Move talk/media to webrtc/media

talk/media/base/mediachannel.h

Index: talk/media/base/mediachannel.h
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
deleted file mode 100644
index b72af4d904f291ab25a9c9a92cd95a5dfcd05235..0000000000000000000000000000000000000000
--- a/talk/media/base/mediachannel.h
+++ /dev/null
@@ -1,1137 +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_BASE_MEDIACHANNEL_H_
-#define TALK_MEDIA_BASE_MEDIACHANNEL_H_
-
-#include <string>
-#include <vector>
-
-#include "talk/media/base/codec.h"
-#include "talk/media/base/constants.h"
-#include "talk/media/base/streamparams.h"
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/dscp.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/optional.h"
-#include "webrtc/base/sigslot.h"
-#include "webrtc/base/socket.h"
-#include "webrtc/base/window.h"
-#include "webrtc/media/base/videosinkinterface.h"
-// TODO(juberti): re-evaluate this include
-#include "talk/session/media/audiomonitor.h"
-
-namespace rtc {
-class Buffer;
-class RateLimiter;
-class Timing;
-}
-
-namespace webrtc {
-class AudioSinkInterface;
-}
-
-namespace cricket {
-
-class AudioRenderer;
-class ScreencastId;
-class VideoCapturer;
-class VideoFrame;
-struct RtpHeader;
-struct VideoFormat;
-
-const int kMinRtpHeaderExtensionId = 1;
-const int kMaxRtpHeaderExtensionId = 255;
-const int kScreencastDefaultFps = 5;
-
-template <class T>
-static std::string ToStringIfSet(const char* key, const rtc::Optional<T>& val) {
-  std::string str;
-  if (val) {
-    str = key;
-    str += ": ";
-    str += val ? rtc::ToString(*val) : "";
-    str += ", ";
-  }
-  return str;
-}
-
-template <class T>
-static std::string VectorToString(const std::vector<T>& vals) {
-    std::ostringstream ost;
-    ost << "[";
-    for (size_t i = 0; i < vals.size(); ++i) {
-      if (i > 0) {
-        ost << ", ";
-      }
-      ost << vals[i].ToString();
-    }
-    ost << "]";
-    return ost.str();
-}
-
-// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
-// Used to be flags, but that makes it hard to selectively apply options.
-// We are moving all of the setting of options to structs like this,
-// but some things currently still use flags.
-struct AudioOptions {
-  void SetAll(const AudioOptions& change) {
-    SetFrom(&echo_cancellation, change.echo_cancellation);
-    SetFrom(&auto_gain_control, change.auto_gain_control);
-    SetFrom(&noise_suppression, change.noise_suppression);
-    SetFrom(&highpass_filter, change.highpass_filter);
-    SetFrom(&stereo_swapping, change.stereo_swapping);
-    SetFrom(&audio_jitter_buffer_max_packets,
-            change.audio_jitter_buffer_max_packets);
-    SetFrom(&audio_jitter_buffer_fast_accelerate,
-            change.audio_jitter_buffer_fast_accelerate);
-    SetFrom(&typing_detection, change.typing_detection);
-    SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
-    SetFrom(&conference_mode, change.conference_mode);
-    SetFrom(&adjust_agc_delta, change.adjust_agc_delta);
-    SetFrom(&experimental_agc, change.experimental_agc);
-    SetFrom(&extended_filter_aec, change.extended_filter_aec);
-    SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
-    SetFrom(&experimental_ns, change.experimental_ns);
-    SetFrom(&aec_dump, change.aec_dump);
-    SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
-    SetFrom(&tx_agc_digital_compression_gain,
-            change.tx_agc_digital_compression_gain);
-    SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
-    SetFrom(&recording_sample_rate, change.recording_sample_rate);
-    SetFrom(&playout_sample_rate, change.playout_sample_rate);
-    SetFrom(&dscp, change.dscp);
-    SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
-  }
-
-  bool operator==(const AudioOptions& o) const {
-    return echo_cancellation == o.echo_cancellation &&
-        auto_gain_control == o.auto_gain_control &&
-        noise_suppression == o.noise_suppression &&
-        highpass_filter == o.highpass_filter &&
-        stereo_swapping == o.stereo_swapping &&
-        audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
-        audio_jitter_buffer_fast_accelerate ==
-            o.audio_jitter_buffer_fast_accelerate &&
-        typing_detection == o.typing_detection &&
-        aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
-        conference_mode == o.conference_mode &&
-        experimental_agc == o.experimental_agc &&
-        extended_filter_aec == o.extended_filter_aec &&
-        delay_agnostic_aec == o.delay_agnostic_aec &&
-        experimental_ns == o.experimental_ns &&
-        adjust_agc_delta == o.adjust_agc_delta &&
-        aec_dump == o.aec_dump &&
-        tx_agc_target_dbov == o.tx_agc_target_dbov &&
-        tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
-        tx_agc_limiter == o.tx_agc_limiter &&
-        recording_sample_rate == o.recording_sample_rate &&
-        playout_sample_rate == o.playout_sample_rate &&
-        dscp == o.dscp &&
-        combined_audio_video_bwe == o.combined_audio_video_bwe;
-  }
-
-  std::string ToString() const {
-    std::ostringstream ost;
-    ost << "AudioOptions {";
-    ost << ToStringIfSet("aec", echo_cancellation);
-    ost << ToStringIfSet("agc", auto_gain_control);
-    ost << ToStringIfSet("ns", noise_suppression);
-    ost << ToStringIfSet("hf", highpass_filter);
-    ost << ToStringIfSet("swap", stereo_swapping);
-    ost << ToStringIfSet("audio_jitter_buffer_max_packets",
-                         audio_jitter_buffer_max_packets);
-    ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
-                         audio_jitter_buffer_fast_accelerate);
-    ost << ToStringIfSet("typing", typing_detection);
-    ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
-    ost << ToStringIfSet("conference", conference_mode);
-    ost << ToStringIfSet("agc_delta", adjust_agc_delta);
-    ost << ToStringIfSet("experimental_agc", experimental_agc);
-    ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
-    ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
-    ost << ToStringIfSet("experimental_ns", experimental_ns);
-    ost << ToStringIfSet("aec_dump", aec_dump);
-    ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
-    ost << ToStringIfSet("tx_agc_digital_compression_gain",
-        tx_agc_digital_compression_gain);
-    ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
-    ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
-    ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
-    ost << ToStringIfSet("dscp", dscp);
-    ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
-    ost << "}";
-    return ost.str();
-  }
-
-  // Audio processing that attempts to filter away the output signal from
-  // later inbound pickup.
-  rtc::Optional<bool> echo_cancellation;
-  // Audio processing to adjust the sensitivity of the local mic dynamically.
-  rtc::Optional<bool> auto_gain_control;
-  // Audio processing to filter out background noise.
-  rtc::Optional<bool> noise_suppression;
-  // Audio processing to remove background noise of lower frequencies.
-  rtc::Optional<bool> highpass_filter;
-  // Audio processing to swap the left and right channels.
-  rtc::Optional<bool> stereo_swapping;
-  // Audio receiver jitter buffer (NetEq) max capacity in number of packets.
-  rtc::Optional<int> audio_jitter_buffer_max_packets;
-  // Audio receiver jitter buffer (NetEq) fast accelerate mode.
-  rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
-  // Audio processing to detect typing.
-  rtc::Optional<bool> typing_detection;
-  rtc::Optional<bool> aecm_generate_comfort_noise;
-  rtc::Optional<bool> conference_mode;
-  rtc::Optional<int> adjust_agc_delta;
-  rtc::Optional<bool> experimental_agc;
-  rtc::Optional<bool> extended_filter_aec;
-  rtc::Optional<bool> delay_agnostic_aec;
-  rtc::Optional<bool> experimental_ns;
-  rtc::Optional<bool> aec_dump;
-  // Note that tx_agc_* only applies to non-experimental AGC.
-  rtc::Optional<uint16_t> tx_agc_target_dbov;
-  rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
-  rtc::Optional<bool> tx_agc_limiter;
-  rtc::Optional<uint32_t> recording_sample_rate;
-  rtc::Optional<uint32_t> playout_sample_rate;
-  // Set DSCP value for packet sent from audio channel.
-  rtc::Optional<bool> dscp;
-  // Enable combined audio+bandwidth BWE.
-  rtc::Optional<bool> combined_audio_video_bwe;
-
- private:
-  template <typename T>
-  static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
-    if (o) {
-      *s = o;
-    }
-  }
-};
-
-// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
-// Used to be flags, but that makes it hard to selectively apply options.
-// We are moving all of the setting of options to structs like this,
-// but some things currently still use flags.
-struct VideoOptions {
-  void SetAll(const VideoOptions& change) {
-    SetFrom(&video_noise_reduction, change.video_noise_reduction);
-    SetFrom(&cpu_overuse_detection, change.cpu_overuse_detection);
-    SetFrom(&conference_mode, change.conference_mode);
-    SetFrom(&dscp, change.dscp);
-    SetFrom(&suspend_below_min_bitrate, change.suspend_below_min_bitrate);
-    SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps);
-    SetFrom(&disable_prerenderer_smoothing,
-            change.disable_prerenderer_smoothing);
-  }
-
-  bool operator==(const VideoOptions& o) const {
-    return video_noise_reduction == o.video_noise_reduction &&
-           cpu_overuse_detection == o.cpu_overuse_detection &&
-           conference_mode == o.conference_mode &&
-           dscp == o.dscp &&
-           suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
-           screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps &&
-           disable_prerenderer_smoothing == o.disable_prerenderer_smoothing;
-  }
-
-  std::string ToString() const {
-    std::ostringstream ost;
-    ost << "VideoOptions {";
-    ost << ToStringIfSet("noise reduction", video_noise_reduction);
-    ost << ToStringIfSet("cpu overuse detection", cpu_overuse_detection);
-    ost << ToStringIfSet("conference mode", conference_mode);
-    ost << ToStringIfSet("dscp", dscp);
-    ost << ToStringIfSet("suspend below min bitrate",
-                         suspend_below_min_bitrate);
-    ost << ToStringIfSet("screencast min bitrate kbps",
-                         screencast_min_bitrate_kbps);
-    ost << "}";
-    return ost.str();
-  }
-
-  // Enable denoising? This flag comes from the getUserMedia
-  // constraint 'googNoiseReduction', and WebRtcVideoEngine2 passes it
-  // on to the codec options. Disabled by default.
-  rtc::Optional<bool> video_noise_reduction;
-  // Enable WebRTC Cpu Overuse Detection. This flag comes from the
-  // PeerConnection constraint 'googCpuOveruseDetection' and is
-  // checked in WebRtcVideoChannel2::OnLoadUpdate, where it's passed
-  // to VideoCapturer::video_adapter()->OnCpuResolutionRequest.
-  rtc::Optional<bool> cpu_overuse_detection;
-  // Use conference mode? This flag comes from the remote
-  // description's SDP line 'a=x-google-flag:conference', copied over
-  // by VideoChannel::SetRemoteContent_w, and ultimately used by
-  // conference mode screencast logic in
-  // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig.
-  // The special screencast behaviour is disabled by default.
-  rtc::Optional<bool> conference_mode;
-  // Set DSCP value for packet sent from video channel. This flag
-  // comes from the PeerConnection constraint 'googDscp' and,
-  // WebRtcVideoChannel2::SetOptions checks it before calling
-  // MediaChannel::SetDscp. If enabled, rtc::DSCP_AF41 is used. If
-  // disabled, which is the default, rtc::DSCP_DEFAULT is used.
-  rtc::Optional<bool> dscp;
-  // Enable WebRTC suspension of video. No video frames will be sent
-  // when the bitrate is below the configured minimum bitrate. This
-  // flag comes from the PeerConnection constraint
-  // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel2 copies it
-  // to VideoSendStream::Config::suspend_below_min_bitrate.
-  rtc::Optional<bool> suspend_below_min_bitrate;
-  // Force screencast to use a minimum bitrate. This flag comes from
-  // the PeerConnection constraint 'googScreencastMinBitrate'. It is
-  // copied to the encoder config by WebRtcVideoChannel2.
-  rtc::Optional<int> screencast_min_bitrate_kbps;
-  // Set to true if the renderer has an algorithm of frame selection.
-  // If the value is true, then WebRTC will hand over a frame as soon as
-  // possible without delay, and rendering smoothness is completely the duty
-  // of the renderer;
-  // If the value is false, then WebRTC is responsible to delay frame release
-  // in order to increase rendering smoothness.
-  //
-  // This flag comes from PeerConnection's RtcConfiguration, but is
-  // currently only set by the command line flag
-  // 'disable-rtc-smoothness-algorithm'.
-  // WebRtcVideoChannel2::AddRecvStream copies it to the created
-  // WebRtcVideoReceiveStream, where it is returned by the
-  // SmoothsRenderedFrames method. This method is used by the
-  // VideoReceiveStream, where the value is passed on to the
-  // IncomingVideoStream constructor.
-  rtc::Optional<bool> disable_prerenderer_smoothing;
-
- private:
-  template <typename T>
-  static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
-    if (o) {
-      *s = o;
-    }
-  }
-};
-
-struct RtpHeaderExtension {
-  RtpHeaderExtension() : id(0) {}
-  RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {}
-
-  bool operator==(const RtpHeaderExtension& ext) const {
-    // id is a reserved word in objective-c. Therefore the id attribute has to
-    // be a fully qualified name in order to compile on IOS.
-    return this->id == ext.id &&
-        uri == ext.uri;
-  }
-
-  std::string ToString() const {
-    std::ostringstream ost;
-    ost << "{";
-    ost << "uri: " << uri;
-    ost << ", id: " << id;
-    ost << "}";
-    return ost.str();
-  }
-
-  std::string uri;
-  int id;
-  // TODO(juberti): SendRecv direction;
-};
-
-// Returns the named header extension if found among all extensions, NULL
-// otherwise.
-inline const RtpHeaderExtension* FindHeaderExtension(
-    const std::vector<RtpHeaderExtension>& extensions,
-    const std::string& name) {
-  for (std::vector<RtpHeaderExtension>::const_iterator it = extensions.begin();
-       it != extensions.end(); ++it) {
-    if (it->uri == name)
-      return &(*it);
-  }
-  return NULL;
-}
-
-class MediaChannel : public sigslot::has_slots<> {
- public:
-  class NetworkInterface {
-   public:
-    enum SocketType { ST_RTP, ST_RTCP };
-    virtual bool SendPacket(rtc::Buffer* packet,
-                            const rtc::PacketOptions& options) = 0;
-    virtual bool SendRtcp(rtc::Buffer* packet,
-                          const rtc::PacketOptions& options) = 0;
-    virtual int SetOption(SocketType type, rtc::Socket::Option opt,
-                          int option) = 0;
-    virtual ~NetworkInterface() {}
-  };
-
-  MediaChannel() : network_interface_(NULL) {}
-  virtual ~MediaChannel() {}
-
-  // Sets the abstract interface class for sending RTP/RTCP data.
-  virtual void SetInterface(NetworkInterface *iface) {
-    rtc::CritScope cs(&network_interface_crit_);
-    network_interface_ = iface;
-  }
-
-  // Called when a RTP packet is received.
-  virtual void OnPacketReceived(rtc::Buffer* packet,
-                                const rtc::PacketTime& packet_time) = 0;
-  // Called when a RTCP packet is received.
-  virtual void OnRtcpReceived(rtc::Buffer* packet,
-                              const rtc::PacketTime& packet_time) = 0;
-  // Called when the socket's ability to send has changed.
-  virtual void OnReadyToSend(bool ready) = 0;
-  // Creates a new outgoing media stream with SSRCs and CNAME as described
-  // by sp.
-  virtual bool AddSendStream(const StreamParams& sp) = 0;
-  // Removes an outgoing media stream.
-  // ssrc must be the first SSRC of the media stream if the stream uses
-  // multiple SSRCs.
-  virtual bool RemoveSendStream(uint32_t ssrc) = 0;
-  // Creates a new incoming media stream with SSRCs and CNAME as described
-  // by sp.
-  virtual bool AddRecvStream(const StreamParams& sp) = 0;
-  // Removes an incoming media stream.
-  // ssrc must be the first SSRC of the media stream if the stream uses
-  // multiple SSRCs.
-  virtual bool RemoveRecvStream(uint32_t ssrc) = 0;
-
-  // Returns the absoulte sendtime extension id value from media channel.
-  virtual int GetRtpSendTimeExtnId() const {
-    return -1;
-  }
-
-  // Base method to send packet using NetworkInterface.
-  bool SendPacket(rtc::Buffer* packet, const rtc::PacketOptions& options) {
-    return DoSendPacket(packet, false, options);
-  }
-
-  bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) {
-    return DoSendPacket(packet, true, options);
-  }
-
-  int SetOption(NetworkInterface::SocketType type,
-                rtc::Socket::Option opt,
-                int option) {
-    rtc::CritScope cs(&network_interface_crit_);
-    if (!network_interface_)
-      return -1;
-
-    return network_interface_->SetOption(type, opt, option);
-  }
-
- protected:
-  // This method sets DSCP |value| on both RTP and RTCP channels.
-  int SetDscp(rtc::DiffServCodePoint value) {
-    int ret;
-    ret = SetOption(NetworkInterface::ST_RTP,
-                    rtc::Socket::OPT_DSCP,
-                    value);
-    if (ret == 0) {
-      ret = SetOption(NetworkInterface::ST_RTCP,
-                      rtc::Socket::OPT_DSCP,
-                      value);
-    }
-    return ret;
-  }
-
- private:
-  bool DoSendPacket(rtc::Buffer* packet,
-                    bool rtcp,
-                    const rtc::PacketOptions& options) {
-    rtc::CritScope cs(&network_interface_crit_);
-    if (!network_interface_)
-      return false;
-
-    return (!rtcp) ? network_interface_->SendPacket(packet, options)
-                   : network_interface_->SendRtcp(packet, options);
-  }
-
-  // |network_interface_| can be accessed from the worker_thread and
-  // from any MediaEngine threads. This critical section is to protect accessing
-  // of network_interface_ object.
-  rtc::CriticalSection network_interface_crit_;
-  NetworkInterface* network_interface_;
-};
-
-enum SendFlags {
-  SEND_NOTHING,
-  SEND_MICROPHONE
-};
-
-// The stats information is structured as follows:
-// Media are represented by either MediaSenderInfo or MediaReceiverInfo.
-// Media contains a vector of SSRC infos that are exclusively used by this
-// media. (SSRCs shared between media streams can't be represented.)
-
-// Information about an SSRC.
-// This data may be locally recorded, or received in an RTCP SR or RR.
-struct SsrcSenderInfo {
-  SsrcSenderInfo()
-      : ssrc(0),
-    timestamp(0) {
-  }
-  uint32_t ssrc;
-  double timestamp;  // NTP timestamp, represented as seconds since epoch.
-};
-
-struct SsrcReceiverInfo {
-  SsrcReceiverInfo()
-      : ssrc(0),
-        timestamp(0) {
-  }
-  uint32_t ssrc;
-  double timestamp;
-};
-
-struct MediaSenderInfo {
-  MediaSenderInfo()
-      : bytes_sent(0),
-        packets_sent(0),
-        packets_lost(0),
-        fraction_lost(0.0),
-        rtt_ms(0) {
-  }
-  void add_ssrc(const SsrcSenderInfo& stat) {
-    local_stats.push_back(stat);
-  }
-  // Temporary utility function for call sites that only provide SSRC.
-  // As more info is added into SsrcSenderInfo, this function should go away.
-  void add_ssrc(uint32_t ssrc) {
-    SsrcSenderInfo stat;
-    stat.ssrc = ssrc;
-    add_ssrc(stat);
-  }
-  // Utility accessor for clients that are only interested in ssrc numbers.
-  std::vector<uint32_t> ssrcs() const {
-    std::vector<uint32_t> retval;
-    for (std::vector<SsrcSenderInfo>::const_iterator it = local_stats.begin();
-         it != local_stats.end(); ++it) {
-      retval.push_back(it->ssrc);
-    }
-    return retval;
-  }
-  // Utility accessor for clients that make the assumption only one ssrc
-  // exists per media.
-  // This will eventually go away.
-  uint32_t ssrc() const {
-    if (local_stats.size() > 0) {
-      return local_stats[0].ssrc;
-    } else {
-      return 0;
-    }
-  }
-  int64_t bytes_sent;
-  int packets_sent;
-  int packets_lost;
-  float fraction_lost;
-  int64_t rtt_ms;
-  std::string codec_name;
-  std::vector<SsrcSenderInfo> local_stats;
-  std::vector<SsrcReceiverInfo> remote_stats;
-};
-
-template<class T>
-struct VariableInfo {
-  VariableInfo()
-      : min_val(),
-        mean(0.0),
-        max_val(),
-        variance(0.0) {
-  }
-  T min_val;
-  double mean;
-  T max_val;
-  double variance;
-};
-
-struct MediaReceiverInfo {
-  MediaReceiverInfo()
-      : bytes_rcvd(0),
-        packets_rcvd(0),
-        packets_lost(0),
-        fraction_lost(0.0) {
-  }
-  void add_ssrc(const SsrcReceiverInfo& stat) {
-    local_stats.push_back(stat);
-  }
-  // Temporary utility function for call sites that only provide SSRC.
-  // As more info is added into SsrcSenderInfo, this function should go away.
-  void add_ssrc(uint32_t ssrc) {
-    SsrcReceiverInfo stat;
-    stat.ssrc = ssrc;
-    add_ssrc(stat);
-  }
-  std::vector<uint32_t> ssrcs() const {
-    std::vector<uint32_t> retval;
-    for (std::vector<SsrcReceiverInfo>::const_iterator it = local_stats.begin();
-         it != local_stats.end(); ++it) {
-      retval.push_back(it->ssrc);
-    }
-    return retval;
-  }
-  // Utility accessor for clients that make the assumption only one ssrc
-  // exists per media.
-  // This will eventually go away.
-  uint32_t ssrc() const {
-    if (local_stats.size() > 0) {
-      return local_stats[0].ssrc;
-    } else {
-      return 0;
-    }
-  }
-
-  int64_t bytes_rcvd;
-  int packets_rcvd;
-  int packets_lost;
-  float fraction_lost;
-  std::string codec_name;
-  std::vector<SsrcReceiverInfo> local_stats;
-  std::vector<SsrcSenderInfo> remote_stats;
-};
-
-struct VoiceSenderInfo : public MediaSenderInfo {
-  VoiceSenderInfo()
-      : ext_seqnum(0),
-        jitter_ms(0),
-        audio_level(0),
-        aec_quality_min(0.0),
-        echo_delay_median_ms(0),
-        echo_delay_std_ms(0),
-        echo_return_loss(0),
-        echo_return_loss_enhancement(0),
-        typing_noise_detected(false) {
-  }
-
-  int ext_seqnum;
-  int jitter_ms;
-  int audio_level;
-  float aec_quality_min;
-  int echo_delay_median_ms;
-  int echo_delay_std_ms;
-  int echo_return_loss;
-  int echo_return_loss_enhancement;
-  bool typing_noise_detected;
-};
-
-struct VoiceReceiverInfo : public MediaReceiverInfo {
-  VoiceReceiverInfo()
-      : ext_seqnum(0),
-        jitter_ms(0),
-        jitter_buffer_ms(0),
-        jitter_buffer_preferred_ms(0),
-        delay_estimate_ms(0),
-        audio_level(0),
-        expand_rate(0),
-        speech_expand_rate(0),
-        secondary_decoded_rate(0),
-        accelerate_rate(0),
-        preemptive_expand_rate(0),
-        decoding_calls_to_silence_generator(0),
-        decoding_calls_to_neteq(0),
-        decoding_normal(0),
-        decoding_plc(0),
-        decoding_cng(0),
-        decoding_plc_cng(0),
-        capture_start_ntp_time_ms(-1) {}
-
-  int ext_seqnum;
-  int jitter_ms;
-  int jitter_buffer_ms;
-  int jitter_buffer_preferred_ms;
-  int delay_estimate_ms;
-  int audio_level;
-  // fraction of synthesized audio inserted through expansion.
-  float expand_rate;
-  // fraction of synthesized speech inserted through expansion.
-  float speech_expand_rate;
-  // fraction of data out of secondary decoding, including FEC and RED.
-  float secondary_decoded_rate;
-  // Fraction of data removed through time compression.
-  float accelerate_rate;
-  // Fraction of data inserted through time stretching.
-  float preemptive_expand_rate;
-  int decoding_calls_to_silence_generator;
-  int decoding_calls_to_neteq;
-  int decoding_normal;
-  int decoding_plc;
-  int decoding_cng;
-  int decoding_plc_cng;
-  // Estimated capture start time in NTP time in ms.
-  int64_t capture_start_ntp_time_ms;
-};
-
-struct VideoSenderInfo : public MediaSenderInfo {
-  VideoSenderInfo()
-      : packets_cached(0),
-        firs_rcvd(0),
-        plis_rcvd(0),
-        nacks_rcvd(0),
-        input_frame_width(0),
-        input_frame_height(0),
-        send_frame_width(0),
-        send_frame_height(0),
-        framerate_input(0),
-        framerate_sent(0),
-        nominal_bitrate(0),
-        preferred_bitrate(0),
-        adapt_reason(0),
-        adapt_changes(0),
-        avg_encode_ms(0),
-        encode_usage_percent(0) {
-  }
-
-  std::vector<SsrcGroup> ssrc_groups;
-  std::string encoder_implementation_name;
-  int packets_cached;
-  int firs_rcvd;
-  int plis_rcvd;
-  int nacks_rcvd;
-  int input_frame_width;
-  int input_frame_height;
-  int send_frame_width;
-  int send_frame_height;
-  int framerate_input;
-  int framerate_sent;
-  int nominal_bitrate;
-  int preferred_bitrate;
-  int adapt_reason;
-  int adapt_changes;
-  int avg_encode_ms;
-  int encode_usage_percent;
-  VariableInfo<int> adapt_frame_drops;
-  VariableInfo<int> effects_frame_drops;
-  VariableInfo<double> capturer_frame_time;
-};
-
-struct VideoReceiverInfo : public MediaReceiverInfo {
-  VideoReceiverInfo()
-      : packets_concealed(0),
-        firs_sent(0),
-        plis_sent(0),
-        nacks_sent(0),
-        frame_width(0),
-        frame_height(0),
-        framerate_rcvd(0),
-        framerate_decoded(0),
-        framerate_output(0),
-        framerate_render_input(0),
-        framerate_render_output(0),
-        decode_ms(0),
-        max_decode_ms(0),
-        jitter_buffer_ms(0),
-        min_playout_delay_ms(0),
-        render_delay_ms(0),
-        target_delay_ms(0),
-        current_delay_ms(0),
-        capture_start_ntp_time_ms(-1) {
-  }
-
-  std::vector<SsrcGroup> ssrc_groups;
-  std::string decoder_implementation_name;
-  int packets_concealed;
-  int firs_sent;
-  int plis_sent;
-  int nacks_sent;
-  int frame_width;
-  int frame_height;
-  int framerate_rcvd;
-  int framerate_decoded;
-  int framerate_output;
-  // Framerate as sent to the renderer.
-  int framerate_render_input;
-  // Framerate that the renderer reports.
-  int framerate_render_output;
-
-  // All stats below are gathered per-VideoReceiver, but some will be correlated
-  // across MediaStreamTracks.  NOTE(hta): when sinking stats into per-SSRC
-  // structures, reflect this in the new layout.
-
-  // Current frame decode latency.
-  int decode_ms;
-  // Maximum observed frame decode latency.
-  int max_decode_ms;
-  // Jitter (network-related) latency.
-  int jitter_buffer_ms;
-  // Requested minimum playout latency.
-  int min_playout_delay_ms;
-  // Requested latency to account for rendering delay.
-  int render_delay_ms;
-  // Target overall delay: network+decode+render, accounting for
-  // min_playout_delay_ms.
-  int target_delay_ms;
-  // Current overall delay, possibly ramping towards target_delay_ms.
-  int current_delay_ms;
-
-  // Estimated capture start time in NTP time in ms.
-  int64_t capture_start_ntp_time_ms;
-};
-
-struct DataSenderInfo : public MediaSenderInfo {
-  DataSenderInfo()
-      : ssrc(0) {
-  }
-
-  uint32_t ssrc;
-};
-
-struct DataReceiverInfo : public MediaReceiverInfo {
-  DataReceiverInfo()
-      : ssrc(0) {
-  }
-
-  uint32_t ssrc;
-};
-
-struct BandwidthEstimationInfo {
-  BandwidthEstimationInfo()
-      : available_send_bandwidth(0),
-        available_recv_bandwidth(0),
-        target_enc_bitrate(0),
-        actual_enc_bitrate(0),
-        retransmit_bitrate(0),
-        transmit_bitrate(0),
-        bucket_delay(0) {
-  }
-
-  int available_send_bandwidth;
-  int available_recv_bandwidth;
-  int target_enc_bitrate;
-  int actual_enc_bitrate;
-  int retransmit_bitrate;
-  int transmit_bitrate;
-  int64_t bucket_delay;
-};
-
-struct VoiceMediaInfo {
-  void Clear() {
-    senders.clear();
-    receivers.clear();
-  }
-  std::vector<VoiceSenderInfo> senders;
-  std::vector<VoiceReceiverInfo> receivers;
-};
-
-struct VideoMediaInfo {
-  void Clear() {
-    senders.clear();
-    receivers.clear();
-    bw_estimations.clear();
-  }
-  std::vector<VideoSenderInfo> senders;
-  std::vector<VideoReceiverInfo> receivers;
-  std::vector<BandwidthEstimationInfo> bw_estimations;
-};
-
-struct DataMediaInfo {
-  void Clear() {
-    senders.clear();
-    receivers.clear();
-  }
-  std::vector<DataSenderInfo> senders;
-  std::vector<DataReceiverInfo> receivers;
-};
-
-struct RtcpParameters {
-  bool reduced_size = false;
-};
-
-template <class Codec>
-struct RtpParameters {
-  virtual std::string ToString() const {
-    std::ostringstream ost;
-    ost << "{";
-    ost << "codecs: " << VectorToString(codecs) << ", ";
-    ost << "extensions: " << VectorToString(extensions);
-    ost << "}";
-    return ost.str();
-  }
-
-  std::vector<Codec> codecs;
-  std::vector<RtpHeaderExtension> extensions;
-  // TODO(pthatcher): Add streams.
-  RtcpParameters rtcp;
-};
-
-template <class Codec, class Options>
-struct RtpSendParameters : RtpParameters<Codec> {
-  std::string ToString() const override {
-    std::ostringstream ost;
-    ost << "{";
-    ost << "codecs: " << VectorToString(this->codecs) << ", ";
-    ost << "extensions: " << VectorToString(this->extensions) << ", ";
-    ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
-    ost << "options: " << options.ToString();
-    ost << "}";
-    return ost.str();
-  }
-
-  int max_bandwidth_bps = -1;
-  Options options;
-};
-
-struct AudioSendParameters : RtpSendParameters<AudioCodec, AudioOptions> {
-};
-
-struct AudioRecvParameters : RtpParameters<AudioCodec> {
-};
-
-class VoiceMediaChannel : public MediaChannel {
- public:
-  enum Error {
-    ERROR_NONE = 0,                       // No error.
-    ERROR_OTHER,                          // Other errors.
-    ERROR_REC_DEVICE_OPEN_FAILED = 100,   // Could not open mic.
-    ERROR_REC_DEVICE_MUTED,               // Mic was muted by OS.
-    ERROR_REC_DEVICE_SILENT,              // No background noise picked up.
-    ERROR_REC_DEVICE_SATURATION,          // Mic input is clipping.
-    ERROR_REC_DEVICE_REMOVED,             // Mic was removed while active.
-    ERROR_REC_RUNTIME_ERROR,              // Processing is encountering errors.
-    ERROR_REC_SRTP_ERROR,                 // Generic SRTP failure.
-    ERROR_REC_SRTP_AUTH_FAILED,           // Failed to authenticate packets.
-    ERROR_REC_TYPING_NOISE_DETECTED,      // Typing noise is detected.
-    ERROR_PLAY_DEVICE_OPEN_FAILED = 200,  // Could not open playout.
-    ERROR_PLAY_DEVICE_MUTED,              // Playout muted by OS.
-    ERROR_PLAY_DEVICE_REMOVED,            // Playout removed while active.
-    ERROR_PLAY_RUNTIME_ERROR,             // Errors in voice processing.
-    ERROR_PLAY_SRTP_ERROR,                // Generic SRTP failure.
-    ERROR_PLAY_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
-    ERROR_PLAY_SRTP_REPLAY,               // Packet replay detected.
-  };
-
-  VoiceMediaChannel() {}
-  virtual ~VoiceMediaChannel() {}
-  virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
-  virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
-  // Starts or stops playout of received audio.
-  virtual bool SetPlayout(bool playout) = 0;
-  // Starts or stops sending (and potentially capture) of local audio.
-  virtual bool SetSend(SendFlags flag) = 0;
-  // Configure stream for sending.
-  virtual bool SetAudioSend(uint32_t ssrc,
-                            bool enable,
-                            const AudioOptions* options,
-                            AudioRenderer* renderer) = 0;
-  // Gets current energy levels for all incoming streams.
-  virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0;
-  // Get the current energy level of the stream sent to the speaker.
-  virtual int GetOutputLevel() = 0;
-  // Get the time in milliseconds since last recorded keystroke, or negative.
-  virtual int GetTimeSinceLastTyping() = 0;
-  // Temporarily exposed field for tuning typing detect options.
-  virtual void SetTypingDetectionParameters(int time_window,
-    int cost_per_typing, int reporting_threshold, int penalty_decay,
-    int type_event_delay) = 0;
-  // Set speaker output volume of the specified ssrc.
-  virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0;
-  // Returns if the telephone-event has been negotiated.
-  virtual bool CanInsertDtmf() = 0;
-  // Send a DTMF |event|. The DTMF out-of-band signal will be used.
-  // The |ssrc| should be either 0 or a valid send stream ssrc.
-  // The valid value for the |event| are 0 to 15 which corresponding to
-  // DTMF event 0-9, *, #, A-D.
-  virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
-  // Gets quality stats for the channel.
-  virtual bool GetStats(VoiceMediaInfo* info) = 0;
-
-  virtual void SetRawAudioSink(
-      uint32_t ssrc,
-      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
-};
-
-struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> {
-};
-
-struct VideoRecvParameters : RtpParameters<VideoCodec> {
-};
-
-class VideoMediaChannel : public MediaChannel {
- public:
-  enum Error {
-    ERROR_NONE = 0,                       // No error.
-    ERROR_OTHER,                          // Other errors.
-    ERROR_REC_DEVICE_OPEN_FAILED = 100,   // Could not open camera.
-    ERROR_REC_DEVICE_NO_DEVICE,           // No camera.
-    ERROR_REC_DEVICE_IN_USE,              // Device is in already use.
-    ERROR_REC_DEVICE_REMOVED,             // Device is removed.
-    ERROR_REC_SRTP_ERROR,                 // Generic sender SRTP failure.
-    ERROR_REC_SRTP_AUTH_FAILED,           // Failed to authenticate packets.
-    ERROR_REC_CPU_MAX_CANT_DOWNGRADE,     // Can't downgrade capture anymore.
-    ERROR_PLAY_SRTP_ERROR = 200,          // Generic receiver SRTP failure.
-    ERROR_PLAY_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
-    ERROR_PLAY_SRTP_REPLAY,               // Packet replay detected.
-  };
-
-  VideoMediaChannel() {}
-  virtual ~VideoMediaChannel() {}
-
-  virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
-  virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
-  // Gets the currently set codecs/payload types to be used for outgoing media.
-  virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
-  // Starts or stops transmission (and potentially capture) of local video.
-  virtual bool SetSend(bool send) = 0;
-  // Configure stream for sending.
-  virtual bool SetVideoSend(uint32_t ssrc,
-                            bool enable,
-                            const VideoOptions* options) = 0;
-  // Sets the sink object to be used for the specified stream.
-  // If SSRC is 0, the renderer is used for the 'default' stream.
-  virtual bool SetSink(uint32_t ssrc,
-                       rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0;
-  // If |ssrc| is 0, replace the default capturer (engine capturer) with
-  // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC.
-  virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0;
-  // Gets quality stats for the channel.
-  virtual bool GetStats(VideoMediaInfo* info) = 0;
-};
-
-enum DataMessageType {
-  // Chrome-Internal use only.  See SctpDataMediaChannel for the actual PPID
-  // values.
-  DMT_NONE = 0,
-  DMT_CONTROL = 1,
-  DMT_BINARY = 2,
-  DMT_TEXT = 3,
-};
-
-// Info about data received in DataMediaChannel.  For use in
-// DataMediaChannel::SignalDataReceived and in all of the signals that
-// signal fires, on up the chain.
-struct ReceiveDataParams {
-  // The in-packet stream indentifier.
-  // For SCTP, this is really SID, not SSRC.
-  uint32_t ssrc;
-  // The type of message (binary, text, or control).
-  DataMessageType type;
-  // A per-stream value incremented per packet in the stream.
-  int seq_num;
-  // A per-stream value monotonically increasing with time.
-  int timestamp;
-
-  ReceiveDataParams() :
-      ssrc(0),
-      type(DMT_TEXT),
-      seq_num(0),
-      timestamp(0) {
-  }
-};
-
-struct SendDataParams {
-  // The in-packet stream indentifier.
-  // For SCTP, this is really SID, not SSRC.
-  uint32_t ssrc;
-  // The type of message (binary, text, or control).
-  DataMessageType type;
-
-  // For SCTP, whether to send messages flagged as ordered or not.
-  // If false, messages can be received out of order.
-  bool ordered;
-  // For SCTP, whether the messages are sent reliably or not.
-  // If false, messages may be lost.
-  bool reliable;
-  // For SCTP, if reliable == false, provide partial reliability by
-  // resending up to this many times.  Either count or millis
-  // is supported, not both at the same time.
-  int max_rtx_count;
-  // For SCTP, if reliable == false, provide partial reliability by
-  // resending for up to this many milliseconds.  Either count or millis
-  // is supported, not both at the same time.
-  int max_rtx_ms;
-
-  SendDataParams() :
-      ssrc(0),
-      type(DMT_TEXT),
-      // TODO(pthatcher): Make these true by default?
-      ordered(false),
-      reliable(false),
-      max_rtx_count(0),
-      max_rtx_ms(0) {
-  }
-};
-
-enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
-
-struct DataOptions {
-  std::string ToString() const {
-    return "{}";
-  }
-};
-
-struct DataSendParameters : RtpSendParameters<DataCodec, DataOptions> {
-  std::string ToString() const {
-    std::ostringstream ost;
-    // Options and extensions aren't used.
-    ost << "{";
-    ost << "codecs: " << VectorToString(codecs) << ", ";
-    ost << "max_bandwidth_bps: " << max_bandwidth_bps;
-    ost << "}";
-    return ost.str();
-  }
-};
-
-struct DataRecvParameters : RtpParameters<DataCodec> {
-};
-
-class DataMediaChannel : public MediaChannel {
- public:
-  enum Error {
-    ERROR_NONE = 0,                       // No error.
-    ERROR_OTHER,                          // Other errors.
-    ERROR_SEND_SRTP_ERROR = 200,          // Generic SRTP failure.
-    ERROR_SEND_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
-    ERROR_RECV_SRTP_ERROR,                // Generic SRTP failure.
-    ERROR_RECV_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
-    ERROR_RECV_SRTP_REPLAY,               // Packet replay detected.
-  };
-
-  virtual ~DataMediaChannel() {}
-
-  virtual bool SetSendParameters(const DataSendParameters& params) = 0;
-  virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
-
-  // TODO(pthatcher): Implement this.
-  virtual bool GetStats(DataMediaInfo* info) { return true; }
-
-  virtual bool SetSend(bool send) = 0;
-  virtual bool SetReceive(bool receive) = 0;
-
-  virtual bool SendData(
-      const SendDataParams& params,
-      const rtc::Buffer& payload,
-      SendDataResult* result = NULL) = 0;
-  // Signals when data is received (params, data, len)
-  sigslot::signal3<const ReceiveDataParams&,
-                   const char*,
-                   size_t> SignalDataReceived;
-  // Signal when the media channel is ready to send the stream. Arguments are:
-  //     writable(bool)
-  sigslot::signal1<bool> SignalReadyToSend;
-  // Signal for notifying that the remote side has closed the DataChannel.
-  sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
-};
-
-}  // namespace cricket
-
-#endif  // TALK_MEDIA_BASE_MEDIACHANNEL_H_