Move talk/media to webrtc/media

talk/media/webrtc/fakewebrtcvoiceengine.h

Index: talk/media/webrtc/fakewebrtcvoiceengine.h
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
deleted file mode 100644
index 4c6c016a38af2221c207cbff694cfd7e02bc5b0f..0000000000000000000000000000000000000000
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ /dev/null
@@ -1,825 +0,0 @@
-/*
- * libjingle
- * Copyright 2010 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_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
-#define TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
-
-#include <list>
-#include <map>
-#include <vector>
-
-#include "talk/media/base/codec.h"
-#include "talk/media/base/rtputils.h"
-#include "talk/media/webrtc/fakewebrtccommon.h"
-#include "talk/media/webrtc/webrtcvoe.h"
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/config.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-
-namespace cricket {
-
-static const int kOpusBandwidthNb = 4000;
-static const int kOpusBandwidthMb = 6000;
-static const int kOpusBandwidthWb = 8000;
-static const int kOpusBandwidthSwb = 12000;
-static const int kOpusBandwidthFb = 20000;
-
-#define WEBRTC_CHECK_CHANNEL(channel) \
-  if (channels_.find(channel) == channels_.end()) return -1;
-
-class FakeAudioProcessing : public webrtc::AudioProcessing {
- public:
-  FakeAudioProcessing() : experimental_ns_enabled_(false) {}
-
-  WEBRTC_STUB(Initialize, ())
-  WEBRTC_STUB(Initialize, (
-      int input_sample_rate_hz,
-      int output_sample_rate_hz,
-      int reverse_sample_rate_hz,
-      webrtc::AudioProcessing::ChannelLayout input_layout,
-      webrtc::AudioProcessing::ChannelLayout output_layout,
-      webrtc::AudioProcessing::ChannelLayout reverse_layout));
-  WEBRTC_STUB(Initialize, (
-      const webrtc::ProcessingConfig& processing_config));
-
-  WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
-    experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
-  }
-
-  WEBRTC_STUB_CONST(input_sample_rate_hz, ());
-  WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
-  WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
-  size_t num_input_channels() const override { return 0; }
-  size_t num_proc_channels() const override { return 0; }
-  size_t num_output_channels() const override { return 0; }
-  size_t num_reverse_channels() const override { return 0; }
-  WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
-  WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
-  WEBRTC_STUB(ProcessStream, (
-      const float* const* src,
-      size_t samples_per_channel,
-      int input_sample_rate_hz,
-      webrtc::AudioProcessing::ChannelLayout input_layout,
-      int output_sample_rate_hz,
-      webrtc::AudioProcessing::ChannelLayout output_layout,
-      float* const* dest));
-  WEBRTC_STUB(ProcessStream,
-              (const float* const* src,
-               const webrtc::StreamConfig& input_config,
-               const webrtc::StreamConfig& output_config,
-               float* const* dest));
-  WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame));
-  WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame));
-  WEBRTC_STUB(AnalyzeReverseStream, (
-      const float* const* data,
-      size_t samples_per_channel,
-      int sample_rate_hz,
-      webrtc::AudioProcessing::ChannelLayout layout));
-  WEBRTC_STUB(ProcessReverseStream,
-              (const float* const* src,
-               const webrtc::StreamConfig& reverse_input_config,
-               const webrtc::StreamConfig& reverse_output_config,
-               float* const* dest));
-  WEBRTC_STUB(set_stream_delay_ms, (int delay));
-  WEBRTC_STUB_CONST(stream_delay_ms, ());
-  WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
-  WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
-  WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
-  WEBRTC_STUB_CONST(delay_offset_ms, ());
-  WEBRTC_STUB(StartDebugRecording,
-              (const char filename[kMaxFilenameSize], int64_t max_size_bytes));
-  WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes));
-  WEBRTC_STUB(StopDebugRecording, ());
-  WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
-  webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
-  webrtc::EchoControlMobile* echo_control_mobile() const override {
-    return NULL;
-  }
-  webrtc::GainControl* gain_control() const override { return NULL; }
-  webrtc::HighPassFilter* high_pass_filter() const override { return NULL; }
-  webrtc::LevelEstimator* level_estimator() const override { return NULL; }
-  webrtc::NoiseSuppression* noise_suppression() const override { return NULL; }
-  webrtc::VoiceDetection* voice_detection() const override { return NULL; }
-
-  bool experimental_ns_enabled() {
-    return experimental_ns_enabled_;
-  }
-
- private:
-  bool experimental_ns_enabled_;
-};
-
-class FakeWebRtcVoiceEngine
-    : public webrtc::VoEAudioProcessing,
-      public webrtc::VoEBase, public webrtc::VoECodec,
-      public webrtc::VoEHardware,
-      public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
-      public webrtc::VoEVolumeControl {
- public:
-  struct Channel {
-    explicit Channel()
-        : external_transport(false),
-          send(false),
-          playout(false),
-          volume_scale(1.0),
-          vad(false),
-          codec_fec(false),
-          max_encoding_bandwidth(0),
-          opus_dtx(false),
-          red(false),
-          nack(false),
-          cn8_type(13),
-          cn16_type(105),
-          red_type(117),
-          nack_max_packets(0),
-          send_ssrc(0),
-          associate_send_channel(-1),
-          recv_codecs(),
-          neteq_capacity(-1),
-          neteq_fast_accelerate(false) {
-      memset(&send_codec, 0, sizeof(send_codec));
-    }
-    bool external_transport;
-    bool send;
-    bool playout;
-    float volume_scale;
-    bool vad;
-    bool codec_fec;
-    int max_encoding_bandwidth;
-    bool opus_dtx;
-    bool red;
-    bool nack;
-    int cn8_type;
-    int cn16_type;
-    int red_type;
-    int nack_max_packets;
-    uint32_t send_ssrc;
-    int associate_send_channel;
-    std::vector<webrtc::CodecInst> recv_codecs;
-    webrtc::CodecInst send_codec;
-    webrtc::PacketTime last_rtp_packet_time;
-    std::list<std::string> packets;
-    int neteq_capacity;
-    bool neteq_fast_accelerate;
-  };
-
-  FakeWebRtcVoiceEngine()
-      : inited_(false),
-        last_channel_(-1),
-        fail_create_channel_(false),
-        num_set_send_codecs_(0),
-        ec_enabled_(false),
-        ec_metrics_enabled_(false),
-        cng_enabled_(false),
-        ns_enabled_(false),
-        agc_enabled_(false),
-        highpass_filter_enabled_(false),
-        stereo_swapping_enabled_(false),
-        typing_detection_enabled_(false),
-        ec_mode_(webrtc::kEcDefault),
-        aecm_mode_(webrtc::kAecmSpeakerphone),
-        ns_mode_(webrtc::kNsDefault),
-        agc_mode_(webrtc::kAgcDefault),
-        observer_(NULL),
-        playout_fail_channel_(-1),
-        send_fail_channel_(-1),
-        recording_sample_rate_(-1),
-        playout_sample_rate_(-1) {
-    memset(&agc_config_, 0, sizeof(agc_config_));
-  }
-  ~FakeWebRtcVoiceEngine() {
-    RTC_CHECK(channels_.empty());
-  }
-
-  bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
-
-  bool IsInited() const { return inited_; }
-  int GetLastChannel() const { return last_channel_; }
-  int GetNumChannels() const { return static_cast<int>(channels_.size()); }
-  uint32_t GetLocalSSRC(int channel) {
-    return channels_[channel]->send_ssrc;
-  }
-  bool GetPlayout(int channel) {
-    return channels_[channel]->playout;
-  }
-  bool GetSend(int channel) {
-    return channels_[channel]->send;
-  }
-  bool GetVAD(int channel) {
-    return channels_[channel]->vad;
-  }
-  bool GetOpusDtx(int channel) {
-    return channels_[channel]->opus_dtx;
-  }
-  bool GetRED(int channel) {
-    return channels_[channel]->red;
-  }
-  bool GetCodecFEC(int channel) {
-    return channels_[channel]->codec_fec;
-  }
-  int GetMaxEncodingBandwidth(int channel) {
-    return channels_[channel]->max_encoding_bandwidth;
-  }
-  bool GetNACK(int channel) {
-    return channels_[channel]->nack;
-  }
-  int GetNACKMaxPackets(int channel) {
-    return channels_[channel]->nack_max_packets;
-  }
-  const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
-    RTC_DCHECK(channels_.find(channel) != channels_.end());
-    return channels_[channel]->last_rtp_packet_time;
-  }
-  int GetSendCNPayloadType(int channel, bool wideband) {
-    return (wideband) ?
-        channels_[channel]->cn16_type :
-        channels_[channel]->cn8_type;
-  }
-  int GetSendREDPayloadType(int channel) {
-    return channels_[channel]->red_type;
-  }
-  bool CheckPacket(int channel, const void* data, size_t len) {
-    bool result = !CheckNoPacket(channel);
-    if (result) {
-      std::string packet = channels_[channel]->packets.front();
-      result = (packet == std::string(static_cast<const char*>(data), len));
-      channels_[channel]->packets.pop_front();
-    }
-    return result;
-  }
-  bool CheckNoPacket(int channel) {
-    return channels_[channel]->packets.empty();
-  }
-  void TriggerCallbackOnError(int channel_num, int err_code) {
-    RTC_DCHECK(observer_ != NULL);
-    observer_->CallbackOnError(channel_num, err_code);
-  }
-  void set_playout_fail_channel(int channel) {
-    playout_fail_channel_ = channel;
-  }
-  void set_send_fail_channel(int channel) {
-    send_fail_channel_ = channel;
-  }
-  void set_fail_create_channel(bool fail_create_channel) {
-    fail_create_channel_ = fail_create_channel;
-  }
-  int AddChannel(const webrtc::Config& config) {
-    if (fail_create_channel_) {
-      return -1;
-    }
-    Channel* ch = new Channel();
-    auto db = webrtc::acm2::RentACodec::Database();
-    ch->recv_codecs.assign(db.begin(), db.end());
-    if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
-      ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
-    }
-    ch->neteq_fast_accelerate =
-        config.Get<webrtc::NetEqFastAccelerate>().enabled;
-    channels_[++last_channel_] = ch;
-    return last_channel_;
-  }
-
-  int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
-
-  int GetAssociateSendChannel(int channel) {
-    return channels_[channel]->associate_send_channel;
-  }
-
-  WEBRTC_STUB(Release, ());
-
-  // webrtc::VoEBase
-  WEBRTC_FUNC(RegisterVoiceEngineObserver, (
-      webrtc::VoiceEngineObserver& observer)) {
-    observer_ = &observer;
-    return 0;
-  }
-  WEBRTC_STUB(DeRegisterVoiceEngineObserver, ());
-  WEBRTC_FUNC(Init, (webrtc::AudioDeviceModule* adm,
-                     webrtc::AudioProcessing* audioproc)) {
-    inited_ = true;
-    return 0;
-  }
-  WEBRTC_FUNC(Terminate, ()) {
-    inited_ = false;
-    return 0;
-  }
-  webrtc::AudioProcessing* audio_processing() override {
-    return &audio_processing_;
-  }
-  WEBRTC_FUNC(CreateChannel, ()) {
-    webrtc::Config empty_config;
-    return AddChannel(empty_config);
-  }
-  WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
-    return AddChannel(config);
-  }
-  WEBRTC_FUNC(DeleteChannel, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    for (const auto& ch : channels_) {
-      if (ch.second->associate_send_channel == channel) {
-        ch.second->associate_send_channel = -1;
-      }
-    }
-    delete channels_[channel];
-    channels_.erase(channel);
-    return 0;
-  }
-  WEBRTC_STUB(StartReceive, (int channel));
-  WEBRTC_FUNC(StartPlayout, (int channel)) {
-    if (playout_fail_channel_ != channel) {
-      WEBRTC_CHECK_CHANNEL(channel);
-      channels_[channel]->playout = true;
-      return 0;
-    } else {
-      // When playout_fail_channel_ == channel, fail the StartPlayout on this
-      // channel.
-      return -1;
-    }
-  }
-  WEBRTC_FUNC(StartSend, (int channel)) {
-    if (send_fail_channel_ != channel) {
-      WEBRTC_CHECK_CHANNEL(channel);
-      channels_[channel]->send = true;
-      return 0;
-    } else {
-      // When send_fail_channel_ == channel, fail the StartSend on this
-      // channel.
-      return -1;
-    }
-  }
-  WEBRTC_STUB(StopReceive, (int channel));
-  WEBRTC_FUNC(StopPlayout, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->playout = false;
-    return 0;
-  }
-  WEBRTC_FUNC(StopSend, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->send = false;
-    return 0;
-  }
-  WEBRTC_STUB(GetVersion, (char version[1024]));
-  WEBRTC_STUB(LastError, ());
-  WEBRTC_FUNC(AssociateSendChannel, (int channel,
-                                     int accociate_send_channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->associate_send_channel = accociate_send_channel;
-    return 0;
-  }
-  webrtc::RtcEventLog* GetEventLog() { return nullptr; }
-
-  // webrtc::VoECodec
-  WEBRTC_STUB(NumOfCodecs, ());
-  WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
-  WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    // To match the behavior of the real implementation.
-    if (_stricmp(codec.plname, "telephone-event") == 0 ||
-        _stricmp(codec.plname, "audio/telephone-event") == 0 ||
-        _stricmp(codec.plname, "CN") == 0 ||
-        _stricmp(codec.plname, "red") == 0 ) {
-      return -1;
-    }
-    channels_[channel]->send_codec = codec;
-    ++num_set_send_codecs_;
-    return 0;
-  }
-  WEBRTC_FUNC(GetSendCodec, (int channel, webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    codec = channels_[channel]->send_codec;
-    return 0;
-  }
-  WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
-  WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
-  WEBRTC_FUNC(SetRecPayloadType, (int channel,
-                                  const webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    Channel* ch = channels_[channel];
-    if (ch->playout)
-      return -1;  // Channel is in use.
-    // Check if something else already has this slot.
-    if (codec.pltype != -1) {
-      for (std::vector<webrtc::CodecInst>::iterator it =
-          ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) {
-        if (it->pltype == codec.pltype &&
-            _stricmp(it->plname, codec.plname) != 0) {
-          return -1;
-        }
-      }
-    }
-    // Otherwise try to find this codec and update its payload type.
-    int result = -1;  // not found
-    for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
-         it != ch->recv_codecs.end(); ++it) {
-      if (strcmp(it->plname, codec.plname) == 0 &&
-          it->plfreq == codec.plfreq &&
-          it->channels == codec.channels) {
-        it->pltype = codec.pltype;
-        result = 0;
-      }
-    }
-    return result;
-  }
-  WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type,
-                                     webrtc::PayloadFrequencies frequency)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (frequency == webrtc::kFreq8000Hz) {
-      channels_[channel]->cn8_type = type;
-    } else if (frequency == webrtc::kFreq16000Hz) {
-      channels_[channel]->cn16_type = type;
-    }
-    return 0;
-  }
-  WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    Channel* ch = channels_[channel];
-    for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
-         it != ch->recv_codecs.end(); ++it) {
-      if (strcmp(it->plname, codec.plname) == 0 &&
-          it->plfreq == codec.plfreq &&
-          it->channels == codec.channels &&
-          it->pltype != -1) {
-        codec.pltype = it->pltype;
-        return 0;
-      }
-    }
-    return -1;  // not found
-  }
-  WEBRTC_FUNC(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode,
-                             bool disableDTX)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (channels_[channel]->send_codec.channels == 2) {
-      // Replicating VoE behavior; VAD cannot be enabled for stereo.
-      return -1;
-    }
-    channels_[channel]->vad = enable;
-    return 0;
-  }
-  WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
-                             webrtc::VadModes& mode, bool& disabledDTX));
-
-  WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
-      // Return -1 if current send codec is not Opus.
-      // TODO(minyue): Excludes other codecs if they support inband FEC.
-      return -1;
-    }
-    channels_[channel]->codec_fec = enable;
-    return 0;
-  }
-  WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    enable = channels_[channel]->codec_fec;
-    return 0;
-  }
-
-  WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
-      // Return -1 if current send codec is not Opus.
-      return -1;
-    }
-    if (frequency_hz <= 8000)
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb;
-    else if (frequency_hz <= 12000)
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb;
-    else if (frequency_hz <= 16000)
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb;
-    else if (frequency_hz <= 24000)
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb;
-    else
-      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb;
-    return 0;
-  }
-
-  WEBRTC_FUNC(SetOpusDtx, (int channel, bool enable_dtx)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
-      // Return -1 if current send codec is not Opus.
-      return -1;
-    }
-    channels_[channel]->opus_dtx = enable_dtx;
-    return 0;
-  }
-
-  // webrtc::VoEHardware
-  WEBRTC_STUB(GetNumOfRecordingDevices, (int& num));
-  WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num));
-  WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid));
-  WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
-  WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
-  WEBRTC_STUB(SetPlayoutDevice, (int));
-  WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
-  WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
-  WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) {
-    recording_sample_rate_ = samples_per_sec;
-    return 0;
-  }
-  WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) {
-    *samples_per_sec = recording_sample_rate_;
-    return 0;
-  }
-  WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) {
-    playout_sample_rate_ = samples_per_sec;
-    return 0;
-  }
-  WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) {
-    *samples_per_sec = playout_sample_rate_;
-    return 0;
-  }
-  WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
-  virtual bool BuiltInAECIsAvailable() const { return false; }
-  WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
-  virtual bool BuiltInAGCIsAvailable() const { return false; }
-  WEBRTC_STUB(EnableBuiltInNS, (bool enable));
-  virtual bool BuiltInNSIsAvailable() const { return false; }
-
-  // webrtc::VoENetwork
-  WEBRTC_FUNC(RegisterExternalTransport, (int channel,
-                                          webrtc::Transport& transport)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->external_transport = true;
-    return 0;
-  }
-  WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->external_transport = false;
-    return 0;
-  }
-  WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
-                                  size_t length)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (!channels_[channel]->external_transport) return -1;
-    channels_[channel]->packets.push_back(
-        std::string(static_cast<const char*>(data), length));
-    return 0;
-  }
-  WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
-                                  size_t length,
-                                  const webrtc::PacketTime& packet_time)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (ReceivedRTPPacket(channel, data, length) == -1) {
-      return -1;
-    }
-    channels_[channel]->last_rtp_packet_time = packet_time;
-    return 0;
-  }
-
-  WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
-                                   size_t length));
-
-  // webrtc::VoERTP_RTCP
-  WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->send_ssrc = ssrc;
-    return 0;
-  }
-  WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc));
-  WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
-  WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
-      unsigned char id));
-  WEBRTC_STUB(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
-      unsigned char id));
-  WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
-      unsigned char id));
-  WEBRTC_STUB(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable,
-      unsigned char id));
-  WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable));
-  WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled));
-  WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256]));
-  WEBRTC_STUB(GetRTCP_CNAME, (int channel, char cname[256]));
-  WEBRTC_STUB(GetRemoteRTCP_CNAME, (int channel, char* cname));
-  WEBRTC_STUB(GetRemoteRTCPData, (int channel, unsigned int& NTPHigh,
-                                  unsigned int& NTPLow,
-                                  unsigned int& timestamp,
-                                  unsigned int& playoutTimestamp,
-                                  unsigned int* jitter,
-                                  unsigned short* fractionLost));
-  WEBRTC_STUB(GetRemoteRTCPReportBlocks,
-              (int channel, std::vector<webrtc::ReportBlock>* receive_blocks));
-  WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs,
-                                 unsigned int& maxJitterMs,
-                                 unsigned int& discardedPackets));
-  WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats));
-  WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->red = enable;
-    channels_[channel]->red_type = redPayloadtype;
-    return 0;
-  }
-  WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    enable = channels_[channel]->red;
-    redPayloadtype = channels_[channel]->red_type;
-    return 0;
-  }
-  WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->nack = enable;
-    channels_[channel]->nack_max_packets = maxNoPackets;
-    return 0;
-  }
-
-  // webrtc::VoEVolumeControl
-  WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
-  WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
-  WEBRTC_STUB(SetMicVolume, (unsigned int));
-  WEBRTC_STUB(GetMicVolume, (unsigned int&));
-  WEBRTC_STUB(SetInputMute, (int, bool));
-  WEBRTC_STUB(GetInputMute, (int, bool&));
-  WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&));
-  WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&));
-  WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&));
-  WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&));
-  WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->volume_scale= scale;
-    return 0;
-  }
-  WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    scale = channels_[channel]->volume_scale;
-    return 0;
-  }
-  WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right));
-  WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right));
-
-  // webrtc::VoEAudioProcessing
-  WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) {
-    ns_enabled_ = enable;
-    ns_mode_ = mode;
-    return 0;
-  }
-  WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) {
-    enabled = ns_enabled_;
-    mode = ns_mode_;
-    return 0;
-  }
-
-  WEBRTC_FUNC(SetAgcStatus, (bool enable, webrtc::AgcModes mode)) {
-    agc_enabled_ = enable;
-    agc_mode_ = mode;
-    return 0;
-  }
-  WEBRTC_FUNC(GetAgcStatus, (bool& enabled, webrtc::AgcModes& mode)) {
-    enabled = agc_enabled_;
-    mode = agc_mode_;
-    return 0;
-  }
-
-  WEBRTC_FUNC(SetAgcConfig, (webrtc::AgcConfig config)) {
-    agc_config_ = config;
-    return 0;
-  }
-  WEBRTC_FUNC(GetAgcConfig, (webrtc::AgcConfig& config)) {
-    config = agc_config_;
-    return 0;
-  }
-  WEBRTC_FUNC(SetEcStatus, (bool enable, webrtc::EcModes mode)) {
-    ec_enabled_ = enable;
-    ec_mode_ = mode;
-    return 0;
-  }
-  WEBRTC_FUNC(GetEcStatus, (bool& enabled, webrtc::EcModes& mode)) {
-    enabled = ec_enabled_;
-    mode = ec_mode_;
-    return 0;
-  }
-  WEBRTC_STUB(EnableDriftCompensation, (bool enable))
-  WEBRTC_BOOL_STUB(DriftCompensationEnabled, ())
-  WEBRTC_VOID_STUB(SetDelayOffsetMs, (int offset))
-  WEBRTC_STUB(DelayOffsetMs, ());
-  WEBRTC_FUNC(SetAecmMode, (webrtc::AecmModes mode, bool enableCNG)) {
-    aecm_mode_ = mode;
-    cng_enabled_ = enableCNG;
-    return 0;
-  }
-  WEBRTC_FUNC(GetAecmMode, (webrtc::AecmModes& mode, bool& enabledCNG)) {
-    mode = aecm_mode_;
-    enabledCNG = cng_enabled_;
-    return 0;
-  }
-  WEBRTC_STUB(SetRxNsStatus, (int channel, bool enable, webrtc::NsModes mode));
-  WEBRTC_STUB(GetRxNsStatus, (int channel, bool& enabled,
-                              webrtc::NsModes& mode));
-  WEBRTC_STUB(SetRxAgcStatus, (int channel, bool enable,
-                               webrtc::AgcModes mode));
-  WEBRTC_STUB(GetRxAgcStatus, (int channel, bool& enabled,
-                               webrtc::AgcModes& mode));
-  WEBRTC_STUB(SetRxAgcConfig, (int channel, webrtc::AgcConfig config));
-  WEBRTC_STUB(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config));
-
-  WEBRTC_STUB(RegisterRxVadObserver, (int, webrtc::VoERxVadCallback&));
-  WEBRTC_STUB(DeRegisterRxVadObserver, (int channel));
-  WEBRTC_STUB(VoiceActivityIndicator, (int channel));
-  WEBRTC_FUNC(SetEcMetricsStatus, (bool enable)) {
-    ec_metrics_enabled_ = enable;
-    return 0;
-  }
-  WEBRTC_STUB(GetEcMetricsStatus, (bool& enabled));
-  WEBRTC_STUB(GetEchoMetrics, (int& ERL, int& ERLE, int& RERL, int& A_NLP));
-  WEBRTC_STUB(GetEcDelayMetrics, (int& delay_median, int& delay_std,
-      float& fraction_poor_delays));
-
-  WEBRTC_STUB(StartDebugRecording, (const char* fileNameUTF8));
-  WEBRTC_STUB(StartDebugRecording, (FILE* handle));
-  WEBRTC_STUB(StopDebugRecording, ());
-
-  WEBRTC_FUNC(SetTypingDetectionStatus, (bool enable)) {
-    typing_detection_enabled_ = enable;
-    return 0;
-  }
-  WEBRTC_FUNC(GetTypingDetectionStatus, (bool& enabled)) {
-    enabled = typing_detection_enabled_;
-    return 0;
-  }
-
-  WEBRTC_STUB(TimeSinceLastTyping, (int& seconds));
-  WEBRTC_STUB(SetTypingDetectionParameters, (int timeWindow,
-                                             int costPerTyping,
-                                             int reportingThreshold,
-                                             int penaltyDecay,
-                                             int typeEventDelay));
-  int EnableHighPassFilter(bool enable) {
-    highpass_filter_enabled_ = enable;
-    return 0;
-  }
-  bool IsHighPassFilterEnabled() {
-    return highpass_filter_enabled_;
-  }
-  bool IsStereoChannelSwappingEnabled() {
-    return stereo_swapping_enabled_;
-  }
-  void EnableStereoChannelSwapping(bool enable) {
-    stereo_swapping_enabled_ = enable;
-  }
-  int GetNetEqCapacity() const {
-    auto ch = channels_.find(last_channel_);
-    ASSERT(ch != channels_.end());
-    return ch->second->neteq_capacity;
-  }
-  bool GetNetEqFastAccelerate() const {
-    auto ch = channels_.find(last_channel_);
-    ASSERT(ch != channels_.end());
-    return ch->second->neteq_fast_accelerate;
-  }
-
- private:
-  bool inited_;
-  int last_channel_;
-  std::map<int, Channel*> channels_;
-  bool fail_create_channel_;
-  int num_set_send_codecs_;  // how many times we call SetSendCodec().
-  bool ec_enabled_;
-  bool ec_metrics_enabled_;
-  bool cng_enabled_;
-  bool ns_enabled_;
-  bool agc_enabled_;
-  bool highpass_filter_enabled_;
-  bool stereo_swapping_enabled_;
-  bool typing_detection_enabled_;
-  webrtc::EcModes ec_mode_;
-  webrtc::AecmModes aecm_mode_;
-  webrtc::NsModes ns_mode_;
-  webrtc::AgcModes agc_mode_;
-  webrtc::AgcConfig agc_config_;
-  webrtc::VoiceEngineObserver* observer_;
-  int playout_fail_channel_;
-  int send_fail_channel_;
-  int recording_sample_rate_;
-  int playout_sample_rate_;
-  FakeAudioProcessing audio_processing_;
-};
-
-}  // namespace cricket
-
-#endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_