blob: 4bc61d540775ad5540fcd7d29ee65019b0b99ebb [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 "api/audio_codecs/audio_encoder_factory.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/transport/webrtc_key_value_config.h"
#include "call/audio_state.h"
#include "call/call.h"
#include "media/base/media_engine.h"
#include "media/base/rtp_utils.h"
#include "modules/async_audio_processing/async_audio_processing.h"
#include "rtc_base/buffer.h"
#include "rtc_base/network_route.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
namespace webrtc {
class AudioFrameProcessor;
namespace cricket {
class AudioDeviceModule;
class AudioMixer;
class AudioSource;
class WebRtcVoiceMediaChannel;
// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
// It uses the WebRtc VoiceEngine library for audio handling.
class WebRtcVoiceEngine final : public VoiceEngineInterface {
friend class WebRtcVoiceMediaChannel;
webrtc::TaskQueueFactory* task_queue_factory,
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,
webrtc::AudioFrameProcessor* audio_frame_processor,
const webrtc::WebRtcKeyValueConfig& trials);
WebRtcVoiceEngine() = delete;
WebRtcVoiceEngine(const WebRtcVoiceEngine&) = delete;
WebRtcVoiceEngine& operator=(const WebRtcVoiceEngine&) = delete;
~WebRtcVoiceEngine() override;
// Does initialization that needs to occur on the worker thread.
void Init() override;
rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
VoiceMediaChannel* CreateMediaChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options) override;
const std::vector<AudioCodec>& send_codecs() const override;
const std::vector<AudioCodec>& recv_codecs() const override;
std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
const override;
// 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);
// 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(webrtc::FileWrapper file, int64_t max_size_bytes) override;
// Stops AEC dump.
void StopAecDump() override;
// 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);
int CreateVoEChannel();
webrtc::TaskQueueFactory* const task_queue_factory_;
std::unique_ptr<rtc::TaskQueue> low_priority_worker_queue_;
webrtc::AudioDeviceModule* adm();
webrtc::AudioProcessing* apm() const;
webrtc::AudioState* audio_state();
std::vector<AudioCodec> CollectCodecs(
const std::vector<webrtc::AudioCodecSpec>& specs) const;
webrtc::SequenceChecker signal_thread_checker_;
webrtc::SequenceChecker worker_thread_checker_;
// The audio device module.
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_;
// The audio processing module.
rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
// Asynchronous audio processing.
webrtc::AudioFrameProcessor* const audio_frame_processor_;
// The primary instance of WebRtc VoiceEngine.
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::vector<AudioCodec> send_codecs_;
std::vector<AudioCodec> recv_codecs_;
std::vector<WebRtcVoiceMediaChannel*> channels_;
bool is_dumping_aec_ = false;
bool initialized_ = false;
// Cache experimental_ns and apply in case they are missing in the audio
// options.
absl::optional<bool> experimental_ns_;
// Jitter buffer settings for new streams.
size_t audio_jitter_buffer_max_packets_ = 200;
bool audio_jitter_buffer_fast_accelerate_ = false;
int audio_jitter_buffer_min_delay_ms_ = 0;
bool audio_jitter_buffer_enable_rtx_handling_ = false;
// If this field trial is enabled, we will negotiate and use RFC 2198
// redundancy for opus audio.
const bool audio_red_for_opus_trial_enabled_;
const bool minimized_remsampling_on_mobile_trial_enabled_;
// 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,
const webrtc::CryptoOptions& crypto_options,
webrtc::Call* call);
WebRtcVoiceMediaChannel() = delete;
WebRtcVoiceMediaChannel(const WebRtcVoiceMediaChannel&) = delete;
WebRtcVoiceMediaChannel& operator=(const WebRtcVoiceMediaChannel&) = delete;
~WebRtcVoiceMediaChannel() override;
const AudioOptions& options() const { return options_; }
bool SetSendParameters(const AudioSendParameters& params) override;
bool SetRecvParameters(const AudioRecvParameters& params) override;
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) override;
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
webrtc::RtpParameters GetDefaultRtpReceiveParameters() const 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;
void ResetUnsignaledRecvStream() override;
void OnDemuxerCriteriaUpdatePending() override;
void OnDemuxerCriteriaUpdateComplete() override;
// E2EE Frame API
// Set a frame decryptor to a particular ssrc that will intercept all
// incoming audio payloads and attempt to decrypt them before forwarding the
// result.
void SetFrameDecryptor(uint32_t ssrc,
frame_decryptor) override;
// Set a frame encryptor to a particular ssrc that will intercept all
// outgoing audio payloads frames and attempt to encrypt them and forward the
// result to the packetizer.
void SetFrameEncryptor(uint32_t ssrc,
frame_encryptor) override;
bool SetOutputVolume(uint32_t ssrc, double volume) override;
// Applies the new volume to current and future unsignaled streams.
bool SetDefaultOutputVolume(double volume) override;
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
absl::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const override;
bool CanInsertDtmf() override;
bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
void OnReadyToSend(bool ready) override;
bool GetStats(VoiceMediaInfo* info, bool get_and_clear_legacy_stats) override;
// Set the audio sink for an existing stream.
void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
// Will set the audio sink on the latest unsignaled stream, future or
// current. Only one stream at a time will use the sink.
void SetDefaultRawAudioSink(
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
// Sets a frame transformer between encoder and packetizer, to transform
// encoded frames before sending them out the network.
void SetEncoderToPacketizerFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
void SetDepacketizerToDecoderFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
// 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;
if (DscpEnabled()) {
rtc_options.dscp = PreferredDscp();
rtc_options.info_signaled_after_sent.included_in_feedback =
rtc_options.info_signaled_after_sent.included_in_allocation =
return VoiceMediaChannel::SendPacket(&packet, rtc_options);
bool SendRtcp(const uint8_t* data, size_t len) override {
rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
rtc::PacketOptions rtc_options;
if (DscpEnabled()) {
rtc_options.dscp = PreferredDscp();
return VoiceMediaChannel::SendRtcp(&packet, rtc_options);
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 CreateVoEChannel();
bool DeleteVoEChannel(int channel);
bool SetMaxSendBitrate(int bps);
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);
webrtc::TaskQueueBase* const worker_thread_;
webrtc::ScopedTaskSafety task_safety_;
webrtc::SequenceChecker network_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_;
absl::optional<int> dtmf_payload_type_;
int dtmf_payload_freq_ = -1;
bool recv_transport_cc_enabled_ = false;
bool recv_nack_enabled_ = false;
bool playout_ = false;
bool send_ = false;
webrtc::Call* const call_ = nullptr;
const MediaConfig::Audio audio_config_;
// Queue of unsignaled SSRCs; oldest at the beginning.
std::vector<uint32_t> unsignaled_recv_ssrcs_;
// This is a stream param that comes from the remote description, but wasn't
// signaled with any a=ssrc lines. It holds the information that was signaled
// before the unsignaled receive stream is created when the first packet is
// received.
StreamParams unsignaled_stream_params_;
// Volume for unsignaled streams, which may be set before the stream exists.
double default_recv_volume_ = 1.0;
// Delay for unsignaled streams, which may be set before the stream exists.
int default_recv_base_minimum_delay_ms_ = 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_;
std::string mid_;
class WebRtcAudioReceiveStream;
std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
// TODO(kwiberg): Per-SSRC codec pair IDs?
const webrtc::AudioCodecPairId codec_pair_id_ =
// Per peer connection crypto options that last for the lifetime of the peer
// connection.
const webrtc::CryptoOptions crypto_options_;
// Unsignaled streams have an option to have a frame decryptor set on them.
const bool audio_red_for_opus_trial_enabled_;
} // namespace cricket