312 lines
12 KiB
C
312 lines
12 KiB
C
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/*
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* Copyright (c) 2004 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_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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#define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/audio_state.h"
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#include "webrtc/base/buffer.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/networkroute.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/base/stream.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/call.h"
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#include "webrtc/common.h"
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#include "webrtc/config.h"
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#include "webrtc/media/base/rtputils.h"
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#include "webrtc/media/engine/webrtccommon.h"
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#include "webrtc/media/engine/webrtcvoe.h"
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#include "webrtc/pc/channel.h"
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namespace cricket {
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class AudioDeviceModule;
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class AudioSource;
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class VoEWrapper;
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class WebRtcVoiceMediaChannel;
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struct SendCodecSpec {
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SendCodecSpec() {
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webrtc::CodecInst empty_inst = {0};
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codec_inst = empty_inst;
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codec_inst.pltype = -1;
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}
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bool operator==(const SendCodecSpec& rhs) const;
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bool operator!=(const SendCodecSpec& rhs) const;
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bool nack_enabled = false;
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bool transport_cc_enabled = false;
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bool enable_codec_fec = false;
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bool enable_opus_dtx = false;
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int opus_max_playback_rate = 0;
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int red_payload_type = -1;
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int cng_payload_type = -1;
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int cng_plfreq = -1;
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webrtc::CodecInst codec_inst;
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};
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// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
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// It uses the WebRtc VoiceEngine library for audio handling.
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class WebRtcVoiceEngine final : public webrtc::TraceCallback {
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friend class WebRtcVoiceMediaChannel;
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public:
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// Exposed for the WVoE/MC unit test.
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static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
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WebRtcVoiceEngine(
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webrtc::AudioDeviceModule* adm,
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const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory);
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// Dependency injection for testing.
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WebRtcVoiceEngine(
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webrtc::AudioDeviceModule* adm,
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const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
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VoEWrapper* voe_wrapper);
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~WebRtcVoiceEngine() override;
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rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
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VoiceMediaChannel* CreateChannel(webrtc::Call* call,
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const MediaConfig& config,
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const AudioOptions& options);
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int GetInputLevel();
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const std::vector<AudioCodec>& send_codecs() const;
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const std::vector<AudioCodec>& recv_codecs() const;
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RtpCapabilities GetCapabilities() const;
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// For tracking WebRtc channels. Needed because we have to pause them
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// all when switching devices.
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// May only be called by WebRtcVoiceMediaChannel.
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void RegisterChannel(WebRtcVoiceMediaChannel* channel);
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void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
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// Called by WebRtcVoiceMediaChannel to set a gain offset from
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// the default AGC target level.
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bool AdjustAgcLevel(int delta);
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VoEWrapper* voe() { return voe_wrapper_.get(); }
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int GetLastEngineError();
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// Starts AEC dump using an existing file. A maximum file size in bytes can be
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// specified. When the maximum file size is reached, logging is stopped and
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// the file is closed. If max_size_bytes is set to <= 0, no limit will be
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// used.
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bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
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// Stops AEC dump.
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void StopAecDump();
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// Starts recording an RtcEventLog using an existing file until the log file
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// reaches the maximum filesize or the StopRtcEventLog function is called.
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// If the value of max_size_bytes is <= 0, no limit is used.
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bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
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// Stops recording the RtcEventLog.
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void StopRtcEventLog();
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private:
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// Every option that is "set" will be applied. Every option not "set" will be
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// ignored. This allows us to selectively turn on and off different options
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// easily at any time.
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bool ApplyOptions(const AudioOptions& options);
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void SetDefaultDevices();
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// webrtc::TraceCallback:
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void Print(webrtc::TraceLevel level, const char* trace, int length) override;
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void StartAecDump(const std::string& filename);
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int CreateVoEChannel();
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webrtc::AudioDeviceModule* adm();
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rtc::ThreadChecker signal_thread_checker_;
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rtc::ThreadChecker worker_thread_checker_;
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// The audio device manager.
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rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
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// The primary instance of WebRtc VoiceEngine.
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std::unique_ptr<VoEWrapper> voe_wrapper_;
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rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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std::vector<AudioCodec> codecs_;
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std::vector<WebRtcVoiceMediaChannel*> channels_;
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webrtc::Config voe_config_;
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bool is_dumping_aec_ = false;
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webrtc::AgcConfig default_agc_config_;
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// Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns and
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// intelligibility_enhancer values, and apply them in case they are missing
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// in the audio options. We need to do this because SetExtraOptions() will
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// revert to defaults for options which are not provided.
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rtc::Optional<bool> extended_filter_aec_;
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rtc::Optional<bool> delay_agnostic_aec_;
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rtc::Optional<bool> experimental_ns_;
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rtc::Optional<bool> intelligibility_enhancer_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine);
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};
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// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
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// WebRtc Voice Engine.
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class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
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public webrtc::Transport {
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public:
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WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
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const MediaConfig& config,
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const AudioOptions& options,
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webrtc::Call* call);
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~WebRtcVoiceMediaChannel() override;
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const AudioOptions& options() const { return options_; }
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rtc::DiffServCodePoint PreferredDscp() const override;
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bool SetSendParameters(const AudioSendParameters& params) override;
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bool SetRecvParameters(const AudioRecvParameters& params) override;
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webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
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bool SetRtpSendParameters(uint32_t ssrc,
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const webrtc::RtpParameters& parameters) override;
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webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
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bool SetRtpReceiveParameters(
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uint32_t ssrc,
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const webrtc::RtpParameters& parameters) override;
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bool SetPlayout(bool playout) override;
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bool PausePlayout();
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bool ResumePlayout();
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void SetSend(bool send) override;
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bool SetAudioSend(uint32_t ssrc,
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bool enable,
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const AudioOptions* options,
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AudioSource* source) override;
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bool AddSendStream(const StreamParams& sp) override;
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bool RemoveSendStream(uint32_t ssrc) override;
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bool AddRecvStream(const StreamParams& sp) override;
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bool RemoveRecvStream(uint32_t ssrc) override;
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bool GetActiveStreams(AudioInfo::StreamList* actives) override;
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int GetOutputLevel() override;
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int GetTimeSinceLastTyping() override;
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void SetTypingDetectionParameters(int time_window,
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int cost_per_typing,
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int reporting_threshold,
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int penalty_decay,
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int type_event_delay) override;
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bool SetOutputVolume(uint32_t ssrc, double volume) override;
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bool CanInsertDtmf() override;
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bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
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void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time) override;
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void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time) override;
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void OnNetworkRouteChanged(const std::string& transport_name,
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const rtc::NetworkRoute& network_route) override;
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void OnReadyToSend(bool ready) override;
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bool GetStats(VoiceMediaInfo* info) override;
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void SetRawAudioSink(
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uint32_t ssrc,
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std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
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// implements Transport interface
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bool SendRtp(const uint8_t* data,
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size_t len,
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const webrtc::PacketOptions& options) override {
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rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
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rtc::PacketOptions rtc_options;
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rtc_options.packet_id = options.packet_id;
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return VoiceMediaChannel::SendPacket(&packet, rtc_options);
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}
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bool SendRtcp(const uint8_t* data, size_t len) override {
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rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
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return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
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}
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int GetReceiveChannelId(uint32_t ssrc) const;
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int GetSendChannelId(uint32_t ssrc) const;
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private:
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bool SetOptions(const AudioOptions& options);
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bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
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bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
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bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters);
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bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
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bool SetLocalSource(uint32_t ssrc, AudioSource* source);
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bool MuteStream(uint32_t ssrc, bool mute);
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WebRtcVoiceEngine* engine() { return engine_; }
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int GetLastEngineError() { return engine()->GetLastEngineError(); }
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int GetOutputLevel(int channel);
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bool SetPlayout(int channel, bool playout);
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bool ChangePlayout(bool playout);
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int CreateVoEChannel();
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bool DeleteVoEChannel(int channel);
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bool IsDefaultRecvStream(uint32_t ssrc) {
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return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
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}
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bool SetMaxSendBitrate(int bps);
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bool SetChannelSendParameters(int channel,
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const webrtc::RtpParameters& parameters);
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bool SetMaxSendBitrate(int channel, int bps);
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bool HasSendCodec() const {
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return send_codec_spec_.codec_inst.pltype != -1;
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}
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bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
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void SetupRecording();
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rtc::ThreadChecker worker_thread_checker_;
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WebRtcVoiceEngine* const engine_ = nullptr;
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std::vector<AudioCodec> send_codecs_;
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std::vector<AudioCodec> recv_codecs_;
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int max_send_bitrate_bps_ = 0;
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AudioOptions options_;
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rtc::Optional<int> dtmf_payload_type_;
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bool desired_playout_ = false;
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bool recv_transport_cc_enabled_ = false;
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bool recv_nack_enabled_ = false;
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bool playout_ = false;
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bool send_ = false;
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webrtc::Call* const call_ = nullptr;
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// SSRC of unsignalled receive stream, or -1 if there isn't one.
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int64_t default_recv_ssrc_ = -1;
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// Volume for unsignalled stream, which may be set before the stream exists.
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double default_recv_volume_ = 1.0;
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// Sink for unsignalled stream, which may be set before the stream exists.
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std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
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// Default SSRC to use for RTCP receiver reports in case of no signaled
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// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
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// and https://code.google.com/p/chromium/issues/detail?id=547661
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uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
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class WebRtcAudioSendStream;
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std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
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std::vector<webrtc::RtpExtension> send_rtp_extensions_;
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class WebRtcAudioReceiveStream;
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std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
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std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
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SendCodecSpec send_codec_spec_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
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};
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} // namespace cricket
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#endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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