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webrtc/talk/media/base/mediachannel.h
buildbot@webrtc.org d4e598d57a (Auto)update libjingle 72097588-> 72159069 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00

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/*
* 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 "webrtc/base/basictypes.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/socket.h"
#include "webrtc/base/window.h"
#include "talk/media/base/codec.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
// TODO(juberti): re-evaluate this include
#include "talk/session/media/audiomonitor.h"
namespace rtc {
class Buffer;
class RateLimiter;
class Timing;
}
namespace cricket {
class AudioRenderer;
struct RtpHeader;
class ScreencastId;
struct VideoFormat;
class VideoCapturer;
class VideoRenderer;
const int kMinRtpHeaderExtensionId = 1;
const int kMaxRtpHeaderExtensionId = 255;
const int kScreencastDefaultFps = 5;
const int kHighStartBitrate = 1500;
// Used in AudioOptions and VideoOptions to signify "unset" values.
template <class T>
class Settable {
public:
Settable() : set_(false), val_() {}
explicit Settable(T val) : set_(true), val_(val) {}
bool IsSet() const {
return set_;
}
bool Get(T* out) const {
*out = val_;
return set_;
}
T GetWithDefaultIfUnset(const T& default_value) const {
return set_ ? val_ : default_value;
}
virtual void Set(T val) {
set_ = true;
val_ = val;
}
void Clear() {
Set(T());
set_ = false;
}
void SetFrom(const Settable<T>& o) {
// Set this value based on the value of o, iff o is set. If this value is
// set and o is unset, the current value will be unchanged.
T val;
if (o.Get(&val)) {
Set(val);
}
}
std::string ToString() const {
return set_ ? rtc::ToString(val_) : "";
}
bool operator==(const Settable<T>& o) const {
// Equal if both are unset with any value or both set with the same value.
return (set_ == o.set_) && (!set_ || (val_ == o.val_));
}
bool operator!=(const Settable<T>& o) const {
return !operator==(o);
}
protected:
void InitializeValue(const T &val) {
val_ = val;
}
private:
bool set_;
T val_;
};
class SettablePercent : public Settable<float> {
public:
virtual void Set(float val) {
if (val < 0) {
val = 0;
}
if (val > 1.0) {
val = 1.0;
}
Settable<float>::Set(val);
}
};
template <class T>
static std::string ToStringIfSet(const char* key, const Settable<T>& val) {
std::string str;
if (val.IsSet()) {
str = key;
str += ": ";
str += val.ToString();
str += ", ";
}
return 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) {
echo_cancellation.SetFrom(change.echo_cancellation);
auto_gain_control.SetFrom(change.auto_gain_control);
rx_auto_gain_control.SetFrom(change.rx_auto_gain_control);
noise_suppression.SetFrom(change.noise_suppression);
highpass_filter.SetFrom(change.highpass_filter);
stereo_swapping.SetFrom(change.stereo_swapping);
typing_detection.SetFrom(change.typing_detection);
aecm_generate_comfort_noise.SetFrom(change.aecm_generate_comfort_noise);
conference_mode.SetFrom(change.conference_mode);
adjust_agc_delta.SetFrom(change.adjust_agc_delta);
experimental_agc.SetFrom(change.experimental_agc);
experimental_aec.SetFrom(change.experimental_aec);
experimental_ns.SetFrom(change.experimental_ns);
aec_dump.SetFrom(change.aec_dump);
tx_agc_target_dbov.SetFrom(change.tx_agc_target_dbov);
tx_agc_digital_compression_gain.SetFrom(
change.tx_agc_digital_compression_gain);
tx_agc_limiter.SetFrom(change.tx_agc_limiter);
rx_agc_target_dbov.SetFrom(change.rx_agc_target_dbov);
rx_agc_digital_compression_gain.SetFrom(
change.rx_agc_digital_compression_gain);
rx_agc_limiter.SetFrom(change.rx_agc_limiter);
recording_sample_rate.SetFrom(change.recording_sample_rate);
playout_sample_rate.SetFrom(change.playout_sample_rate);
dscp.SetFrom(change.dscp);
opus_fec.SetFrom(change.opus_fec);
}
bool operator==(const AudioOptions& o) const {
return echo_cancellation == o.echo_cancellation &&
auto_gain_control == o.auto_gain_control &&
rx_auto_gain_control == o.rx_auto_gain_control &&
noise_suppression == o.noise_suppression &&
highpass_filter == o.highpass_filter &&
stereo_swapping == o.stereo_swapping &&
typing_detection == o.typing_detection &&
aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
conference_mode == o.conference_mode &&
experimental_agc == o.experimental_agc &&
experimental_aec == o.experimental_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 &&
rx_agc_target_dbov == o.rx_agc_target_dbov &&
rx_agc_digital_compression_gain == o.rx_agc_digital_compression_gain &&
rx_agc_limiter == o.rx_agc_limiter &&
recording_sample_rate == o.recording_sample_rate &&
playout_sample_rate == o.playout_sample_rate &&
dscp == o.dscp &&
opus_fec == o.opus_fec;
}
std::string ToString() const {
std::ostringstream ost;
ost << "AudioOptions {";
ost << ToStringIfSet("aec", echo_cancellation);
ost << ToStringIfSet("agc", auto_gain_control);
ost << ToStringIfSet("rx_agc", rx_auto_gain_control);
ost << ToStringIfSet("ns", noise_suppression);
ost << ToStringIfSet("hf", highpass_filter);
ost << ToStringIfSet("swap", stereo_swapping);
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("experimental_aec", experimental_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("rx_agc_target_dbov", rx_agc_target_dbov);
ost << ToStringIfSet("rx_agc_digital_compression_gain",
rx_agc_digital_compression_gain);
ost << ToStringIfSet("rx_agc_limiter", rx_agc_limiter);
ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
ost << ToStringIfSet("dscp", dscp);
ost << ToStringIfSet("opus_fec", opus_fec);
ost << "}";
return ost.str();
}
// Audio processing that attempts to filter away the output signal from
// later inbound pickup.
Settable<bool> echo_cancellation;
// Audio processing to adjust the sensitivity of the local mic dynamically.
Settable<bool> auto_gain_control;
// Audio processing to apply gain to the remote audio.
Settable<bool> rx_auto_gain_control;
// Audio processing to filter out background noise.
Settable<bool> noise_suppression;
// Audio processing to remove background noise of lower frequencies.
Settable<bool> highpass_filter;
// Audio processing to swap the left and right channels.
Settable<bool> stereo_swapping;
// Audio processing to detect typing.
Settable<bool> typing_detection;
Settable<bool> aecm_generate_comfort_noise;
Settable<bool> conference_mode;
Settable<int> adjust_agc_delta;
Settable<bool> experimental_agc;
Settable<bool> experimental_aec;
Settable<bool> experimental_ns;
Settable<bool> aec_dump;
// Note that tx_agc_* only applies to non-experimental AGC.
Settable<uint16> tx_agc_target_dbov;
Settable<uint16> tx_agc_digital_compression_gain;
Settable<bool> tx_agc_limiter;
Settable<uint16> rx_agc_target_dbov;
Settable<uint16> rx_agc_digital_compression_gain;
Settable<bool> rx_agc_limiter;
Settable<uint32> recording_sample_rate;
Settable<uint32> playout_sample_rate;
// Set DSCP value for packet sent from audio channel.
Settable<bool> dscp;
// Set Opus FEC
Settable<bool> opus_fec;
};
// 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 {
enum HighestBitrate {
NORMAL,
HIGH,
VERY_HIGH
};
VideoOptions() {
process_adaptation_threshhold.Set(kProcessCpuThreshold);
system_low_adaptation_threshhold.Set(kLowSystemCpuThreshold);
system_high_adaptation_threshhold.Set(kHighSystemCpuThreshold);
unsignalled_recv_stream_limit.Set(kNumDefaultUnsignalledVideoRecvStreams);
}
void SetAll(const VideoOptions& change) {
adapt_input_to_encoder.SetFrom(change.adapt_input_to_encoder);
adapt_input_to_cpu_usage.SetFrom(change.adapt_input_to_cpu_usage);
adapt_cpu_with_smoothing.SetFrom(change.adapt_cpu_with_smoothing);
adapt_view_switch.SetFrom(change.adapt_view_switch);
video_adapt_third.SetFrom(change.video_adapt_third);
video_noise_reduction.SetFrom(change.video_noise_reduction);
video_one_layer_screencast.SetFrom(change.video_one_layer_screencast);
video_high_bitrate.SetFrom(change.video_high_bitrate);
video_start_bitrate.SetFrom(change.video_start_bitrate);
video_temporal_layer_screencast.SetFrom(
change.video_temporal_layer_screencast);
video_leaky_bucket.SetFrom(change.video_leaky_bucket);
video_highest_bitrate.SetFrom(change.video_highest_bitrate);
cpu_overuse_detection.SetFrom(change.cpu_overuse_detection);
cpu_underuse_threshold.SetFrom(change.cpu_underuse_threshold);
cpu_overuse_threshold.SetFrom(change.cpu_overuse_threshold);
cpu_underuse_encode_rsd_threshold.SetFrom(
change.cpu_underuse_encode_rsd_threshold);
cpu_overuse_encode_rsd_threshold.SetFrom(
change.cpu_overuse_encode_rsd_threshold);
cpu_overuse_encode_usage.SetFrom(change.cpu_overuse_encode_usage);
conference_mode.SetFrom(change.conference_mode);
process_adaptation_threshhold.SetFrom(change.process_adaptation_threshhold);
system_low_adaptation_threshhold.SetFrom(
change.system_low_adaptation_threshhold);
system_high_adaptation_threshhold.SetFrom(
change.system_high_adaptation_threshhold);
buffered_mode_latency.SetFrom(change.buffered_mode_latency);
dscp.SetFrom(change.dscp);
suspend_below_min_bitrate.SetFrom(change.suspend_below_min_bitrate);
unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit);
use_simulcast_adapter.SetFrom(change.use_simulcast_adapter);
screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
use_improved_wifi_bandwidth_estimator.SetFrom(
change.use_improved_wifi_bandwidth_estimator);
use_payload_padding.SetFrom(change.use_payload_padding);
}
bool operator==(const VideoOptions& o) const {
return adapt_input_to_encoder == o.adapt_input_to_encoder &&
adapt_input_to_cpu_usage == o.adapt_input_to_cpu_usage &&
adapt_cpu_with_smoothing == o.adapt_cpu_with_smoothing &&
adapt_view_switch == o.adapt_view_switch &&
video_adapt_third == o.video_adapt_third &&
video_noise_reduction == o.video_noise_reduction &&
video_one_layer_screencast == o.video_one_layer_screencast &&
video_high_bitrate == o.video_high_bitrate &&
video_start_bitrate == o.video_start_bitrate &&
video_temporal_layer_screencast == o.video_temporal_layer_screencast &&
video_leaky_bucket == o.video_leaky_bucket &&
video_highest_bitrate == o.video_highest_bitrate &&
cpu_overuse_detection == o.cpu_overuse_detection &&
cpu_underuse_threshold == o.cpu_underuse_threshold &&
cpu_overuse_threshold == o.cpu_overuse_threshold &&
cpu_underuse_encode_rsd_threshold ==
o.cpu_underuse_encode_rsd_threshold &&
cpu_overuse_encode_rsd_threshold ==
o.cpu_overuse_encode_rsd_threshold &&
cpu_overuse_encode_usage == o.cpu_overuse_encode_usage &&
conference_mode == o.conference_mode &&
process_adaptation_threshhold == o.process_adaptation_threshhold &&
system_low_adaptation_threshhold ==
o.system_low_adaptation_threshhold &&
system_high_adaptation_threshhold ==
o.system_high_adaptation_threshhold &&
buffered_mode_latency == o.buffered_mode_latency &&
dscp == o.dscp &&
suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit &&
use_simulcast_adapter == o.use_simulcast_adapter &&
screencast_min_bitrate == o.screencast_min_bitrate &&
use_improved_wifi_bandwidth_estimator ==
o.use_improved_wifi_bandwidth_estimator &&
use_payload_padding == o.use_payload_padding;
}
std::string ToString() const {
std::ostringstream ost;
ost << "VideoOptions {";
ost << ToStringIfSet("encoder adaption", adapt_input_to_encoder);
ost << ToStringIfSet("cpu adaption", adapt_input_to_cpu_usage);
ost << ToStringIfSet("cpu adaptation smoothing", adapt_cpu_with_smoothing);
ost << ToStringIfSet("adapt view switch", adapt_view_switch);
ost << ToStringIfSet("video adapt third", video_adapt_third);
ost << ToStringIfSet("noise reduction", video_noise_reduction);
ost << ToStringIfSet("1 layer screencast", video_one_layer_screencast);
ost << ToStringIfSet("high bitrate", video_high_bitrate);
ost << ToStringIfSet("start bitrate", video_start_bitrate);
ost << ToStringIfSet("video temporal layer screencast",
video_temporal_layer_screencast);
ost << ToStringIfSet("leaky bucket", video_leaky_bucket);
ost << ToStringIfSet("highest video bitrate", video_highest_bitrate);
ost << ToStringIfSet("cpu overuse detection", cpu_overuse_detection);
ost << ToStringIfSet("cpu underuse threshold", cpu_underuse_threshold);
ost << ToStringIfSet("cpu overuse threshold", cpu_overuse_threshold);
ost << ToStringIfSet("cpu underuse encode rsd threshold",
cpu_underuse_encode_rsd_threshold);
ost << ToStringIfSet("cpu overuse encode rsd threshold",
cpu_overuse_encode_rsd_threshold);
ost << ToStringIfSet("cpu overuse encode usage",
cpu_overuse_encode_usage);
ost << ToStringIfSet("conference mode", conference_mode);
ost << ToStringIfSet("process", process_adaptation_threshhold);
ost << ToStringIfSet("low", system_low_adaptation_threshhold);
ost << ToStringIfSet("high", system_high_adaptation_threshhold);
ost << ToStringIfSet("buffered mode latency", buffered_mode_latency);
ost << ToStringIfSet("dscp", dscp);
ost << ToStringIfSet("suspend below min bitrate",
suspend_below_min_bitrate);
ost << ToStringIfSet("num channels for early receive",
unsignalled_recv_stream_limit);
ost << ToStringIfSet("use simulcast adapter", use_simulcast_adapter);
ost << ToStringIfSet("screencast min bitrate", screencast_min_bitrate);
ost << ToStringIfSet("improved wifi bwe",
use_improved_wifi_bandwidth_estimator);
ost << ToStringIfSet("payload padding", use_payload_padding);
ost << "}";
return ost.str();
}
// Encoder adaption, which is the gd callback in LMI, and TBA in WebRTC.
Settable<bool> adapt_input_to_encoder;
// Enable CPU adaptation?
Settable<bool> adapt_input_to_cpu_usage;
// Enable CPU adaptation smoothing?
Settable<bool> adapt_cpu_with_smoothing;
// Enable Adapt View Switch?
Settable<bool> adapt_view_switch;
// Enable video adapt third?
Settable<bool> video_adapt_third;
// Enable denoising?
Settable<bool> video_noise_reduction;
// Experimental: Enable one layer screencast?
Settable<bool> video_one_layer_screencast;
// Experimental: Enable WebRtc higher bitrate?
Settable<bool> video_high_bitrate;
// Experimental: Enable WebRtc higher start bitrate?
Settable<int> video_start_bitrate;
// Experimental: Enable WebRTC layered screencast.
Settable<bool> video_temporal_layer_screencast;
// Enable WebRTC leaky bucket when sending media packets.
Settable<bool> video_leaky_bucket;
// Set highest bitrate mode for video.
Settable<HighestBitrate> video_highest_bitrate;
// Enable WebRTC Cpu Overuse Detection, which is a new version of the CPU
// adaptation algorithm. So this option will override the
// |adapt_input_to_cpu_usage|.
Settable<bool> cpu_overuse_detection;
// Low threshold (t1) for cpu overuse adaptation. (Adapt up)
// Metric: encode usage (m1). m1 < t1 => underuse.
Settable<int> cpu_underuse_threshold;
// High threshold (t1) for cpu overuse adaptation. (Adapt down)
// Metric: encode usage (m1). m1 > t1 => overuse.
Settable<int> cpu_overuse_threshold;
// Low threshold (t2) for cpu overuse adaptation. (Adapt up)
// Metric: relative standard deviation of encode time (m2).
// Optional threshold. If set, (m1 < t1 && m2 < t2) => underuse.
// Note: t2 will have no effect if t1 is not set.
Settable<int> cpu_underuse_encode_rsd_threshold;
// High threshold (t2) for cpu overuse adaptation. (Adapt down)
// Metric: relative standard deviation of encode time (m2).
// Optional threshold. If set, (m1 > t1 || m2 > t2) => overuse.
// Note: t2 will have no effect if t1 is not set.
Settable<int> cpu_overuse_encode_rsd_threshold;
// Use encode usage for cpu detection.
Settable<bool> cpu_overuse_encode_usage;
// Use conference mode?
Settable<bool> conference_mode;
// Threshhold for process cpu adaptation. (Process limit)
SettablePercent process_adaptation_threshhold;
// Low threshhold for cpu adaptation. (Adapt up)
SettablePercent system_low_adaptation_threshhold;
// High threshhold for cpu adaptation. (Adapt down)
SettablePercent system_high_adaptation_threshhold;
// Specify buffered mode latency in milliseconds.
Settable<int> buffered_mode_latency;
// Set DSCP value for packet sent from video channel.
Settable<bool> dscp;
// Enable WebRTC suspension of video. No video frames will be sent when the
// bitrate is below the configured minimum bitrate.
Settable<bool> suspend_below_min_bitrate;
// Limit on the number of early receive channels that can be created.
Settable<int> unsignalled_recv_stream_limit;
// Enable use of simulcast adapter.
Settable<bool> use_simulcast_adapter;
// Force screencast to use a minimum bitrate
Settable<int> screencast_min_bitrate;
// Enable improved bandwidth estiamtor on wifi.
Settable<bool> use_improved_wifi_bandwidth_estimator;
// Enable payload padding.
Settable<bool> use_payload_padding;
};
// A class for playing out soundclips.
class SoundclipMedia {
public:
enum SoundclipFlags {
SF_LOOP = 1,
};
virtual ~SoundclipMedia() {}
// Plays a sound out to the speakers with the given audio stream. The stream
// must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
// on this SoundclipMedia, it is stopped. If clip is NULL, nothing is played.
// Returns whether it was successful.
virtual bool PlaySound(const char *clip, int len, int flags) = 0;
};
struct RtpHeaderExtension {
RtpHeaderExtension() : id(0) {}
RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {}
std::string uri;
int id;
// TODO(juberti): SendRecv direction;
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;
}
};
// 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;
}
enum MediaChannelOptions {
// Tune the stream for conference mode.
OPT_CONFERENCE = 0x0001
};
enum VoiceMediaChannelOptions {
// Tune the audio stream for vcs with different target levels.
OPT_AGC_MINUS_10DB = 0x80000000
};
// DTMF flags to control if a DTMF tone should be played and/or sent.
enum DtmfFlags {
DF_PLAY = 0x01,
DF_SEND = 0x02,
};
class MediaChannel : public sigslot::has_slots<> {
public:
class NetworkInterface {
public:
enum SocketType { ST_RTP, ST_RTCP };
virtual bool SendPacket(
rtc::Buffer* packet,
rtc::DiffServCodePoint dscp = rtc::DSCP_NO_CHANGE) = 0;
virtual bool SendRtcp(
rtc::Buffer* packet,
rtc::DiffServCodePoint dscp = rtc::DSCP_NO_CHANGE) = 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 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 ssrc) = 0;
// Mutes the channel.
virtual bool MuteStream(uint32 ssrc, bool on) = 0;
// Sets the RTP extension headers and IDs to use when sending RTP.
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) = 0;
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) = 0;
// Returns the absoulte sendtime extension id value from media channel.
virtual int GetRtpSendTimeExtnId() const {
return -1;
}
// Sets the initial bandwidth to use when sending starts.
virtual bool SetStartSendBandwidth(int bps) = 0;
// Sets the maximum allowed bandwidth to use when sending data.
virtual bool SetMaxSendBandwidth(int bps) = 0;
// Base method to send packet using NetworkInterface.
bool SendPacket(rtc::Buffer* packet) {
return DoSendPacket(packet, false);
}
bool SendRtcp(rtc::Buffer* packet) {
return DoSendPacket(packet, true);
}
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) {
rtc::CritScope cs(&network_interface_crit_);
if (!network_interface_)
return false;
return (!rtcp) ? network_interface_->SendPacket(packet) :
network_interface_->SendRtcp(packet);
}
// |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_RINGBACKTONE,
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 ssrc;
double timestamp; // NTP timestamp, represented as seconds since epoch.
};
struct SsrcReceiverInfo {
SsrcReceiverInfo()
: ssrc(0),
timestamp(0) {
}
uint32 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 ssrc) {
SsrcSenderInfo stat;
stat.ssrc = ssrc;
add_ssrc(stat);
}
// Utility accessor for clients that are only interested in ssrc numbers.
std::vector<uint32> ssrcs() const {
std::vector<uint32> 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 ssrc() const {
if (local_stats.size() > 0) {
return local_stats[0].ssrc;
} else {
return 0;
}
}
int64 bytes_sent;
int packets_sent;
int packets_lost;
float fraction_lost;
int 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 ssrc) {
SsrcReceiverInfo stat;
stat.ssrc = ssrc;
add_ssrc(stat);
}
std::vector<uint32> ssrcs() const {
std::vector<uint32> 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 ssrc() const {
if (local_stats.size() > 0) {
return local_stats[0].ssrc;
} else {
return 0;
}
}
int64 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),
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 speech inserted through pre-emptive expansion
float 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 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),
capture_jitter_ms(0),
avg_encode_ms(0),
encode_usage_percent(0),
encode_rsd(0),
capture_queue_delay_ms_per_s(0) {
}
std::vector<SsrcGroup> ssrc_groups;
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 capture_jitter_ms;
int avg_encode_ms;
int encode_usage_percent;
int encode_rsd;
int capture_queue_delay_ms_per_s;
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;
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 capture_start_ntp_time_ms;
};
struct DataSenderInfo : public MediaSenderInfo {
DataSenderInfo()
: ssrc(0) {
}
uint32 ssrc;
};
struct DataReceiverInfo : public MediaReceiverInfo {
DataReceiverInfo()
: ssrc(0) {
}
uint32 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),
total_received_propagation_delta_ms(0) {
}
int available_send_bandwidth;
int available_recv_bandwidth;
int target_enc_bitrate;
int actual_enc_bitrate;
int retransmit_bitrate;
int transmit_bitrate;
int bucket_delay;
// The following stats are only valid when
// StatsOptions::include_received_propagation_stats is true.
int total_received_propagation_delta_ms;
std::vector<int> recent_received_propagation_delta_ms;
std::vector<int64> recent_received_packet_group_arrival_time_ms;
};
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 StatsOptions {
StatsOptions() : include_received_propagation_stats(false) {}
bool include_received_propagation_stats;
};
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() {}
// Sets the codecs/payload types to be used for incoming media.
virtual bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) = 0;
// Sets the codecs/payload types to be used for outgoing media.
virtual bool SetSendCodecs(const std::vector<AudioCodec>& codecs) = 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;
// Sets the renderer object to be used for the specified remote audio stream.
virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) = 0;
// Sets the renderer object to be used for the specified local audio stream.
virtual bool SetLocalRenderer(uint32 ssrc, 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 left and right scale for speaker output volume of the specified ssrc.
virtual bool SetOutputScaling(uint32 ssrc, double left, double right) = 0;
// Get left and right scale for speaker output volume of the specified ssrc.
virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) = 0;
// Specifies a ringback tone to be played during call setup.
virtual bool SetRingbackTone(const char *buf, int len) = 0;
// Plays or stops the aforementioned ringback tone
virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) = 0;
// Returns if the telephone-event has been negotiated.
virtual bool CanInsertDtmf() { return false; }
// Send and/or play a DTMF |event| according to the |flags|.
// The DTMF out-of-band signal will be used on sending.
// 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 ssrc, int event, int duration, int flags) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0;
// Gets last reported error for this media channel.
virtual void GetLastMediaError(uint32* ssrc,
VoiceMediaChannel::Error* error) {
ASSERT(error != NULL);
*error = ERROR_NONE;
}
// Sets the media options to use.
virtual bool SetOptions(const AudioOptions& options) = 0;
virtual bool GetOptions(AudioOptions* options) const = 0;
// Signal errors from MediaChannel. Arguments are:
// ssrc(uint32), and error(VoiceMediaChannel::Error).
sigslot::signal2<uint32, VoiceMediaChannel::Error> SignalMediaError;
};
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() : renderer_(NULL) {}
virtual ~VideoMediaChannel() {}
// Sets the codecs/payload types to be used for incoming media.
virtual bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) = 0;
// Sets the codecs/payload types to be used for outgoing media.
virtual bool SetSendCodecs(const std::vector<VideoCodec>& codecs) = 0;
// Gets the currently set codecs/payload types to be used for outgoing media.
virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
// Sets the format of a specified outgoing stream.
virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) = 0;
// Starts or stops playout of received video.
virtual bool SetRender(bool render) = 0;
// Starts or stops transmission (and potentially capture) of local video.
virtual bool SetSend(bool send) = 0;
// Sets the renderer object to be used for the specified stream.
// If SSRC is 0, the renderer is used for the 'default' stream.
virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) = 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 ssrc, VideoCapturer* capturer) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(const StatsOptions& options, VideoMediaInfo* info) = 0;
// This is needed for MediaMonitor to use the same template for voice, video
// and data MediaChannels.
bool GetStats(VideoMediaInfo* info) {
return GetStats(StatsOptions(), info);
}
// Send an intra frame to the receivers.
virtual bool SendIntraFrame() = 0;
// Reuqest each of the remote senders to send an intra frame.
virtual bool RequestIntraFrame() = 0;
// Sets the media options to use.
virtual bool SetOptions(const VideoOptions& options) = 0;
virtual bool GetOptions(VideoOptions* options) const = 0;
virtual void UpdateAspectRatio(int ratio_w, int ratio_h) = 0;
// Signal errors from MediaChannel. Arguments are:
// ssrc(uint32), and error(VideoMediaChannel::Error).
sigslot::signal2<uint32, Error> SignalMediaError;
protected:
VideoRenderer *renderer_;
};
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 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 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 };
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 SetSendCodecs(const std::vector<DataCodec>& codecs) = 0;
virtual bool SetRecvCodecs(const std::vector<DataCodec>& codecs) = 0;
virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
// 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 errors from MediaChannel. Arguments are:
// ssrc(uint32), and error(DataMediaChannel::Error).
sigslot::signal2<uint32, DataMediaChannel::Error> SignalMediaError;
// 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> SignalStreamClosedRemotely;
};
} // namespace cricket
#endif // TALK_MEDIA_BASE_MEDIACHANNEL_H_
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