData(FrameType frameType,
|
||
|
|
uint8_t payloadType,
|
||
|
|
uint32_t timeStamp,
|
||
|
|
const uint8_t* payloadData,
|
||
|
|
size_t payloadSize,
|
||
|
|
const RTPFragmentationHeader* fragmentation) override;
|
||
|
|
|
||
|
|
// From ACMVADCallback in the ACM
|
||
|
|
int32_t InFrameType(FrameType frame_type) override;
|
||
|
|
|
||
|
|
int32_t OnRxVadDetected(int vadDecision);
|
||
|
|
|
||
|
|
// From RtpData in the RTP/RTCP module
|
||
|
|
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
|
||
|
|
size_t payloadSize,
|
||
|
|
const WebRtcRTPHeader* rtpHeader) override;
|
||
|
|
bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
|
||
|
|
|
||
|
|
// From RtpFeedback in the RTP/RTCP module
|
||
|
|
int32_t OnInitializeDecoder(int8_t payloadType,
|
||
|
|
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
||
|
|
int frequency,
|
||
|
|
size_t channels,
|
||
|
|
uint32_t rate) override;
|
||
|
|
void OnIncomingSSRCChanged(uint32_t ssrc) override;
|
||
|
|
void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
|
||
|
|
|
||
|
|
// From Transport (called by the RTP/RTCP module)
|
||
|
|
bool SendRtp(const uint8_t* data,
|
||
|
|
size_t len,
|
||
|
|
const PacketOptions& packet_options) override;
|
||
|
|
bool SendRtcp(const uint8_t* data, size_t len) override;
|
||
|
|
|
||
|
|
// From MixerParticipant
|
||
|
|
MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted(
|
||
|
|
int32_t id,
|
||
|
|
AudioFrame* audioFrame) override;
|
||
|
|
int32_t NeededFrequency(int32_t id) const override;
|
||
|
|
|
||
|
|
// From FileCallback
|
||
|
|
void PlayNotification(int32_t id, uint32_t durationMs) override;
|
||
|
|
void RecordNotification(int32_t id, uint32_t durationMs) override;
|
||
|
|
void PlayFileEnded(int32_t id) override;
|
||
|
|
void RecordFileEnded(int32_t id) override;
|
||
|
|
|
||
|
|
uint32_t InstanceId() const { return _instanceId; }
|
||
|
|
int32_t ChannelId() const { return _channelId; }
|
||
|
|
bool Playing() const { return channel_state_.Get().playing; }
|
||
|
|
bool Sending() const { return channel_state_.Get().sending; }
|
||
|
|
bool Receiving() const { return channel_state_.Get().receiving; }
|
||
|
|
bool ExternalTransport() const {
|
||
|
|
rtc::CritScope cs(&_callbackCritSect);
|
||
|
|
return _externalTransport;
|
||
|
|
}
|
||
|
|
bool ExternalMixing() const { return _externalMixing; }
|
||
|
|
RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
|
||
|
|
int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
|
||
|
|
uint32_t Demultiplex(const AudioFrame& audioFrame);
|
||
|
|
// Demultiplex the data to the channel's |_audioFrame|. The difference
|
||
|
|
// between this method and the overloaded method above is that |audio_data|
|
||
|
|
// does not go through transmit_mixer and APM.
|
||
|
|
void Demultiplex(const int16_t* audio_data,
|
||
|
|
int sample_rate,
|
||
|
|
size_t number_of_frames,
|
||
|
|
size_t number_of_channels);
|
||
|
|
uint32_t PrepareEncodeAndSend(int mixingFrequency);
|
||
|
|
uint32_t EncodeAndSend();
|
||
|
|
|
||
|
|
// Associate to a send channel.
|
||
|
|
// Used for obtaining RTT for a receive-only channel.
|
||
|
|
void set_associate_send_channel(const ChannelOwner& channel) {
|
||
|
|
assert(_channelId != channel.channel()->ChannelId());
|
||
|
|
rtc::CritScope lock(&assoc_send_channel_lock_);
|
||
|
|
associate_send_channel_ = channel;
|
||
|
|
}
|
||
|
|
|
||
|
|
// Disassociate a send channel if it was associated.
|
||
|
|
void DisassociateSendChannel(int channel_id);
|
||
|
|
|
||
|
|
protected:
|
||
|
|
void OnIncomingFractionLoss(int fraction_lost);
|
||
|
|
|
||
|
|
private:
|
||
|
|
bool ReceivePacket(const uint8_t* packet,
|
||
|
|
size_t packet_length,
|
||
|
|
const RTPHeader& header,
|
||
|
|
bool in_order);
|
||
|
|
bool HandleRtxPacket(const uint8_t* packet,
|
||
|
|
size_t packet_length,
|
||
|
|
const RTPHeader& header);
|
||
|
|
bool IsPacketInOrder(const RTPHeader& header) const;
|
||
|
|
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
|
||
|
|
int ResendPackets(const uint16_t* sequence_numbers, int length);
|
||
|
|
int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
|
||
|
|
int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
|
||
|
|
void UpdatePlayoutTimestamp(bool rtcp);
|
||
|
|
void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber);
|
||
|
|
void RegisterReceiveCodecsToRTPModule();
|
||
|
|
|
||
|
|
int SetSendRtpHeaderExtension(bool enable,
|
||
|
|
RTPExtensionType type,
|
||
|
|
unsigned char id);
|
||
|
|
|
||
|
|
int32_t GetPlayoutFrequency();
|
||
|
|
int64_t GetRTT(bool allow_associate_channel) const;
|
||
|
|
|
||
|
|
rtc::CriticalSection _fileCritSect;
|
||
|
|
rtc::CriticalSection _callbackCritSect;
|
||
|
|
rtc::CriticalSection volume_settings_critsect_;
|
||
|
|
uint32_t _instanceId;
|
||
|
|
int32_t _channelId;
|
||
|
|
|
||
|
|
ChannelState channel_state_;
|
||
|
|
|
||
|
|
RtcEventLog* const event_log_;
|
||
|
|
|
||
|
|
std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
|
||
|
|
std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
|
||
|
|
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
|
||
|
|
std::unique_ptr<StatisticsProxy> statistics_proxy_;
|
||
|
|
std::unique_ptr<RtpReceiver> rtp_receiver_;
|
||
|
|
TelephoneEventHandler* telephone_event_handler_;
|
||
|
|
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
|
||
|
|
std::unique_ptr<AudioCodingModule> audio_coding_;
|
||
|
|
acm2::CodecManager codec_manager_;
|
||
|
|
acm2::RentACodec rent_a_codec_;
|
||
|
|
std::unique_ptr<AudioSinkInterface> audio_sink_;
|
||
|
|
AudioLevel _outputAudioLevel;
|
||
|
|
bool _externalTransport;
|
||
|
|
AudioFrame _audioFrame;
|
||
|
|
// Downsamples to the codec rate if necessary.
|
||
|
|
PushResampler<int16_t> input_resampler_;
|
||
|
|
FilePlayer* _inputFilePlayerPtr;
|
||
|
|
FilePlayer* _outputFilePlayerPtr;
|
||
|
|
FileRecorder* _outputFileRecorderPtr;
|
||
|
|
int _inputFilePlayerId;
|
||
|
|
int _outputFilePlayerId;
|
||
|
|
int _outputFileRecorderId;
|
||
|
|
bool _outputFileRecording;
|
||
|
|
bool _outputExternalMedia;
|
||
|
|
VoEMediaProcess* _inputExternalMediaCallbackPtr;
|
||
|
|
VoEMediaProcess* _outputExternalMediaCallbackPtr;
|
||
|
|
uint32_t _timeStamp;
|
||
|
|
|
||
|
|
RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
|
||
|
|
|
||
|
|
// Timestamp of the audio pulled from NetEq.
|
||
|
|
rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
|
||
|
|
uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
|
||
|
|
uint32_t playout_timestamp_rtcp_;
|
||
|
|
uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
|
||
|
|
uint32_t _numberOfDiscardedPackets;
|
||
|
|
uint16_t send_sequence_number_;
|
||
|
|
uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
|
||
|
|
|
||
|
|
rtc::CriticalSection ts_stats_lock_;
|
||
|
|
|
||
|
|
std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
|
||
|
|
// The rtp timestamp of the first played out audio frame.
|
||
|
|
int64_t capture_start_rtp_time_stamp_;
|
||
|
|
// The capture ntp time (in local timebase) of the first played out audio
|
||
|
|
// frame.
|
||
|
|
int64_t capture_start_ntp_time_ms_ GUARDED_BY(ts_stats_lock_);
|
||
|
|
|
||
|
|
// uses
|
||
|
|
Statistics* _engineStatisticsPtr;
|
||
|
|
OutputMixer* _outputMixerPtr;
|
||
|
|
TransmitMixer* _transmitMixerPtr;
|
||
|
|
ProcessThread* _moduleProcessThreadPtr;
|
||
|
|
AudioDeviceModule* _audioDeviceModulePtr;
|
||
|
|
VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base
|
||
|
|
rtc::CriticalSection* _callbackCritSectPtr; // owned by base
|
||
|
|
Transport* _transportPtr; // WebRtc socket or external transport
|
||
|
|
RMSLevel rms_level_;
|
||
|
|
std::unique_ptr<AudioProcessing> rx_audioproc_; // far end AudioProcessing
|
||
|
|
VoERxVadCallback* _rxVadObserverPtr;
|
||
|
|
int32_t _oldVadDecision;
|
||
|
|
int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
|
||
|
|
// VoEBase
|
||
|
|
bool _externalMixing;
|
||
|
|
bool _mixFileWithMicrophone;
|
||
|
|
// VoEVolumeControl
|
||
|
|
bool input_mute_ GUARDED_BY(volume_settings_critsect_);
|
||
|
|
bool previous_frame_muted_; // Only accessed from PrepareEncodeAndSend().
|
||
|
|
float _panLeft GUARDED_BY(volume_settings_critsect_);
|
||
|
|
float _panRight GUARDED_BY(volume_settings_critsect_);
|
||
|
|
float _outputGain GUARDED_BY(volume_settings_critsect_);
|
||
|
|
// VoeRTP_RTCP
|
||
|
|
uint32_t _lastLocalTimeStamp;
|
||
|
|
int8_t _lastPayloadType;
|
||
|
|
bool _includeAudioLevelIndication;
|
||
|
|
// VoENetwork
|
||
|
|
AudioFrame::SpeechType _outputSpeechType;
|
||
|
|
// VoEVideoSync
|
||
|
|
rtc::CriticalSection video_sync_lock_;
|
||
|
|
uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
|
||
|
|
uint32_t _previousTimestamp;
|
||
|
|
uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
|
||
|
|
// VoEAudioProcessing
|
||
|
|
bool _RxVadDetection;
|
||
|
|
bool _rxAgcIsEnabled;
|
||
|
|
bool _rxNsIsEnabled;
|
||
|
|
bool restored_packet_in_use_;
|
||
|
|
// RtcpBandwidthObserver
|
||
|
|
std::unique_ptr<VoERtcpObserver> rtcp_observer_;
|
||
|
|
std::unique_ptr<NetworkPredictor> network_predictor_;
|
||
|
|
// An associated send channel.
|
||
|
|
rtc::CriticalSection assoc_send_channel_lock_;
|
||
|
|
ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
|
||
|
|
|
||
|
|
bool pacing_enabled_;
|
||
|
|
PacketRouter* packet_router_ = nullptr;
|
||
|
|
std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
|
||
|
|
std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
|
||
|
|
std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
|
||
|
|
|
||
|
|
// TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
|
||
|
|
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
|
||
|
|
};
|
||
|
|
|
||
|
|
} // namespace voe
|
||
|
|
} // namespace webrtc
|
||
|
|
|
||
|
|
#endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
|