blob: 826117b9aa8300a273241d1018c75c7873378264 [file] [log] [blame]
* Copyright (c) 2015 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.
#include "webrtc/api/audio/audio_mixer.h"
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/call/rtp_packet_sink_interface.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/race_checker.h"
#include "webrtc/rtc_base/thread_checker.h"
#include "webrtc/voice_engine/channel_manager.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include <memory>
#include <string>
#include <vector>
namespace webrtc {
class AudioSinkInterface;
class PacketRouter;
class RtcEventLog;
class RtcpBandwidthObserver;
class RtcpRttStats;
class RtpPacketSender;
class RtpPacketReceived;
class RtpReceiver;
class RtpRtcp;
class RtpTransportControllerSendInterface;
class Transport;
class TransportFeedbackObserver;
namespace voe {
class Channel;
// This class provides the "view" of a voe::Channel that we need to implement
// webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
// purposes:
// 1. Allow mocking just the interfaces used, instead of the entire
// voe::Channel class.
// 2. Provide a refined interface for the stream classes, including assumptions
// on return values and input adaptation.
class ChannelProxy : public RtpPacketSinkInterface {
explicit ChannelProxy(const ChannelOwner& channel_owner);
virtual ~ChannelProxy();
virtual bool SetEncoder(int payload_type,
std::unique_ptr<AudioEncoder> encoder);
virtual void ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
virtual void SetRTCPStatus(bool enable);
virtual void SetLocalSSRC(uint32_t ssrc);
virtual void SetRTCP_CNAME(const std::string& c_name);
virtual void SetNACKStatus(bool enable, int max_packets);
virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
virtual void EnableSendTransportSequenceNumber(int id);
virtual void EnableReceiveTransportSequenceNumber(int id);
virtual void RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer);
virtual void RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router);
virtual void ResetSenderCongestionControlObjects();
virtual void ResetReceiverCongestionControlObjects();
virtual CallStatistics GetRTCPStatistics() const;
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
virtual NetworkStatistics GetNetworkStatistics() const;
virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
virtual ANAStats GetANAStatistics() const;
virtual int GetSpeechOutputLevel() const;
virtual int GetSpeechOutputLevelFullRange() const;
// See description of "totalAudioEnergy" in the WebRTC stats spec:
virtual double GetTotalOutputEnergy() const;
virtual double GetTotalOutputDuration() const;
virtual uint32_t GetDelayEstimate() const;
virtual bool SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency);
virtual bool SendTelephoneEventOutband(int event, int duration_ms);
virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
virtual void SetRecPayloadType(int payload_type,
const SdpAudioFormat& format);
virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
virtual void SetInputMute(bool muted);
virtual void RegisterExternalTransport(Transport* transport);
virtual void DeRegisterExternalTransport();
// Implements RtpPacketSinkInterface
void OnRtpPacket(const RtpPacketReceived& packet) override;
virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
virtual const rtc::scoped_refptr<AudioDecoderFactory>&
GetAudioDecoderFactory() const;
virtual void SetChannelOutputVolumeScaling(float scaling);
virtual void SetRtcEventLog(RtcEventLog* event_log);
virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame);
virtual int NeededFrequency() const;
virtual void SetTransportOverhead(int transport_overhead_per_packet);
virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
virtual void DisassociateSendChannel();
virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp,
RtpReceiver** rtp_receiver) const;
virtual uint32_t GetPlayoutTimestamp() const;
virtual void SetMinimumPlayoutDelay(int delay_ms);
virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
virtual bool GetRecCodec(CodecInst* codec_inst) const;
virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
virtual void OnRecoverableUplinkPacketLossRate(
float recoverable_packet_loss_rate);
virtual void RegisterLegacyReceiveCodecs();
virtual std::vector<webrtc::RtpSource> GetSources() const;
Channel* channel() const;
// Thread checkers document and lock usage of some methods on voe::Channel to
// specific threads we know about. The goal is to eventually split up
// voe::Channel into parts with single-threaded semantics, and thereby reduce
// the need for locks.
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker module_process_thread_checker_;
// Methods accessed from audio and video threads are checked for sequential-
// only access. We don't necessarily own and control these threads, so thread
// checkers cannot be used. E.g. Chromium may transfer "ownership" from one
// audio thread to another, but access is still sequential.
rtc::RaceChecker audio_thread_race_checker_;
rtc::RaceChecker video_capture_thread_race_checker_;
ChannelOwner channel_owner_;
} // namespace voe
} // namespace webrtc