Move talk/media to webrtc/media

talk/media/webrtc/webrtcvoiceengine.h

Index: talk/media/webrtc/webrtcvoiceengine.h
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
deleted file mode 100644
index 9f894c93ead83839d763bae7064ff953dd93af89..0000000000000000000000000000000000000000
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ /dev/null
@@ -1,292 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_MEDIA_WEBRTCVOICEENGINE_H_
-#define TALK_MEDIA_WEBRTCVOICEENGINE_H_
-
-#include <map>
-#include <string>
-#include <vector>
-
-#include "talk/media/base/rtputils.h"
-#include "talk/media/webrtc/webrtccommon.h"
-#include "talk/media/webrtc/webrtcvoe.h"
-#include "talk/session/media/channel.h"
-#include "webrtc/audio_state.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/stream.h"
-#include "webrtc/base/thread_checker.h"
-#include "webrtc/call.h"
-#include "webrtc/common.h"
-#include "webrtc/config.h"
-
-namespace cricket {
-
-class AudioDeviceModule;
-class AudioRenderer;
-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;
- public:
-  // Exposed for the WVoE/MC unit test.
-  static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
-
-  WebRtcVoiceEngine();
-  // Dependency injection for testing.
-  explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
-  ~WebRtcVoiceEngine();
-  bool Init(rtc::Thread* worker_thread);
-  void Terminate();
-
-  rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
-  VoiceMediaChannel* CreateChannel(webrtc::Call* call,
-                                   const AudioOptions& options);
-
-  bool GetOutputVolume(int* level);
-  bool SetOutputVolume(int level);
-  int GetInputLevel();
-
-  const std::vector<AudioCodec>& codecs();
-  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);
-
-  // Called by WebRtcVoiceMediaChannel to set a gain offset from
-  // the default AGC target level.
-  bool AdjustAgcLevel(int delta);
-
-  VoEWrapper* voe() { return voe_wrapper_.get(); }
-  int GetLastEngineError();
-
-  // Set the external ADM. This can only be called before Init.
-  bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
-
-  // 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();
-
-  // Starts recording an RtcEventLog using an existing file until 10 minutes
-  // pass or the StopRtcEventLog function is called.
-  bool StartRtcEventLog(rtc::PlatformFile file);
-
-  // Stops recording the RtcEventLog.
-  void StopRtcEventLog();
-
- private:
-  void Construct();
-  bool InitInternal();
-  // 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 SetDefaultDevices();
-
-  // webrtc::TraceCallback:
-  void Print(webrtc::TraceLevel level, const char* trace, int length) override;
-
-  void StartAecDump(const std::string& filename);
-  int CreateVoEChannel();
-
-  rtc::ThreadChecker signal_thread_checker_;
-  rtc::ThreadChecker worker_thread_checker_;
-
-  // The primary instance of WebRtc VoiceEngine.
-  rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
-  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
-  // The external audio device manager
-  webrtc::AudioDeviceModule* adm_ = nullptr;
-  std::vector<AudioCodec> codecs_;
-  std::vector<WebRtcVoiceMediaChannel*> channels_;
-  webrtc::Config voe_config_;
-  bool initialized_ = false;
-  bool is_dumping_aec_ = false;
-
-  webrtc::AgcConfig default_agc_config_;
-  // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
-  // 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_DISALLOW_COPY_AND_ASSIGN(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 AudioOptions& options,
-                          webrtc::Call* call);
-  ~WebRtcVoiceMediaChannel() override;
-
-  const AudioOptions& options() const { return options_; }
-
-  bool SetSendParameters(const AudioSendParameters& params) override;
-  bool SetRecvParameters(const AudioRecvParameters& params) override;
-  bool SetPlayout(bool playout) override;
-  bool PausePlayout();
-  bool ResumePlayout();
-  bool SetSend(SendFlags send) override;
-  bool PauseSend();
-  bool ResumeSend();
-  bool SetAudioSend(uint32_t ssrc,
-                    bool enable,
-                    const AudioOptions* options,
-                    AudioRenderer* renderer) 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;
-  int GetTimeSinceLastTyping() override;
-  void SetTypingDetectionParameters(int time_window,
-                                    int cost_per_typing,
-                                    int reporting_threshold,
-                                    int penalty_decay,
-                                    int type_event_delay) override;
-  bool SetOutputVolume(uint32_t ssrc, double volume) override;
-
-  bool CanInsertDtmf() override;
-  bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
-
-  void OnPacketReceived(rtc::Buffer* packet,
-                        const rtc::PacketTime& packet_time) override;
-  void OnRtcpReceived(rtc::Buffer* packet,
-                      const rtc::PacketTime& packet_time) override;
-  void OnReadyToSend(bool ready) override {}
-  bool GetStats(VoiceMediaInfo* info) override;
-
-  void SetRawAudioSink(
-      uint32_t ssrc,
-      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
-
-  // implements Transport interface
-  bool SendRtp(const uint8_t* data,
-               size_t len,
-               const webrtc::PacketOptions& options) override {
-    rtc::Buffer packet(reinterpret_cast<const uint8_t*>(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::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
-                       kMaxRtpPacketLen);
-    return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
-  }
-
-  int GetReceiveChannelId(uint32_t ssrc) const;
-  int GetSendChannelId(uint32_t ssrc) const;
-
- private:
-  bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
-  bool SetOptions(const AudioOptions& options);
-  bool SetMaxSendBandwidth(int bps);
-  bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
-  bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer);
-  bool MuteStream(uint32_t ssrc, bool mute);
-
-  WebRtcVoiceEngine* engine() { return engine_; }
-  int GetLastEngineError() { return engine()->GetLastEngineError(); }
-  int GetOutputLevel(int channel);
-  bool SetPlayout(int channel, bool playout);
-  void SetNack(int channel, bool nack_enabled);
-  bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
-  bool ChangePlayout(bool playout);
-  bool ChangeSend(SendFlags send);
-  bool ChangeSend(int channel, SendFlags send);
-  int CreateVoEChannel();
-  bool DeleteVoEChannel(int channel);
-  bool IsDefaultRecvStream(uint32_t ssrc) {
-    return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
-  }
-  bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
-  bool SetSendBitrateInternal(int bps);
-
-  rtc::ThreadChecker worker_thread_checker_;
-
-  WebRtcVoiceEngine* const engine_ = nullptr;
-  std::vector<AudioCodec> recv_codecs_;
-  std::vector<AudioCodec> send_codecs_;
-  rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
-  bool send_bitrate_setting_ = false;
-  int send_bitrate_bps_ = 0;
-  AudioOptions options_;
-  rtc::Optional<int> dtmf_payload_type_;
-  bool desired_playout_ = false;
-  bool nack_enabled_ = false;
-  bool transport_cc_enabled_ = false;
-  bool playout_ = false;
-  SendFlags desired_send_ = SEND_NOTHING;
-  SendFlags send_ = SEND_NOTHING;
-  webrtc::Call* const call_ = nullptr;
-
-  // SSRC of unsignalled receive stream, or -1 if there isn't one.
-  int64_t default_recv_ssrc_ = -1;
-  // Volume for unsignalled stream, which may be set before the stream exists.
-  double default_recv_volume_ = 1.0;
-  // Sink for unsignalled stream, which may be set before the stream exists.
-  rtc::scoped_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_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
-};
-}  // namespace cricket
-
-#endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_