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340 lines
13 KiB
340 lines
13 KiB
/*
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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 MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_
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#define MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_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 "api/audio_codecs/audio_encoder_factory.h"
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#include "api/scoped_refptr.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "api/transport/rtp/rtp_source.h"
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#include "call/audio_state.h"
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#include "call/call.h"
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#include "media/base/media_engine.h"
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#include "media/base/rtp_utils.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/thread_checker.h"
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namespace cricket {
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class AudioDeviceModule;
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class AudioMixer;
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class AudioSource;
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class WebRtcVoiceMediaChannel;
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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 VoiceEngineInterface {
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friend class WebRtcVoiceMediaChannel;
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public:
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WebRtcVoiceEngine(
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webrtc::TaskQueueFactory* task_queue_factory,
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webrtc::AudioDeviceModule* adm,
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const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
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const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
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rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
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rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
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~WebRtcVoiceEngine() override;
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// Does initialization that needs to occur on the worker thread.
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void Init() override;
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rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
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VoiceMediaChannel* CreateMediaChannel(
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webrtc::Call* call,
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const MediaConfig& config,
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const AudioOptions& options,
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const webrtc::CryptoOptions& crypto_options) override;
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const std::vector<AudioCodec>& send_codecs() const override;
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const std::vector<AudioCodec>& recv_codecs() const override;
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std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
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const override;
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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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// 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(webrtc::FileWrapper file, int64_t max_size_bytes) override;
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// Stops AEC dump.
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void StopAecDump() override;
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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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int CreateVoEChannel();
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webrtc::TaskQueueFactory* const task_queue_factory_;
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std::unique_ptr<rtc::TaskQueue> low_priority_worker_queue_;
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webrtc::AudioDeviceModule* adm();
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webrtc::AudioProcessing* apm() const;
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webrtc::AudioState* audio_state();
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std::vector<AudioCodec> CollectCodecs(
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const std::vector<webrtc::AudioCodecSpec>& specs) const;
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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 module.
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rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
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rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_;
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
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rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer_;
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// The audio processing module.
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rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
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// The primary instance of WebRtc VoiceEngine.
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rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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std::vector<AudioCodec> send_codecs_;
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std::vector<AudioCodec> recv_codecs_;
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std::vector<WebRtcVoiceMediaChannel*> channels_;
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bool is_dumping_aec_ = false;
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bool initialized_ = false;
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// Cache experimental_ns and apply in case they are missing in the audio
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// options. We need to do this because SetExtraOptions() will revert to
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// defaults for options which are not provided.
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absl::optional<bool> experimental_ns_;
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// Jitter buffer settings for new streams.
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size_t audio_jitter_buffer_max_packets_ = 200;
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bool audio_jitter_buffer_fast_accelerate_ = false;
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int audio_jitter_buffer_min_delay_ms_ = 0;
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bool audio_jitter_buffer_enable_rtx_handling_ = false;
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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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const webrtc::CryptoOptions& crypto_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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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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webrtc::RTCError SetRtpSendParameters(
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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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webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
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void SetPlayout(bool playout) override;
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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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void ResetUnsignaledRecvStream() override;
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// E2EE Frame API
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// Set a frame decryptor to a particular ssrc that will intercept all
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// incoming audio payloads and attempt to decrypt them before forwarding the
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// result.
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void SetFrameDecryptor(uint32_t ssrc,
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rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
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frame_decryptor) override;
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// Set a frame encryptor to a particular ssrc that will intercept all
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// outgoing audio payloads frames and attempt to encrypt them and forward the
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// result to the packetizer.
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void SetFrameEncryptor(uint32_t ssrc,
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rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
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frame_encryptor) override;
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bool SetOutputVolume(uint32_t ssrc, double volume) override;
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// Applies the new volume to current and future unsignaled streams.
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bool SetDefaultOutputVolume(double volume) override;
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bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
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absl::optional<int> GetBaseMinimumPlayoutDelayMs(
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uint32_t ssrc) const 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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int64_t packet_time_us) 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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// Set the audio sink for an existing stream.
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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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// Will set the audio sink on the latest unsignaled stream, future or
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// current. Only one stream at a time will use the sink.
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void SetDefaultRawAudioSink(
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std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
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std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
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// Sets a frame transformer between encoder and packetizer, to transform
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// encoded frames before sending them out the network.
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void SetEncoderToPacketizerFrameTransformer(
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uint32_t ssrc,
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rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
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override;
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void SetDepacketizerToDecoderFrameTransformer(
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uint32_t ssrc,
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rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
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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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if (DscpEnabled()) {
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rtc_options.dscp = PreferredDscp();
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}
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rtc_options.info_signaled_after_sent.included_in_feedback =
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options.included_in_feedback;
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rtc_options.info_signaled_after_sent.included_in_allocation =
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options.included_in_allocation;
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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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rtc::PacketOptions rtc_options;
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if (DscpEnabled()) {
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rtc_options.dscp = PreferredDscp();
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}
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return VoiceMediaChannel::SendRtcp(&packet, rtc_options);
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}
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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 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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void ChangePlayout(bool playout);
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int CreateVoEChannel();
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bool DeleteVoEChannel(int channel);
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bool SetMaxSendBitrate(int bps);
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void SetupRecording();
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// Check if 'ssrc' is an unsignaled stream, and if so mark it as not being
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// unsignaled anymore (i.e. it is now removed, or signaled), and return true.
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bool MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc);
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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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// TODO(kwiberg): decoder_map_ and recv_codecs_ store the exact same
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// information, in slightly different formats. Eliminate recv_codecs_.
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std::map<int, webrtc::SdpAudioFormat> decoder_map_;
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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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absl::optional<int> dtmf_payload_type_;
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int dtmf_payload_freq_ = -1;
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bool recv_transport_cc_enabled_ = false;
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bool recv_nack_enabled_ = false;
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bool desired_playout_ = 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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const MediaConfig::Audio audio_config_;
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// Queue of unsignaled SSRCs; oldest at the beginning.
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std::vector<uint32_t> unsignaled_recv_ssrcs_;
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// This is a stream param that comes from the remote description, but wasn't
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// signaled with any a=ssrc lines. It holds the information that was signaled
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// before the unsignaled receive stream is created when the first packet is
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// received.
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StreamParams unsignaled_stream_params_;
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// Volume for unsignaled streams, which may be set before the stream exists.
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double default_recv_volume_ = 1.0;
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// Delay for unsignaled streams, which may be set before the stream exists.
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int default_recv_base_minimum_delay_ms_ = 0;
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// Sink for latest unsignaled stream - 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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std::string mid_;
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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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absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
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send_codec_spec_;
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// TODO(kwiberg): Per-SSRC codec pair IDs?
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const webrtc::AudioCodecPairId codec_pair_id_ =
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webrtc::AudioCodecPairId::Create();
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// Per peer connection crypto options that last for the lifetime of the peer
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// connection.
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const webrtc::CryptoOptions crypto_options_;
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// Unsignaled streams have an option to have a frame decryptor set on them.
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rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
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unsignaled_frame_decryptor_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
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};
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} // namespace cricket
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#endif // MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_
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