blob: a42459ab795787f010aaac2b94491403aebff62b [file] [log] [blame]
* Copyright (c) 2004 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 <map>
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/call/audio_state.h"
#include "webrtc/call/call.h"
#include "webrtc/media/base/rtputils.h"
#include "webrtc/media/engine/apm_helpers.h"
#include "webrtc/media/engine/webrtccommon.h"
#include "webrtc/media/engine/webrtcvoe.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/pc/channel.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/networkroute.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "webrtc/rtc_base/task_queue.h"
#include "webrtc/rtc_base/thread_checker.h"
namespace webrtc {
namespace voe {
class TransmitMixer;
} // namespace voe
} // namespace webrtc
namespace cricket {
class AudioDeviceModule;
class AudioMixer;
class AudioSource;
class VoEWrapper;
class WebRtcVoiceMediaChannel;
// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
// It uses the WebRtc VoiceEngine library for audio handling.
class WebRtcVoiceEngine final : public webrtc::TraceCallback {
friend class WebRtcVoiceMediaChannel;
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
// Dependency injection for testing.
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
VoEWrapper* voe_wrapper);
~WebRtcVoiceEngine() override;
// Does initialization that needs to occur on the worker thread.
void Init();
rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options);
int GetInputLevel();
const std::vector<AudioCodec>& send_codecs() const;
const std::vector<AudioCodec>& recv_codecs() const;
RtpCapabilities GetCapabilities() const;
// For tracking WebRtc channels. Needed because we have to pause them
// all when switching devices.
// May only be called by WebRtcVoiceMediaChannel.
void RegisterChannel(WebRtcVoiceMediaChannel* channel);
void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
VoEWrapper* voe() { return voe_wrapper_.get(); }
int GetLastEngineError();
// Starts AEC dump using an existing file. A maximum file size in bytes can be
// specified. When the maximum file size is reached, logging is stopped and
// the file is closed. If max_size_bytes is set to <= 0, no limit will be
// used.
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
// Stops AEC dump.
void StopAecDump();
const webrtc::AudioProcessing::Config GetApmConfigForTest() const {
return apm()->GetConfig();
// Every option that is "set" will be applied. Every option not "set" will be
// ignored. This allows us to selectively turn on and off different options
// easily at any time.
bool ApplyOptions(const AudioOptions& options);
// webrtc::TraceCallback:
void Print(webrtc::TraceLevel level, const char* trace, int length) override;
void StartAecDump(const std::string& filename);
int CreateVoEChannel();
std::unique_ptr<rtc::TaskQueue> low_priority_worker_queue_;
webrtc::AudioDeviceModule* adm();
webrtc::AudioProcessing* apm() const;
webrtc::voe::TransmitMixer* transmit_mixer();
AudioCodecs CollectCodecs(
const std::vector<webrtc::AudioCodecSpec>& specs) const;
rtc::ThreadChecker signal_thread_checker_;
rtc::ThreadChecker worker_thread_checker_;
// The audio device manager.
rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_;
rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer_;
// Reference to the APM, owned by VoE.
rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
// Reference to the TransmitMixer, owned by VoE.
webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
// The primary instance of WebRtc VoiceEngine.
std::unique_ptr<VoEWrapper> voe_wrapper_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::vector<AudioCodec> send_codecs_;
std::vector<AudioCodec> recv_codecs_;
std::vector<WebRtcVoiceMediaChannel*> channels_;
webrtc::VoEBase::ChannelConfig channel_config_;
bool is_dumping_aec_ = false;
bool initialized_ = false;
webrtc::AgcConfig default_agc_config_;
// Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns
// level controller, and intelligibility_enhancer values, and apply them
// in case they are missing in the audio options. We need to do this because
// SetExtraOptions() will revert to defaults for options which are not
// provided.
rtc::Optional<bool> extended_filter_aec_;
rtc::Optional<bool> delay_agnostic_aec_;
rtc::Optional<bool> experimental_ns_;
rtc::Optional<bool> intelligibility_enhancer_;
rtc::Optional<bool> level_control_;
// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
// WebRtc Voice Engine.
class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
public webrtc::Transport {
WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
const MediaConfig& config,
const AudioOptions& options,
webrtc::Call* call);
~WebRtcVoiceMediaChannel() override;
const AudioOptions& options() const { return options_; }
rtc::DiffServCodePoint PreferredDscp() const override;
bool SetSendParameters(const AudioSendParameters& params) override;
bool SetRecvParameters(const AudioRecvParameters& params) override;
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
bool SetRtpSendParameters(uint32_t ssrc,
const webrtc::RtpParameters& parameters) override;
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
bool SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) override;
void SetPlayout(bool playout) override;
void SetSend(bool send) override;
bool SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) override;
bool AddSendStream(const StreamParams& sp) override;
bool RemoveSendStream(uint32_t ssrc) override;
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
bool GetActiveStreams(AudioInfo::StreamList* actives) override;
int GetOutputLevel() override;
// SSRC=0 will apply the new volume to current and future unsignaled streams.
bool SetOutputVolume(uint32_t ssrc, double volume) override;
bool CanInsertDtmf() override;
bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) override;
void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
void OnReadyToSend(bool ready) override;
void OnTransportOverheadChanged(int transport_overhead_per_packet) override;
bool GetStats(VoiceMediaInfo* info) override;
// SSRC=0 will set the audio sink on the latest unsignaled stream, future or
// current. Only one stream at a time will use the sink.
void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
// implements Transport interface
bool SendRtp(const uint8_t* data,
size_t len,
const webrtc::PacketOptions& options) override {
rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
rtc::PacketOptions rtc_options;
rtc_options.packet_id = options.packet_id;
return VoiceMediaChannel::SendPacket(&packet, rtc_options);
bool SendRtcp(const uint8_t* data, size_t len) override {
rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
int GetReceiveChannelId(uint32_t ssrc) const;
int GetSendChannelId(uint32_t ssrc) const;
bool SetOptions(const AudioOptions& options);
bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
bool SetLocalSource(uint32_t ssrc, AudioSource* source);
bool MuteStream(uint32_t ssrc, bool mute);
WebRtcVoiceEngine* engine() { return engine_; }
int GetLastEngineError() { return engine()->GetLastEngineError(); }
void ChangePlayout(bool playout);
int CreateVoEChannel();
bool DeleteVoEChannel(int channel);
bool SetMaxSendBitrate(int bps);
bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
void SetupRecording();
// Check if 'ssrc' is an unsignaled stream, and if so mark it as not being
// unsignaled anymore (i.e. it is now removed, or signaled), and return true.
bool MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc);
rtc::ThreadChecker worker_thread_checker_;
WebRtcVoiceEngine* const engine_ = nullptr;
std::vector<AudioCodec> send_codecs_;
// TODO(kwiberg): decoder_map_ and recv_codecs_ store the exact same
// information, in slightly different formats. Eliminate recv_codecs_.
std::map<int, webrtc::SdpAudioFormat> decoder_map_;
std::vector<AudioCodec> recv_codecs_;
int max_send_bitrate_bps_ = 0;
AudioOptions options_;
rtc::Optional<int> dtmf_payload_type_;
int dtmf_payload_freq_ = -1;
bool recv_transport_cc_enabled_ = false;
bool recv_nack_enabled_ = false;
bool desired_playout_ = false;
bool playout_ = false;
bool send_ = false;
webrtc::Call* const call_ = nullptr;
// Queue of unsignaled SSRCs; oldest at the beginning.
std::vector<uint32_t> unsignaled_recv_ssrcs_;
// Volume for unsignaled streams, which may be set before the stream exists.
double default_recv_volume_ = 1.0;
// Sink for latest unsignaled stream - may be set before the stream exists.
std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
// Default SSRC to use for RTCP receiver reports in case of no signaled
// send streams. See:
// and
uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
class WebRtcAudioSendStream;
std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
std::vector<webrtc::RtpExtension> send_rtp_extensions_;
class WebRtcAudioReceiveStream;
std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
} // namespace cricket