Move remaining traces of VoiceEngine

- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
  utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.

NOPRESUBMIT=true

Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}

1 files changed, 145 insertions, 0 deletions

diff --git a/audio/channel_proxy.h b/audio/channel_proxy.h
new file mode 100644
index 0000000000..f5f603ba75
--- /dev/null
+++ b/audio/channel_proxy.h
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2015 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 AUDIO_CHANNEL_PROXY_H_
+#define AUDIO_CHANNEL_PROXY_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/rtpreceiverinterface.h"
+#include "audio/channel.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/thread_checker.h"
+
+namespace webrtc {
+
+class AudioSinkInterface;
+class PacketRouter;
+class RtcEventLog;
+class RtcpBandwidthObserver;
+class RtcpRttStats;
+class RtpPacketSender;
+class RtpPacketReceived;
+class RtpReceiver;
+class RtpRtcp;
+class RtpTransportControllerSendInterface;
+class Transport;
+class TransportFeedbackObserver;
+
+namespace voe {
+
+// This class provides the "view" of a voe::Channel that we need to implement
+// webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
+// purposes:
+//  1. Allow mocking just the interfaces used, instead of the entire
+//     voe::Channel class.
+//  2. Provide a refined interface for the stream classes, including assumptions
+//     on return values and input adaptation.
+class ChannelProxy : public RtpPacketSinkInterface {
+ public:
+  ChannelProxy();
+  explicit ChannelProxy(std::unique_ptr<Channel> channel);
+  virtual ~ChannelProxy();
+
+  virtual bool SetEncoder(int payload_type,
+                          std::unique_ptr<AudioEncoder> encoder);
+  virtual void ModifyEncoder(
+      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
+
+  virtual void SetRTCPStatus(bool enable);
+  virtual void SetLocalSSRC(uint32_t ssrc);
+  virtual void SetRTCP_CNAME(const std::string& c_name);
+  virtual void SetNACKStatus(bool enable, int max_packets);
+  virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
+  virtual void EnableSendTransportSequenceNumber(int id);
+  virtual void RegisterSenderCongestionControlObjects(
+      RtpTransportControllerSendInterface* transport,
+      RtcpBandwidthObserver* bandwidth_observer);
+  virtual void RegisterReceiverCongestionControlObjects(
+      PacketRouter* packet_router);
+  virtual void ResetSenderCongestionControlObjects();
+  virtual void ResetReceiverCongestionControlObjects();
+  virtual CallStatistics GetRTCPStatistics() const;
+  virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
+  virtual NetworkStatistics GetNetworkStatistics() const;
+  virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
+  virtual ANAStats GetANAStatistics() const;
+  virtual int GetSpeechOutputLevel() const;
+  virtual int GetSpeechOutputLevelFullRange() const;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec:
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+  virtual double GetTotalOutputEnergy() const;
+  virtual double GetTotalOutputDuration() const;
+  virtual uint32_t GetDelayEstimate() const;
+  virtual bool SetSendTelephoneEventPayloadType(int payload_type,
+                                                int payload_frequency);
+  virtual bool SendTelephoneEventOutband(int event, int duration_ms);
+  virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
+  virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
+  virtual void SetSink(AudioSinkInterface* sink);
+  virtual void SetInputMute(bool muted);
+  virtual void RegisterTransport(Transport* transport);
+
+  // Implements RtpPacketSinkInterface
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+  virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
+  virtual void SetChannelOutputVolumeScaling(float scaling);
+  virtual void SetRtcEventLog(RtcEventLog* event_log);
+  virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
+      int sample_rate_hz,
+      AudioFrame* audio_frame);
+  virtual int PreferredSampleRate() const;
+  virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame);
+  virtual void SetTransportOverhead(int transport_overhead_per_packet);
+  virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
+  virtual void DisassociateSendChannel();
+  virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp,
+                          RtpReceiver** rtp_receiver) const;
+  virtual uint32_t GetPlayoutTimestamp() const;
+  virtual void SetMinimumPlayoutDelay(int delay_ms);
+  virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
+  virtual bool GetRecCodec(CodecInst* codec_inst) const;
+  virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
+  virtual void OnRecoverableUplinkPacketLossRate(
+      float recoverable_packet_loss_rate);
+  virtual std::vector<webrtc::RtpSource> GetSources() const;
+  virtual void StartSend();
+  virtual void StopSend();
+  virtual void StartPlayout();
+  virtual void StopPlayout();
+
+ private:
+  // Thread checkers document and lock usage of some methods on voe::Channel to
+  // specific threads we know about. The goal is to eventually split up
+  // voe::Channel into parts with single-threaded semantics, and thereby reduce
+  // the need for locks.
+  rtc::ThreadChecker worker_thread_checker_;
+  rtc::ThreadChecker module_process_thread_checker_;
+  // Methods accessed from audio and video threads are checked for sequential-
+  // only access. We don't necessarily own and control these threads, so thread
+  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
+  // audio thread to another, but access is still sequential.
+  rtc::RaceChecker audio_thread_race_checker_;
+  rtc::RaceChecker video_capture_thread_race_checker_;
+  std::unique_ptr<Channel> channel_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
+};
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_CHANNEL_PROXY_H_