2013-07-10 02:45:36 +02:00
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/*
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* libjingle
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* Copyright 2004 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_MEDIA_BASE_MEDIACHANNEL_H_
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#define TALK_MEDIA_BASE_MEDIACHANNEL_H_
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#include <string>
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#include <vector>
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#include "talk/base/basictypes.h"
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#include "talk/base/buffer.h"
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2013-09-23 22:34:45 +02:00
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#include "talk/base/dscp.h"
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2013-07-10 02:45:36 +02:00
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#include "talk/base/logging.h"
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#include "talk/base/sigslot.h"
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#include "talk/base/socket.h"
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#include "talk/base/window.h"
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#include "talk/media/base/codec.h"
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#include "talk/media/base/constants.h"
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#include "talk/media/base/streamparams.h"
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// TODO(juberti): re-evaluate this include
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#include "talk/session/media/audiomonitor.h"
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namespace talk_base {
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class Buffer;
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class RateLimiter;
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class Timing;
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}
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namespace cricket {
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class AudioRenderer;
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struct RtpHeader;
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class ScreencastId;
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struct VideoFormat;
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class VideoCapturer;
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class VideoRenderer;
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const int kMinRtpHeaderExtensionId = 1;
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const int kMaxRtpHeaderExtensionId = 255;
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const int kScreencastDefaultFps = 5;
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2014-03-27 18:03:58 +01:00
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const int kHighStartBitrate = 1500;
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2013-07-10 02:45:36 +02:00
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// Used in AudioOptions and VideoOptions to signify "unset" values.
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template <class T>
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class Settable {
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public:
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Settable() : set_(false), val_() {}
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explicit Settable(T val) : set_(true), val_(val) {}
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bool IsSet() const {
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return set_;
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}
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bool Get(T* out) const {
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*out = val_;
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return set_;
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}
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T GetWithDefaultIfUnset(const T& default_value) const {
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return set_ ? val_ : default_value;
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}
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virtual void Set(T val) {
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set_ = true;
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val_ = val;
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}
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void Clear() {
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Set(T());
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set_ = false;
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}
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void SetFrom(const Settable<T>& o) {
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// Set this value based on the value of o, iff o is set. If this value is
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// set and o is unset, the current value will be unchanged.
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T val;
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if (o.Get(&val)) {
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Set(val);
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}
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}
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std::string ToString() const {
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return set_ ? talk_base::ToString(val_) : "";
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}
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bool operator==(const Settable<T>& o) const {
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// Equal if both are unset with any value or both set with the same value.
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return (set_ == o.set_) && (!set_ || (val_ == o.val_));
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}
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bool operator!=(const Settable<T>& o) const {
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return !operator==(o);
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}
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protected:
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void InitializeValue(const T &val) {
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val_ = val;
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}
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private:
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bool set_;
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T val_;
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};
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class SettablePercent : public Settable<float> {
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public:
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virtual void Set(float val) {
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if (val < 0) {
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val = 0;
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}
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if (val > 1.0) {
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val = 1.0;
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}
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Settable<float>::Set(val);
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}
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};
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template <class T>
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static std::string ToStringIfSet(const char* key, const Settable<T>& val) {
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std::string str;
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if (val.IsSet()) {
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str = key;
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str += ": ";
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str += val.ToString();
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str += ", ";
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}
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return str;
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}
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// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
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// Used to be flags, but that makes it hard to selectively apply options.
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// We are moving all of the setting of options to structs like this,
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// but some things currently still use flags.
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struct AudioOptions {
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void SetAll(const AudioOptions& change) {
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echo_cancellation.SetFrom(change.echo_cancellation);
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auto_gain_control.SetFrom(change.auto_gain_control);
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2013-10-25 23:18:33 +02:00
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rx_auto_gain_control.SetFrom(change.rx_auto_gain_control);
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2013-07-10 02:45:36 +02:00
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noise_suppression.SetFrom(change.noise_suppression);
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highpass_filter.SetFrom(change.highpass_filter);
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stereo_swapping.SetFrom(change.stereo_swapping);
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typing_detection.SetFrom(change.typing_detection);
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2013-10-25 23:18:33 +02:00
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aecm_generate_comfort_noise.SetFrom(change.aecm_generate_comfort_noise);
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2013-07-10 02:45:36 +02:00
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conference_mode.SetFrom(change.conference_mode);
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adjust_agc_delta.SetFrom(change.adjust_agc_delta);
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experimental_agc.SetFrom(change.experimental_agc);
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experimental_aec.SetFrom(change.experimental_aec);
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2014-02-07 20:03:26 +01:00
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experimental_ns.SetFrom(change.experimental_ns);
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2013-07-10 02:45:36 +02:00
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aec_dump.SetFrom(change.aec_dump);
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2013-10-25 23:18:33 +02:00
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tx_agc_target_dbov.SetFrom(change.tx_agc_target_dbov);
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tx_agc_digital_compression_gain.SetFrom(
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change.tx_agc_digital_compression_gain);
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tx_agc_limiter.SetFrom(change.tx_agc_limiter);
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rx_agc_target_dbov.SetFrom(change.rx_agc_target_dbov);
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rx_agc_digital_compression_gain.SetFrom(
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change.rx_agc_digital_compression_gain);
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rx_agc_limiter.SetFrom(change.rx_agc_limiter);
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recording_sample_rate.SetFrom(change.recording_sample_rate);
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playout_sample_rate.SetFrom(change.playout_sample_rate);
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2013-10-31 16:40:38 +01:00
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dscp.SetFrom(change.dscp);
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2013-07-10 02:45:36 +02:00
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}
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bool operator==(const AudioOptions& o) const {
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return echo_cancellation == o.echo_cancellation &&
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auto_gain_control == o.auto_gain_control &&
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2013-10-25 23:18:33 +02:00
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rx_auto_gain_control == o.rx_auto_gain_control &&
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2013-07-10 02:45:36 +02:00
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noise_suppression == o.noise_suppression &&
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highpass_filter == o.highpass_filter &&
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stereo_swapping == o.stereo_swapping &&
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typing_detection == o.typing_detection &&
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2013-10-25 23:18:33 +02:00
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aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
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2013-07-10 02:45:36 +02:00
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conference_mode == o.conference_mode &&
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experimental_agc == o.experimental_agc &&
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experimental_aec == o.experimental_aec &&
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2014-02-07 20:03:26 +01:00
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experimental_ns == o.experimental_ns &&
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2013-07-10 02:45:36 +02:00
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adjust_agc_delta == o.adjust_agc_delta &&
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2013-10-25 23:18:33 +02:00
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aec_dump == o.aec_dump &&
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tx_agc_target_dbov == o.tx_agc_target_dbov &&
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tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
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tx_agc_limiter == o.tx_agc_limiter &&
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rx_agc_target_dbov == o.rx_agc_target_dbov &&
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rx_agc_digital_compression_gain == o.rx_agc_digital_compression_gain &&
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rx_agc_limiter == o.rx_agc_limiter &&
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recording_sample_rate == o.recording_sample_rate &&
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2013-10-31 16:40:38 +01:00
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playout_sample_rate == o.playout_sample_rate &&
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dscp == o.dscp;
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2013-07-10 02:45:36 +02:00
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}
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std::string ToString() const {
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std::ostringstream ost;
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ost << "AudioOptions {";
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ost << ToStringIfSet("aec", echo_cancellation);
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ost << ToStringIfSet("agc", auto_gain_control);
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2013-10-25 23:18:33 +02:00
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ost << ToStringIfSet("rx_agc", rx_auto_gain_control);
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2013-07-10 02:45:36 +02:00
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ost << ToStringIfSet("ns", noise_suppression);
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ost << ToStringIfSet("hf", highpass_filter);
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ost << ToStringIfSet("swap", stereo_swapping);
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ost << ToStringIfSet("typing", typing_detection);
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2013-10-25 23:18:33 +02:00
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ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
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2013-07-10 02:45:36 +02:00
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ost << ToStringIfSet("conference", conference_mode);
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ost << ToStringIfSet("agc_delta", adjust_agc_delta);
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ost << ToStringIfSet("experimental_agc", experimental_agc);
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ost << ToStringIfSet("experimental_aec", experimental_aec);
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2014-02-07 20:03:26 +01:00
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ost << ToStringIfSet("experimental_ns", experimental_ns);
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2013-07-10 02:45:36 +02:00
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ost << ToStringIfSet("aec_dump", aec_dump);
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2013-10-25 23:18:33 +02:00
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ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
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ost << ToStringIfSet("tx_agc_digital_compression_gain",
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tx_agc_digital_compression_gain);
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ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
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ost << ToStringIfSet("rx_agc_target_dbov", rx_agc_target_dbov);
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ost << ToStringIfSet("rx_agc_digital_compression_gain",
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rx_agc_digital_compression_gain);
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ost << ToStringIfSet("rx_agc_limiter", rx_agc_limiter);
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ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
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ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
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2013-10-31 16:40:38 +01:00
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ost << ToStringIfSet("dscp", dscp);
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2013-07-10 02:45:36 +02:00
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ost << "}";
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return ost.str();
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}
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// Audio processing that attempts to filter away the output signal from
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// later inbound pickup.
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Settable<bool> echo_cancellation;
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// Audio processing to adjust the sensitivity of the local mic dynamically.
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Settable<bool> auto_gain_control;
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2013-10-25 23:18:33 +02:00
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// Audio processing to apply gain to the remote audio.
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Settable<bool> rx_auto_gain_control;
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2013-07-10 02:45:36 +02:00
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// Audio processing to filter out background noise.
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Settable<bool> noise_suppression;
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// Audio processing to remove background noise of lower frequencies.
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Settable<bool> highpass_filter;
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// Audio processing to swap the left and right channels.
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Settable<bool> stereo_swapping;
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// Audio processing to detect typing.
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Settable<bool> typing_detection;
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2013-10-25 23:18:33 +02:00
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Settable<bool> aecm_generate_comfort_noise;
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2013-07-10 02:45:36 +02:00
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Settable<bool> conference_mode;
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Settable<int> adjust_agc_delta;
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Settable<bool> experimental_agc;
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Settable<bool> experimental_aec;
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2014-02-07 20:03:26 +01:00
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Settable<bool> experimental_ns;
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2013-07-10 02:45:36 +02:00
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Settable<bool> aec_dump;
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2013-10-25 23:18:33 +02:00
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// Note that tx_agc_* only applies to non-experimental AGC.
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Settable<uint16> tx_agc_target_dbov;
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Settable<uint16> tx_agc_digital_compression_gain;
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Settable<bool> tx_agc_limiter;
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Settable<uint16> rx_agc_target_dbov;
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Settable<uint16> rx_agc_digital_compression_gain;
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Settable<bool> rx_agc_limiter;
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Settable<uint32> recording_sample_rate;
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Settable<uint32> playout_sample_rate;
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2013-10-31 16:40:38 +01:00
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// Set DSCP value for packet sent from audio channel.
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Settable<bool> dscp;
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2013-07-10 02:45:36 +02:00
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};
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// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
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// Used to be flags, but that makes it hard to selectively apply options.
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// We are moving all of the setting of options to structs like this,
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// but some things currently still use flags.
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struct VideoOptions {
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2014-03-13 19:51:34 +01:00
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enum HighestBitrate {
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NORMAL,
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HIGH,
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VERY_HIGH
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};
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2013-07-10 02:45:36 +02:00
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VideoOptions() {
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process_adaptation_threshhold.Set(kProcessCpuThreshold);
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system_low_adaptation_threshhold.Set(kLowSystemCpuThreshold);
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system_high_adaptation_threshhold.Set(kHighSystemCpuThreshold);
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2014-02-27 18:52:04 +01:00
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unsignalled_recv_stream_limit.Set(kNumDefaultUnsignalledVideoRecvStreams);
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2013-07-10 02:45:36 +02:00
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}
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void SetAll(const VideoOptions& change) {
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adapt_input_to_encoder.SetFrom(change.adapt_input_to_encoder);
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adapt_input_to_cpu_usage.SetFrom(change.adapt_input_to_cpu_usage);
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2013-07-22 23:07:49 +02:00
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adapt_cpu_with_smoothing.SetFrom(change.adapt_cpu_with_smoothing);
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2013-07-10 02:45:36 +02:00
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adapt_view_switch.SetFrom(change.adapt_view_switch);
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2013-08-30 23:24:16 +02:00
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video_adapt_third.SetFrom(change.video_adapt_third);
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2013-07-10 02:45:36 +02:00
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video_noise_reduction.SetFrom(change.video_noise_reduction);
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video_one_layer_screencast.SetFrom(change.video_one_layer_screencast);
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video_high_bitrate.SetFrom(change.video_high_bitrate);
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2014-03-24 18:01:50 +01:00
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|
|
video_start_bitrate.SetFrom(change.video_start_bitrate);
|
2013-07-10 02:45:36 +02:00
|
|
|
video_temporal_layer_screencast.SetFrom(
|
|
|
|
change.video_temporal_layer_screencast);
|
2013-10-25 23:18:33 +02:00
|
|
|
video_temporal_layer_realtime.SetFrom(
|
|
|
|
change.video_temporal_layer_realtime);
|
2013-07-10 02:45:36 +02:00
|
|
|
video_leaky_bucket.SetFrom(change.video_leaky_bucket);
|
2014-03-13 19:51:34 +01:00
|
|
|
video_highest_bitrate.SetFrom(change.video_highest_bitrate);
|
2013-08-30 23:24:16 +02:00
|
|
|
cpu_overuse_detection.SetFrom(change.cpu_overuse_detection);
|
2014-03-18 15:36:23 +01:00
|
|
|
cpu_underuse_threshold.SetFrom(change.cpu_underuse_threshold);
|
|
|
|
cpu_overuse_threshold.SetFrom(change.cpu_overuse_threshold);
|
2014-03-26 23:44:28 +01:00
|
|
|
cpu_overuse_encode_usage.SetFrom(change.cpu_overuse_encode_usage);
|
2013-07-10 02:45:36 +02:00
|
|
|
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);
|
2013-10-30 06:18:12 +01:00
|
|
|
lower_min_bitrate.SetFrom(change.lower_min_bitrate);
|
2013-10-31 16:40:38 +01:00
|
|
|
dscp.SetFrom(change.dscp);
|
2013-12-05 01:24:06 +01:00
|
|
|
suspend_below_min_bitrate.SetFrom(change.suspend_below_min_bitrate);
|
2014-02-27 18:52:04 +01:00
|
|
|
unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit);
|
2014-03-11 22:07:25 +01:00
|
|
|
use_simulcast_adapter.SetFrom(change.use_simulcast_adapter);
|
2014-03-26 02:17:30 +01:00
|
|
|
skip_encoding_unused_streams.SetFrom(change.skip_encoding_unused_streams);
|
|
|
|
screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
|
|
|
|
use_improved_wifi_bandwidth_estimator.SetFrom(
|
|
|
|
change.use_improved_wifi_bandwidth_estimator);
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
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 &&
|
2013-07-22 23:07:49 +02:00
|
|
|
adapt_cpu_with_smoothing == o.adapt_cpu_with_smoothing &&
|
2013-07-10 02:45:36 +02:00
|
|
|
adapt_view_switch == o.adapt_view_switch &&
|
2013-08-30 23:24:16 +02:00
|
|
|
video_adapt_third == o.video_adapt_third &&
|
2013-07-10 02:45:36 +02:00
|
|
|
video_noise_reduction == o.video_noise_reduction &&
|
|
|
|
video_one_layer_screencast == o.video_one_layer_screencast &&
|
|
|
|
video_high_bitrate == o.video_high_bitrate &&
|
2014-03-24 18:01:50 +01:00
|
|
|
video_start_bitrate == o.video_start_bitrate &&
|
2013-07-10 02:45:36 +02:00
|
|
|
video_temporal_layer_screencast == o.video_temporal_layer_screencast &&
|
2013-10-25 23:18:33 +02:00
|
|
|
video_temporal_layer_realtime == o.video_temporal_layer_realtime &&
|
2013-07-10 02:45:36 +02:00
|
|
|
video_leaky_bucket == o.video_leaky_bucket &&
|
2014-03-13 19:51:34 +01:00
|
|
|
video_highest_bitrate == o.video_highest_bitrate &&
|
2013-08-30 23:24:16 +02:00
|
|
|
cpu_overuse_detection == o.cpu_overuse_detection &&
|
2014-03-18 15:36:23 +01:00
|
|
|
cpu_underuse_threshold == o.cpu_underuse_threshold &&
|
|
|
|
cpu_overuse_threshold == o.cpu_overuse_threshold &&
|
2014-03-26 23:44:28 +01:00
|
|
|
cpu_overuse_encode_usage == o.cpu_overuse_encode_usage &&
|
2013-07-10 02:45:36 +02:00
|
|
|
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 &&
|
2013-10-30 06:18:12 +01:00
|
|
|
buffered_mode_latency == o.buffered_mode_latency &&
|
2013-10-31 16:40:38 +01:00
|
|
|
lower_min_bitrate == o.lower_min_bitrate &&
|
2013-12-05 01:24:06 +01:00
|
|
|
dscp == o.dscp &&
|
2014-02-27 18:52:04 +01:00
|
|
|
suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
|
2014-03-11 22:07:25 +01:00
|
|
|
unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit &&
|
2014-03-26 02:17:30 +01:00
|
|
|
use_simulcast_adapter == o.use_simulcast_adapter &&
|
|
|
|
skip_encoding_unused_streams == o.skip_encoding_unused_streams &&
|
2014-03-26 03:00:10 +01:00
|
|
|
screencast_min_bitrate == o.screencast_min_bitrate &&
|
2014-03-26 02:17:30 +01:00
|
|
|
use_improved_wifi_bandwidth_estimator ==
|
|
|
|
o.use_improved_wifi_bandwidth_estimator;
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
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);
|
2013-07-22 23:07:49 +02:00
|
|
|
ost << ToStringIfSet("cpu adaptation smoothing", adapt_cpu_with_smoothing);
|
2013-07-10 02:45:36 +02:00
|
|
|
ost << ToStringIfSet("adapt view switch", adapt_view_switch);
|
2013-08-30 23:24:16 +02:00
|
|
|
ost << ToStringIfSet("video adapt third", video_adapt_third);
|
2013-07-10 02:45:36 +02:00
|
|
|
ost << ToStringIfSet("noise reduction", video_noise_reduction);
|
2013-08-30 23:24:16 +02:00
|
|
|
ost << ToStringIfSet("1 layer screencast", video_one_layer_screencast);
|
2013-07-10 02:45:36 +02:00
|
|
|
ost << ToStringIfSet("high bitrate", video_high_bitrate);
|
2014-03-24 18:01:50 +01:00
|
|
|
ost << ToStringIfSet("start bitrate", video_start_bitrate);
|
2013-07-10 02:45:36 +02:00
|
|
|
ost << ToStringIfSet("video temporal layer screencast",
|
|
|
|
video_temporal_layer_screencast);
|
2013-10-25 23:18:33 +02:00
|
|
|
ost << ToStringIfSet("video temporal layer realtime",
|
|
|
|
video_temporal_layer_realtime);
|
2013-07-10 02:45:36 +02:00
|
|
|
ost << ToStringIfSet("leaky bucket", video_leaky_bucket);
|
2014-03-13 19:51:34 +01:00
|
|
|
ost << ToStringIfSet("highest video bitrate", video_highest_bitrate);
|
2013-08-30 23:24:16 +02:00
|
|
|
ost << ToStringIfSet("cpu overuse detection", cpu_overuse_detection);
|
2014-03-18 15:36:23 +01:00
|
|
|
ost << ToStringIfSet("cpu underuse threshold", cpu_underuse_threshold);
|
|
|
|
ost << ToStringIfSet("cpu overuse threshold", cpu_overuse_threshold);
|
2014-03-26 23:44:28 +01:00
|
|
|
ost << ToStringIfSet("cpu overuse encode usage",
|
|
|
|
cpu_overuse_encode_usage);
|
2013-07-10 02:45:36 +02:00
|
|
|
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);
|
2013-10-30 06:18:12 +01:00
|
|
|
ost << ToStringIfSet("lower min bitrate", lower_min_bitrate);
|
2013-10-31 16:40:38 +01:00
|
|
|
ost << ToStringIfSet("dscp", dscp);
|
2013-12-05 01:24:06 +01:00
|
|
|
ost << ToStringIfSet("suspend below min bitrate",
|
|
|
|
suspend_below_min_bitrate);
|
2014-02-27 18:52:04 +01:00
|
|
|
ost << ToStringIfSet("num channels for early receive",
|
|
|
|
unsignalled_recv_stream_limit);
|
2014-03-11 22:07:25 +01:00
|
|
|
ost << ToStringIfSet("use simulcast adapter", use_simulcast_adapter);
|
2014-03-26 02:17:30 +01:00
|
|
|
ost << ToStringIfSet("skip encoding unused streams",
|
|
|
|
skip_encoding_unused_streams);
|
|
|
|
ost << ToStringIfSet("screencast min bitrate", screencast_min_bitrate);
|
|
|
|
ost << ToStringIfSet("improved wifi bwe",
|
|
|
|
use_improved_wifi_bandwidth_estimator);
|
2013-07-10 02:45:36 +02:00
|
|
|
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;
|
2013-07-22 23:07:49 +02:00
|
|
|
// Enable CPU adaptation smoothing?
|
|
|
|
Settable<bool> adapt_cpu_with_smoothing;
|
2013-07-10 02:45:36 +02:00
|
|
|
// Enable Adapt View Switch?
|
|
|
|
Settable<bool> adapt_view_switch;
|
2013-08-30 23:24:16 +02:00
|
|
|
// Enable video adapt third?
|
|
|
|
Settable<bool> video_adapt_third;
|
2013-07-10 02:45:36 +02:00
|
|
|
// 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;
|
2014-03-24 18:01:50 +01:00
|
|
|
// Experimental: Enable WebRtc higher start bitrate?
|
|
|
|
Settable<int> video_start_bitrate;
|
2013-07-10 02:45:36 +02:00
|
|
|
// Experimental: Enable WebRTC layered screencast.
|
|
|
|
Settable<bool> video_temporal_layer_screencast;
|
2013-10-25 23:18:33 +02:00
|
|
|
// Experimental: Enable WebRTC temporal layer strategy for realtime video.
|
|
|
|
Settable<bool> video_temporal_layer_realtime;
|
2013-07-10 02:45:36 +02:00
|
|
|
// Enable WebRTC leaky bucket when sending media packets.
|
|
|
|
Settable<bool> video_leaky_bucket;
|
2014-03-13 19:51:34 +01:00
|
|
|
// Set highest bitrate mode for video.
|
2014-03-27 18:03:58 +01:00
|
|
|
Settable<HighestBitrate> video_highest_bitrate;
|
2013-08-30 23:24:16 +02:00
|
|
|
// 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;
|
2014-03-18 15:36:23 +01:00
|
|
|
// Low threshold for cpu overuse adaptation in ms. (Adapt up)
|
|
|
|
Settable<int> cpu_underuse_threshold;
|
|
|
|
// High threshold for cpu overuse adaptation in ms. (Adapt down)
|
|
|
|
Settable<int> cpu_overuse_threshold;
|
2014-03-26 23:44:28 +01:00
|
|
|
// Use encode usage for cpu detection.
|
|
|
|
Settable<bool> cpu_overuse_encode_usage;
|
2013-07-10 02:45:36 +02:00
|
|
|
// 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;
|
2013-10-30 06:18:12 +01:00
|
|
|
// Make minimum configured send bitrate even lower than usual, at 30kbit.
|
|
|
|
Settable<bool> lower_min_bitrate;
|
2013-10-31 16:40:38 +01:00
|
|
|
// Set DSCP value for packet sent from video channel.
|
|
|
|
Settable<bool> dscp;
|
2013-12-05 01:24:06 +01:00
|
|
|
// 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;
|
2014-02-27 18:52:04 +01:00
|
|
|
// Limit on the number of early receive channels that can be created.
|
|
|
|
Settable<int> unsignalled_recv_stream_limit;
|
2014-03-11 22:07:25 +01:00
|
|
|
// Enable use of simulcast adapter.
|
|
|
|
Settable<bool> use_simulcast_adapter;
|
2014-03-26 02:17:30 +01:00
|
|
|
// Enables the encoder to skip encoding stream not actually sent due to too
|
|
|
|
// low available bit rate.
|
|
|
|
Settable<bool> skip_encoding_unused_streams;
|
|
|
|
// 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;
|
2013-07-10 02:45:36 +02:00
|
|
|
};
|
|
|
|
|
|
|
|
// 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 };
|
2013-09-23 22:34:45 +02:00
|
|
|
virtual bool SendPacket(
|
|
|
|
talk_base::Buffer* packet,
|
|
|
|
talk_base::DiffServCodePoint dscp = talk_base::DSCP_NO_CHANGE) = 0;
|
|
|
|
virtual bool SendRtcp(
|
|
|
|
talk_base::Buffer* packet,
|
|
|
|
talk_base::DiffServCodePoint dscp = talk_base::DSCP_NO_CHANGE) = 0;
|
2013-07-10 02:45:36 +02:00
|
|
|
virtual int SetOption(SocketType type, talk_base::Socket::Option opt,
|
|
|
|
int option) = 0;
|
|
|
|
virtual ~NetworkInterface() {}
|
|
|
|
};
|
|
|
|
|
|
|
|
MediaChannel() : network_interface_(NULL) {}
|
|
|
|
virtual ~MediaChannel() {}
|
|
|
|
|
2013-07-26 21:17:59 +02:00
|
|
|
// Sets the abstract interface class for sending RTP/RTCP data.
|
2013-07-10 02:45:36 +02:00
|
|
|
virtual void SetInterface(NetworkInterface *iface) {
|
2013-07-26 21:17:59 +02:00
|
|
|
talk_base::CritScope cs(&network_interface_crit_);
|
2013-07-10 02:45:36 +02:00
|
|
|
network_interface_ = iface;
|
|
|
|
}
|
|
|
|
|
|
|
|
// Called when a RTP packet is received.
|
2013-12-13 01:21:03 +01:00
|
|
|
virtual void OnPacketReceived(talk_base::Buffer* packet,
|
|
|
|
const talk_base::PacketTime& packet_time) = 0;
|
2013-07-10 02:45:36 +02:00
|
|
|
// Called when a RTCP packet is received.
|
2013-12-13 01:21:03 +01:00
|
|
|
virtual void OnRtcpReceived(talk_base::Buffer* packet,
|
|
|
|
const talk_base::PacketTime& packet_time) = 0;
|
2013-07-10 02:45:36 +02:00
|
|
|
// 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;
|
2014-02-17 19:49:41 +01:00
|
|
|
// Returns the absoulte sendtime extension id value from media channel.
|
|
|
|
virtual int GetRtpSendTimeExtnId() const {
|
|
|
|
return -1;
|
|
|
|
}
|
2014-01-16 00:15:54 +01:00
|
|
|
// 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;
|
2013-07-10 02:45:36 +02:00
|
|
|
|
2013-07-26 21:17:59 +02:00
|
|
|
// Base method to send packet using NetworkInterface.
|
|
|
|
bool SendPacket(talk_base::Buffer* packet) {
|
|
|
|
return DoSendPacket(packet, false);
|
|
|
|
}
|
|
|
|
|
|
|
|
bool SendRtcp(talk_base::Buffer* packet) {
|
|
|
|
return DoSendPacket(packet, true);
|
|
|
|
}
|
|
|
|
|
|
|
|
int SetOption(NetworkInterface::SocketType type,
|
|
|
|
talk_base::Socket::Option opt,
|
|
|
|
int option) {
|
|
|
|
talk_base::CritScope cs(&network_interface_crit_);
|
|
|
|
if (!network_interface_)
|
|
|
|
return -1;
|
|
|
|
|
|
|
|
return network_interface_->SetOption(type, opt, option);
|
|
|
|
}
|
|
|
|
|
2013-10-31 16:40:38 +01:00
|
|
|
protected:
|
|
|
|
// This method sets DSCP |value| on both RTP and RTCP channels.
|
|
|
|
int SetDscp(talk_base::DiffServCodePoint value) {
|
|
|
|
int ret;
|
|
|
|
ret = SetOption(NetworkInterface::ST_RTP,
|
|
|
|
talk_base::Socket::OPT_DSCP,
|
|
|
|
value);
|
|
|
|
if (ret == 0) {
|
|
|
|
ret = SetOption(NetworkInterface::ST_RTCP,
|
|
|
|
talk_base::Socket::OPT_DSCP,
|
|
|
|
value);
|
|
|
|
}
|
|
|
|
return ret;
|
|
|
|
}
|
|
|
|
|
2013-07-26 21:17:59 +02:00
|
|
|
private:
|
|
|
|
bool DoSendPacket(talk_base::Buffer* packet, bool rtcp) {
|
|
|
|
talk_base::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.
|
|
|
|
talk_base::CriticalSection network_interface_crit_;
|
|
|
|
NetworkInterface* network_interface_;
|
2013-07-10 02:45:36 +02:00
|
|
|
};
|
|
|
|
|
|
|
|
enum SendFlags {
|
|
|
|
SEND_NOTHING,
|
|
|
|
SEND_RINGBACKTONE,
|
|
|
|
SEND_MICROPHONE
|
|
|
|
};
|
|
|
|
|
2013-10-25 23:18:33 +02:00
|
|
|
// 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()
|
2013-07-10 02:45:36 +02:00
|
|
|
: ssrc(0),
|
2013-10-25 23:18:33 +02:00
|
|
|
timestamp(0) {
|
|
|
|
}
|
|
|
|
uint32 ssrc;
|
|
|
|
double timestamp;
|
|
|
|
};
|
|
|
|
|
|
|
|
struct MediaSenderInfo {
|
|
|
|
MediaSenderInfo()
|
|
|
|
: bytes_sent(0),
|
2013-07-10 02:45:36 +02:00
|
|
|
packets_sent(0),
|
|
|
|
packets_lost(0),
|
|
|
|
fraction_lost(0.0),
|
2013-10-25 23:18:33 +02:00
|
|
|
rtt_ms(0) {
|
|
|
|
}
|
2013-12-05 01:24:06 +01:00
|
|
|
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;
|
|
|
|
}
|
|
|
|
}
|
2013-10-25 23:18:33 +02:00
|
|
|
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;
|
|
|
|
};
|
|
|
|
|
2014-02-27 18:52:04 +01:00
|
|
|
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;
|
|
|
|
};
|
|
|
|
|
2013-10-25 23:18:33 +02:00
|
|
|
struct MediaReceiverInfo {
|
|
|
|
MediaReceiverInfo()
|
|
|
|
: bytes_rcvd(0),
|
|
|
|
packets_rcvd(0),
|
|
|
|
packets_lost(0),
|
|
|
|
fraction_lost(0.0) {
|
|
|
|
}
|
2013-12-05 01:24:06 +01:00
|
|
|
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;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2013-10-25 23:18:33 +02:00
|
|
|
int64 bytes_rcvd;
|
|
|
|
int packets_rcvd;
|
|
|
|
int packets_lost;
|
|
|
|
float fraction_lost;
|
|
|
|
std::vector<SsrcReceiverInfo> local_stats;
|
|
|
|
std::vector<SsrcSenderInfo> remote_stats;
|
|
|
|
};
|
|
|
|
|
|
|
|
struct VoiceSenderInfo : public MediaSenderInfo {
|
|
|
|
VoiceSenderInfo()
|
2013-12-05 01:24:06 +01:00
|
|
|
: ext_seqnum(0),
|
2013-07-10 02:45:36 +02:00
|
|
|
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),
|
2013-09-19 07:49:50 +02:00
|
|
|
echo_return_loss_enhancement(0),
|
|
|
|
typing_noise_detected(false) {
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
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;
|
2013-09-19 07:49:50 +02:00
|
|
|
bool typing_noise_detected;
|
2013-07-10 02:45:36 +02:00
|
|
|
};
|
|
|
|
|
2013-10-25 23:18:33 +02:00
|
|
|
struct VoiceReceiverInfo : public MediaReceiverInfo {
|
2013-07-10 02:45:36 +02:00
|
|
|
VoiceReceiverInfo()
|
2013-12-05 01:24:06 +01:00
|
|
|
: ext_seqnum(0),
|
2013-07-10 02:45:36 +02:00
|
|
|
jitter_ms(0),
|
|
|
|
jitter_buffer_ms(0),
|
|
|
|
jitter_buffer_preferred_ms(0),
|
|
|
|
delay_estimate_ms(0),
|
|
|
|
audio_level(0),
|
2014-02-15 00:38:45 +01:00
|
|
|
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) {
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
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;
|
2014-02-15 00:38:45 +01:00
|
|
|
int decoding_calls_to_silence_generator;
|
|
|
|
int decoding_calls_to_neteq;
|
|
|
|
int decoding_normal;
|
|
|
|
int decoding_plc;
|
|
|
|
int decoding_cng;
|
|
|
|
int decoding_plc_cng;
|
2013-07-10 02:45:36 +02:00
|
|
|
};
|
|
|
|
|
2013-10-25 23:18:33 +02:00
|
|
|
struct VideoSenderInfo : public MediaSenderInfo {
|
2013-07-10 02:45:36 +02:00
|
|
|
VideoSenderInfo()
|
2013-10-25 23:18:33 +02:00
|
|
|
: packets_cached(0),
|
2013-07-10 02:45:36 +02:00
|
|
|
firs_rcvd(0),
|
2014-02-27 18:52:04 +01:00
|
|
|
plis_rcvd(0),
|
2013-07-10 02:45:36 +02:00
|
|
|
nacks_rcvd(0),
|
2014-03-28 17:22:19 +01:00
|
|
|
input_frame_width(0),
|
|
|
|
input_frame_height(0),
|
|
|
|
send_frame_width(0),
|
|
|
|
send_frame_height(0),
|
2013-07-10 02:45:36 +02:00
|
|
|
framerate_input(0),
|
|
|
|
framerate_sent(0),
|
|
|
|
nominal_bitrate(0),
|
|
|
|
preferred_bitrate(0),
|
2013-12-05 01:24:06 +01:00
|
|
|
adapt_reason(0),
|
|
|
|
capture_jitter_ms(0),
|
2013-12-11 19:25:07 +01:00
|
|
|
avg_encode_ms(0),
|
|
|
|
encode_usage_percent(0),
|
|
|
|
capture_queue_delay_ms_per_s(0) {
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
std::vector<SsrcGroup> ssrc_groups;
|
|
|
|
int packets_cached;
|
|
|
|
int firs_rcvd;
|
2014-02-27 18:52:04 +01:00
|
|
|
int plis_rcvd;
|
2013-07-10 02:45:36 +02:00
|
|
|
int nacks_rcvd;
|
2014-03-28 17:22:19 +01:00
|
|
|
int input_frame_width;
|
|
|
|
int input_frame_height;
|
|
|
|
int send_frame_width;
|
|
|
|
int send_frame_height;
|
2013-07-10 02:45:36 +02:00
|
|
|
int framerate_input;
|
|
|
|
int framerate_sent;
|
|
|
|
int nominal_bitrate;
|
|
|
|
int preferred_bitrate;
|
|
|
|
int adapt_reason;
|
2013-12-05 01:24:06 +01:00
|
|
|
int capture_jitter_ms;
|
|
|
|
int avg_encode_ms;
|
2013-12-11 19:25:07 +01:00
|
|
|
int encode_usage_percent;
|
|
|
|
int capture_queue_delay_ms_per_s;
|
2014-02-27 18:52:04 +01:00
|
|
|
VariableInfo<int> adapt_frame_drops;
|
|
|
|
VariableInfo<int> effects_frame_drops;
|
|
|
|
VariableInfo<double> capturer_frame_time;
|
2013-07-10 02:45:36 +02:00
|
|
|
};
|
|
|
|
|
2013-10-25 23:18:33 +02:00
|
|
|
struct VideoReceiverInfo : public MediaReceiverInfo {
|
2013-07-10 02:45:36 +02:00
|
|
|
VideoReceiverInfo()
|
2013-10-25 23:18:33 +02:00
|
|
|
: packets_concealed(0),
|
2013-07-10 02:45:36 +02:00
|
|
|
firs_sent(0),
|
2014-02-27 18:52:04 +01:00
|
|
|
plis_sent(0),
|
2013-07-10 02:45:36 +02:00
|
|
|
nacks_sent(0),
|
|
|
|
frame_width(0),
|
|
|
|
frame_height(0),
|
|
|
|
framerate_rcvd(0),
|
|
|
|
framerate_decoded(0),
|
|
|
|
framerate_output(0),
|
|
|
|
framerate_render_input(0),
|
2013-10-25 23:18:33 +02:00
|
|
|
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),
|
2014-05-06 23:36:31 +02:00
|
|
|
current_delay_ms(0),
|
|
|
|
capture_start_ntp_time_ms(0) {
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
std::vector<SsrcGroup> ssrc_groups;
|
|
|
|
int packets_concealed;
|
|
|
|
int firs_sent;
|
2014-02-27 18:52:04 +01:00
|
|
|
int plis_sent;
|
2013-07-10 02:45:36 +02:00
|
|
|
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;
|
2013-10-25 23:18:33 +02:00
|
|
|
|
|
|
|
// 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;
|
2014-05-06 23:36:31 +02:00
|
|
|
|
|
|
|
// Estimated capture start time in NTP time in ms.
|
|
|
|
int64 capture_start_ntp_time_ms;
|
2013-07-10 02:45:36 +02:00
|
|
|
};
|
|
|
|
|
2013-10-25 23:18:33 +02:00
|
|
|
struct DataSenderInfo : public MediaSenderInfo {
|
2013-07-10 02:45:36 +02:00
|
|
|
DataSenderInfo()
|
2013-10-25 23:18:33 +02:00
|
|
|
: ssrc(0) {
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
uint32 ssrc;
|
|
|
|
};
|
|
|
|
|
2013-10-25 23:18:33 +02:00
|
|
|
struct DataReceiverInfo : public MediaReceiverInfo {
|
2013-07-10 02:45:36 +02:00
|
|
|
DataReceiverInfo()
|
2013-10-25 23:18:33 +02:00
|
|
|
: ssrc(0) {
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
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),
|
2014-02-14 00:18:49 +01:00
|
|
|
bucket_delay(0),
|
|
|
|
total_received_propagation_delta_ms(0) {
|
2013-07-10 02:45:36 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
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;
|
2014-02-14 00:18:49 +01:00
|
|
|
// 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;
|
2013-07-10 02:45:36 +02:00
|
|
|
};
|
|
|
|
|
|
|
|
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;
|
|
|
|
};
|
|
|
|
|
2014-02-14 00:18:49 +01:00
|
|
|
struct StatsOptions {
|
|
|
|
StatsOptions() : include_received_propagation_stats(false) {}
|
|
|
|
|
|
|
|
bool include_received_propagation_stats;
|
|
|
|
};
|
|
|
|
|
2013-07-10 02:45:36 +02:00
|
|
|
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;
|
2013-07-26 21:17:59 +02:00
|
|
|
// 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;
|
2013-07-10 02:45:36 +02:00
|
|
|
// 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.
|
2013-07-17 16:42:53 +02:00
|
|
|
// The valid value for the |event| are 0 to 15 which corresponding to
|
|
|
|
// DTMF event 0-9, *, #, A-D.
|
2013-07-10 02:45:36 +02:00
|
|
|
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.
|
2014-02-14 00:18:49 +01:00
|
|
|
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);
|
|
|
|
}
|
2013-07-10 02:45:36 +02:00
|
|
|
|
|
|
|
// 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 {
|
2013-09-23 22:34:45 +02:00
|
|
|
// Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID
|
|
|
|
// values.
|
|
|
|
DMT_NONE = 0,
|
|
|
|
DMT_CONTROL = 1,
|
|
|
|
DMT_BINARY = 2,
|
|
|
|
DMT_TEXT = 3,
|
2013-07-10 02:45:36 +02:00
|
|
|
};
|
|
|
|
|
|
|
|
// 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) {
|
|
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}
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};
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enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
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class DataMediaChannel : public MediaChannel {
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public:
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enum Error {
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ERROR_NONE = 0, // No error.
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ERROR_OTHER, // Other errors.
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ERROR_SEND_SRTP_ERROR = 200, // Generic SRTP failure.
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ERROR_SEND_SRTP_AUTH_FAILED, // Failed to authenticate packets.
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ERROR_RECV_SRTP_ERROR, // Generic SRTP failure.
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ERROR_RECV_SRTP_AUTH_FAILED, // Failed to authenticate packets.
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ERROR_RECV_SRTP_REPLAY, // Packet replay detected.
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};
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virtual ~DataMediaChannel() {}
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virtual bool SetSendCodecs(const std::vector<DataCodec>& codecs) = 0;
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virtual bool SetRecvCodecs(const std::vector<DataCodec>& codecs) = 0;
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2013-12-13 01:21:03 +01:00
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2013-07-10 02:45:36 +02:00
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virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
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// TODO(pthatcher): Implement this.
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virtual bool GetStats(DataMediaInfo* info) { return true; }
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virtual bool SetSend(bool send) = 0;
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virtual bool SetReceive(bool receive) = 0;
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virtual bool SendData(
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const SendDataParams& params,
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const talk_base::Buffer& payload,
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SendDataResult* result = NULL) = 0;
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// Signals when data is received (params, data, len)
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sigslot::signal3<const ReceiveDataParams&,
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const char*,
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size_t> SignalDataReceived;
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// Signal errors from MediaChannel. Arguments are:
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// ssrc(uint32), and error(DataMediaChannel::Error).
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sigslot::signal2<uint32, DataMediaChannel::Error> SignalMediaError;
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2013-08-01 02:00:07 +02:00
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// Signal when the media channel is ready to send the stream. Arguments are:
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// writable(bool)
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sigslot::signal1<bool> SignalReadyToSend;
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2013-07-10 02:45:36 +02:00
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};
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} // namespace cricket
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#endif // TALK_MEDIA_BASE_MEDIACHANNEL_H_
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