Adding upstream version 86.0.1.upstream/86.0.1upstream

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

1 files changed, 304 insertions, 0 deletions

diff --git a/third_party/libwebrtc/webrtc/media/engine/webrtcvoiceengine.h b/third_party/libwebrtc/webrtc/media/engine/webrtcvoiceengine.h
new file mode 100644
index 0000000000..8984299638
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/media/engine/webrtcvoiceengine.h
@@ -0,0 +1,304 @@
+/*
+ *  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_WEBRTCVOICEENGINE_H_
+#define MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/rtpreceiverinterface.h"
+#include "call/audio_state.h"
+#include "call/call.h"
+#include "media/base/rtputils.h"
+#include "media/engine/apm_helpers.h"
+#include "media/engine/webrtcvoe.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "pc/channel.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/networkroute.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "rtc_base/task_queue.h"
+#include "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 {
+  friend class WebRtcVoiceMediaChannel;
+ public:
+  WebRtcVoiceEngine(
+      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.
+  WebRtcVoiceEngine(
+      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();
+
+  // 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(); }
+
+  // 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();
+  }
+
+ 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.
+  bool ApplyOptions(const AudioOptions& options);
+
+  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_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine);
+};
+
+// 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,
+                          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(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;
+  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;
+
+ 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_; }
+  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: 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_;
+
+  class WebRtcAudioReceiveStream;
+  std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
+  std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
+
+  rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>
+      send_codec_spec_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
+};
+}  // namespace cricket
+
+#endif  // MEDIA_ENGINE_WEBRTCVOICEENGINE_H_