Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 345 insertions, 0 deletions

diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine.h b/third_party/libwebrtc/media/engine/webrtc_voice_engine.h
new file mode 100644
index 0000000000..8b62c67449
--- /dev/null
+++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine.h
@@ -0,0 +1,345 @@
+/*
+ *  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.
+ */
+
+#ifndef MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_
+#define MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/field_trials_view.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "call/audio_state.h"
+#include "call/call.h"
+#include "media/base/media_channel_impl.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"
+
+namespace webrtc {
+class AudioFrameProcessor;
+}
+
+namespace cricket {
+
+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;
+
+ public:
+  WebRtcVoiceEngine(
+      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::FieldTrialsView& 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;
+
+  // 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;
+
+  absl::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
+      override;
+
+ private:
+  // 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.
+  void 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_;
+  bool is_dumping_aec_ = false;
+  bool initialized_ = false;
+
+  // 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;
+
+  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 {
+ public:
+  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,
+      webrtc::SetParametersCallback callback) 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;
+  absl::optional<uint32_t> GetUnsignaledSsrc() const 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,
+                         rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             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,
+                         rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
+                             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(const webrtc::RtpPacketReceived& packet) override;
+  void OnPacketSent(const rtc::SentPacket& sent_packet) override;
+  void OnNetworkRouteChanged(absl::string_view transport_name,
+                             const rtc::NetworkRoute& network_route) override;
+  void OnReadyToSend(bool ready) override;
+  bool GetSendStats(VoiceMediaSendInfo* info) override;
+  bool GetReceiveStats(VoiceMediaReceiveInfo* 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)
+      override;
+  void SetDepacketizerToDecoderFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+
+  // implements Transport interface
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override;
+
+  bool SendRtcp(const uint8_t* data, size_t len) override;
+
+ private:
+  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();
+
+  // Expected to be invoked once per packet that belongs to this channel that
+  // can not be demuxed. Returns true if a default receive stream has been
+  // created.
+  bool MaybeCreateDefaultReceiveStream(const webrtc::RtpPacketReceived& packet);
+  // 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_nack_enabled_ = false;
+  bool enable_non_sender_rtt_ = 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: https://code.google.com/p/webrtc/issues/detail?id=4740
+  // and https://code.google.com/p/chromium/issues/detail?id=547661
+  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_;
+  webrtc::RtpHeaderExtensionMap recv_rtp_extension_map_;
+
+  absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
+      send_codec_spec_;
+
+  // TODO(kwiberg): Per-SSRC codec pair IDs?
+  const webrtc::AudioCodecPairId codec_pair_id_ =
+      webrtc::AudioCodecPairId::Create();
+
+  // 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.
+  rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+      unsignaled_frame_decryptor_;
+  rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+      unsignaled_frame_transformer_;
+
+  void FillSendCodecStats(VoiceMediaSendInfo* voice_media_info);
+  void FillReceiveCodecStats(VoiceMediaReceiveInfo* voice_media_info);
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_