blob: 99ed29169b04ae8d446b3df4aba858ece1e2db40 [file] [log] [blame]
/*
* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
#define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/rtpparameters.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/api/video/video_timing.h"
#include "webrtc/config.h"
#include "webrtc/media/base/codec.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/streamparams.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
#include "webrtc/rtc_base/basictypes.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/copyonwritebuffer.h"
#include "webrtc/rtc_base/dscp.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/networkroute.h"
#include "webrtc/rtc_base/optional.h"
#include "webrtc/rtc_base/sigslot.h"
#include "webrtc/rtc_base/socket.h"
#include "webrtc/rtc_base/window.h"
// TODO(juberti): re-evaluate this include
#include "webrtc/pc/audiomonitor.h"
namespace rtc {
class RateLimiter;
class Timing;
}
namespace webrtc {
class AudioSinkInterface;
class VideoFrame;
}
namespace cricket {
class AudioSource;
class VideoCapturer;
struct RtpHeader;
struct VideoFormat;
const int kScreencastDefaultFps = 5;
template <class T>
static std::string ToStringIfSet(const char* key, const rtc::Optional<T>& val) {
std::string str;
if (val) {
str = key;
str += ": ";
str += val ? rtc::ToString(*val) : "";
str += ", ";
}
return str;
}
template <class T>
static std::string VectorToString(const std::vector<T>& vals) {
std::ostringstream ost;
ost << "[";
for (size_t i = 0; i < vals.size(); ++i) {
if (i > 0) {
ost << ", ";
}
ost << vals[i].ToString();
}
ost << "]";
return ost.str();
}
// Construction-time settings, passed on when creating
// MediaChannels.
struct MediaConfig {
// Set DSCP value on packets. This flag comes from the
// PeerConnection constraint 'googDscp'.
bool enable_dscp = false;
// Video-specific config.
struct Video {
// Enable WebRTC CPU Overuse Detection. This flag comes from the
// PeerConnection constraint 'googCpuOveruseDetection'.
bool enable_cpu_overuse_detection = true;
// Enable WebRTC suspension of video. No video frames will be sent
// when the bitrate is below the configured minimum bitrate. This
// flag comes from the PeerConnection constraint
// 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
// to VideoSendStream::Config::suspend_below_min_bitrate.
bool suspend_below_min_bitrate = false;
// Set to true if the renderer has an algorithm of frame selection.
// If the value is true, then WebRTC will hand over a frame as soon as
// possible without delay, and rendering smoothness is completely the duty
// of the renderer;
// If the value is false, then WebRTC is responsible to delay frame release
// in order to increase rendering smoothness.
//
// This flag comes from PeerConnection's RtcConfiguration, but is
// currently only set by the command line flag
// 'disable-rtc-smoothness-algorithm'.
// WebRtcVideoChannel::AddRecvStream copies it to the created
// WebRtcVideoReceiveStream, where it is returned by the
// SmoothsRenderedFrames method. This method is used by the
// VideoReceiveStream, where the value is passed on to the
// IncomingVideoStream constructor.
bool disable_prerenderer_smoothing = false;
// Enables periodic bandwidth probing in application-limited region.
bool periodic_alr_bandwidth_probing = false;
} video;
bool operator==(const MediaConfig& o) const {
return enable_dscp == o.enable_dscp &&
video.enable_cpu_overuse_detection ==
o.video.enable_cpu_overuse_detection &&
video.suspend_below_min_bitrate ==
o.video.suspend_below_min_bitrate &&
video.disable_prerenderer_smoothing ==
o.video.disable_prerenderer_smoothing &&
video.periodic_alr_bandwidth_probing ==
o.video.periodic_alr_bandwidth_probing;
}
bool operator!=(const MediaConfig& o) const { return !(*this == o); }
};
// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
// Used to be flags, but that makes it hard to selectively apply options.
// We are moving all of the setting of options to structs like this,
// but some things currently still use flags.
struct AudioOptions {
void SetAll(const AudioOptions& change) {
SetFrom(&echo_cancellation, change.echo_cancellation);
SetFrom(&auto_gain_control, change.auto_gain_control);
SetFrom(&noise_suppression, change.noise_suppression);
SetFrom(&highpass_filter, change.highpass_filter);
SetFrom(&stereo_swapping, change.stereo_swapping);
SetFrom(&audio_jitter_buffer_max_packets,
change.audio_jitter_buffer_max_packets);
SetFrom(&audio_jitter_buffer_fast_accelerate,
change.audio_jitter_buffer_fast_accelerate);
SetFrom(&typing_detection, change.typing_detection);
SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
SetFrom(&adjust_agc_delta, change.adjust_agc_delta);
SetFrom(&experimental_agc, change.experimental_agc);
SetFrom(&extended_filter_aec, change.extended_filter_aec);
SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
SetFrom(&experimental_ns, change.experimental_ns);
SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
SetFrom(&level_control, change.level_control);
SetFrom(&residual_echo_detector, change.residual_echo_detector);
SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
SetFrom(&tx_agc_digital_compression_gain,
change.tx_agc_digital_compression_gain);
SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
SetFrom(&recording_sample_rate, change.recording_sample_rate);
SetFrom(&playout_sample_rate, change.playout_sample_rate);
SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
SetFrom(&level_control_initial_peak_level_dbfs,
change.level_control_initial_peak_level_dbfs);
}
bool operator==(const AudioOptions& o) const {
return echo_cancellation == o.echo_cancellation &&
auto_gain_control == o.auto_gain_control &&
noise_suppression == o.noise_suppression &&
highpass_filter == o.highpass_filter &&
stereo_swapping == o.stereo_swapping &&
audio_jitter_buffer_max_packets ==
o.audio_jitter_buffer_max_packets &&
audio_jitter_buffer_fast_accelerate ==
o.audio_jitter_buffer_fast_accelerate &&
typing_detection == o.typing_detection &&
aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
experimental_agc == o.experimental_agc &&
extended_filter_aec == o.extended_filter_aec &&
delay_agnostic_aec == o.delay_agnostic_aec &&
experimental_ns == o.experimental_ns &&
intelligibility_enhancer == o.intelligibility_enhancer &&
level_control == o.level_control &&
residual_echo_detector == o.residual_echo_detector &&
adjust_agc_delta == o.adjust_agc_delta &&
tx_agc_target_dbov == o.tx_agc_target_dbov &&
tx_agc_digital_compression_gain ==
o.tx_agc_digital_compression_gain &&
tx_agc_limiter == o.tx_agc_limiter &&
recording_sample_rate == o.recording_sample_rate &&
playout_sample_rate == o.playout_sample_rate &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
audio_network_adaptor == o.audio_network_adaptor &&
audio_network_adaptor_config == o.audio_network_adaptor_config &&
level_control_initial_peak_level_dbfs ==
o.level_control_initial_peak_level_dbfs;
}
bool operator!=(const AudioOptions& o) const { return !(*this == o); }
std::string ToString() const {
std::ostringstream ost;
ost << "AudioOptions {";
ost << ToStringIfSet("aec", echo_cancellation);
ost << ToStringIfSet("agc", auto_gain_control);
ost << ToStringIfSet("ns", noise_suppression);
ost << ToStringIfSet("hf", highpass_filter);
ost << ToStringIfSet("swap", stereo_swapping);
ost << ToStringIfSet("audio_jitter_buffer_max_packets",
audio_jitter_buffer_max_packets);
ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
audio_jitter_buffer_fast_accelerate);
ost << ToStringIfSet("typing", typing_detection);
ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
ost << ToStringIfSet("agc_delta", adjust_agc_delta);
ost << ToStringIfSet("experimental_agc", experimental_agc);
ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
ost << ToStringIfSet("experimental_ns", experimental_ns);
ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
ost << ToStringIfSet("level_control", level_control);
ost << ToStringIfSet("level_control_initial_peak_level_dbfs",
level_control_initial_peak_level_dbfs);
ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
ost << ToStringIfSet("tx_agc_digital_compression_gain",
tx_agc_digital_compression_gain);
ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
// The adaptor config is a serialized proto buffer and therefore not human
// readable. So we comment out the following line.
// ost << ToStringIfSet("audio_network_adaptor_config",
// audio_network_adaptor_config);
ost << "}";
return ost.str();
}
// Audio processing that attempts to filter away the output signal from
// later inbound pickup.
rtc::Optional<bool> echo_cancellation;
// Audio processing to adjust the sensitivity of the local mic dynamically.
rtc::Optional<bool> auto_gain_control;
// Audio processing to filter out background noise.
rtc::Optional<bool> noise_suppression;
// Audio processing to remove background noise of lower frequencies.
rtc::Optional<bool> highpass_filter;
// Audio processing to swap the left and right channels.
rtc::Optional<bool> stereo_swapping;
// Audio receiver jitter buffer (NetEq) max capacity in number of packets.
rtc::Optional<int> audio_jitter_buffer_max_packets;
// Audio receiver jitter buffer (NetEq) fast accelerate mode.
rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
// Audio processing to detect typing.
rtc::Optional<bool> typing_detection;
rtc::Optional<bool> aecm_generate_comfort_noise;
rtc::Optional<int> adjust_agc_delta;
rtc::Optional<bool> experimental_agc;
rtc::Optional<bool> extended_filter_aec;
rtc::Optional<bool> delay_agnostic_aec;
rtc::Optional<bool> experimental_ns;
rtc::Optional<bool> intelligibility_enhancer;
rtc::Optional<bool> level_control;
// Specifies an optional initialization value for the level controller.
rtc::Optional<float> level_control_initial_peak_level_dbfs;
// Note that tx_agc_* only applies to non-experimental AGC.
rtc::Optional<bool> residual_echo_detector;
rtc::Optional<uint16_t> tx_agc_target_dbov;
rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
rtc::Optional<bool> tx_agc_limiter;
rtc::Optional<uint32_t> recording_sample_rate;
rtc::Optional<uint32_t> playout_sample_rate;
// Enable combined audio+bandwidth BWE.
// TODO(pthatcher): This flag is set from the
// "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
// and check if any other AudioOptions members are unused.
rtc::Optional<bool> combined_audio_video_bwe;
// Enable audio network adaptor.
rtc::Optional<bool> audio_network_adaptor;
// Config string for audio network adaptor.
rtc::Optional<std::string> audio_network_adaptor_config;
private:
template <typename T>
static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
if (o) {
*s = o;
}
}
};
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
// Used to be flags, but that makes it hard to selectively apply options.
// We are moving all of the setting of options to structs like this,
// but some things currently still use flags.
struct VideoOptions {
void SetAll(const VideoOptions& change) {
SetFrom(&video_noise_reduction, change.video_noise_reduction);
SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps);
SetFrom(&is_screencast, change.is_screencast);
}
bool operator==(const VideoOptions& o) const {
return video_noise_reduction == o.video_noise_reduction &&
screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps &&
is_screencast == o.is_screencast;
}
bool operator!=(const VideoOptions& o) const { return !(*this == o); }
std::string ToString() const {
std::ostringstream ost;
ost << "VideoOptions {";
ost << ToStringIfSet("noise reduction", video_noise_reduction);
ost << ToStringIfSet("screencast min bitrate kbps",
screencast_min_bitrate_kbps);
ost << ToStringIfSet("is_screencast ", is_screencast);
ost << "}";
return ost.str();
}
// Enable denoising? This flag comes from the getUserMedia
// constraint 'googNoiseReduction', and WebRtcVideoEngine passes it
// on to the codec options. Disabled by default.
rtc::Optional<bool> video_noise_reduction;
// Force screencast to use a minimum bitrate. This flag comes from
// the PeerConnection constraint 'googScreencastMinBitrate'. It is
// copied to the encoder config by WebRtcVideoChannel.
rtc::Optional<int> screencast_min_bitrate_kbps;
// Set by screencast sources. Implies selection of encoding settings
// suitable for screencast. Most likely not the right way to do
// things, e.g., screencast of a text document and screencast of a
// youtube video have different needs.
rtc::Optional<bool> is_screencast;
private:
template <typename T>
static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
if (o) {
*s = o;
}
}
};
// TODO(isheriff): Remove this once client usage is fixed to use RtpExtension.
struct RtpHeaderExtension {
RtpHeaderExtension() : id(0) {}
RtpHeaderExtension(const std::string& uri, int id) : uri(uri), id(id) {}
std::string ToString() const {
std::ostringstream ost;
ost << "{";
ost << "uri: " << uri;
ost << ", id: " << id;
ost << "}";
return ost.str();
}
std::string uri;
int id;
};
class MediaChannel : public sigslot::has_slots<> {
public:
class NetworkInterface {
public:
enum SocketType { ST_RTP, ST_RTCP };
virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) = 0;
virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) = 0;
virtual int SetOption(SocketType type, rtc::Socket::Option opt,
int option) = 0;
virtual ~NetworkInterface() {}
};
explicit MediaChannel(const MediaConfig& config)
: enable_dscp_(config.enable_dscp), network_interface_(NULL) {}
MediaChannel() : enable_dscp_(false), 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;
SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
}
virtual rtc::DiffServCodePoint PreferredDscp() const {
return rtc::DSCP_DEFAULT;
}
// Called when a RTP packet is received.
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) = 0;
// Called when a RTCP packet is received.
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) = 0;
// Called when the socket's ability to send has changed.
virtual void OnReadyToSend(bool ready) = 0;
// Called when the network route used for sending packets changed.
virtual void OnNetworkRouteChanged(
const std::string& transport_name,
const rtc::NetworkRoute& network_route) = 0;
// Called when the rtp transport overhead changed.
virtual void OnTransportOverheadChanged(
int transport_overhead_per_packet) = 0;
// Creates a new outgoing media stream with SSRCs and CNAME as described
// by sp.
virtual bool AddSendStream(const StreamParams& sp) = 0;
// Removes an outgoing media stream.
// ssrc must be the first SSRC of the media stream if the stream uses
// multiple SSRCs.
virtual bool RemoveSendStream(uint32_t ssrc) = 0;
// Creates a new incoming media stream with SSRCs and CNAME as described
// by sp.
virtual bool AddRecvStream(const StreamParams& sp) = 0;
// Removes an incoming media stream.
// ssrc must be the first SSRC of the media stream if the stream uses
// multiple SSRCs.
virtual bool RemoveRecvStream(uint32_t ssrc) = 0;
// Returns the absoulte sendtime extension id value from media channel.
virtual int GetRtpSendTimeExtnId() const {
return -1;
}
// Base method to send packet using NetworkInterface.
bool SendPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
return DoSendPacket(packet, false, options);
}
bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
return DoSendPacket(packet, true, options);
}
int SetOption(NetworkInterface::SocketType type,
rtc::Socket::Option opt,
int option) {
rtc::CritScope cs(&network_interface_crit_);
if (!network_interface_)
return -1;
return network_interface_->SetOption(type, opt, option);
}
private:
// 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;
}
bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
bool rtcp,
const rtc::PacketOptions& options) {
rtc::CritScope cs(&network_interface_crit_);
if (!network_interface_)
return false;
return (!rtcp) ? network_interface_->SendPacket(packet, options)
: network_interface_->SendRtcp(packet, options);
}
const bool enable_dscp_;
// |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_;
};
// The stats information is structured as follows:
// Media are represented by either MediaSenderInfo or MediaReceiverInfo.
// Media contains a vector of SSRC infos that are exclusively used by this
// media. (SSRCs shared between media streams can't be represented.)
// Information about an SSRC.
// This data may be locally recorded, or received in an RTCP SR or RR.
struct SsrcSenderInfo {
SsrcSenderInfo()
: ssrc(0),
timestamp(0) {
}
uint32_t ssrc;
double timestamp; // NTP timestamp, represented as seconds since epoch.
};
struct SsrcReceiverInfo {
SsrcReceiverInfo()
: ssrc(0),
timestamp(0) {
}
uint32_t ssrc;
double timestamp;
};
struct MediaSenderInfo {
MediaSenderInfo()
: bytes_sent(0),
packets_sent(0),
packets_lost(0),
fraction_lost(0.0),
rtt_ms(0) {
}
void add_ssrc(const SsrcSenderInfo& stat) {
local_stats.push_back(stat);
}
// Temporary utility function for call sites that only provide SSRC.
// As more info is added into SsrcSenderInfo, this function should go away.
void add_ssrc(uint32_t ssrc) {
SsrcSenderInfo stat;
stat.ssrc = ssrc;
add_ssrc(stat);
}
// Utility accessor for clients that are only interested in ssrc numbers.
std::vector<uint32_t> ssrcs() const {
std::vector<uint32_t> retval;
for (std::vector<SsrcSenderInfo>::const_iterator it = local_stats.begin();
it != local_stats.end(); ++it) {
retval.push_back(it->ssrc);
}
return retval;
}
// Utility accessor for clients that make the assumption only one ssrc
// exists per media.
// This will eventually go away.
uint32_t ssrc() const {
if (local_stats.size() > 0) {
return local_stats[0].ssrc;
} else {
return 0;
}
}
int64_t bytes_sent;
int packets_sent;
int packets_lost;
float fraction_lost;
int64_t rtt_ms;
std::string codec_name;
rtc::Optional<int> codec_payload_type;
std::vector<SsrcSenderInfo> local_stats;
std::vector<SsrcReceiverInfo> remote_stats;
};
struct MediaReceiverInfo {
MediaReceiverInfo()
: bytes_rcvd(0),
packets_rcvd(0),
packets_lost(0),
fraction_lost(0.0) {
}
void add_ssrc(const SsrcReceiverInfo& stat) {
local_stats.push_back(stat);
}
// Temporary utility function for call sites that only provide SSRC.
// As more info is added into SsrcSenderInfo, this function should go away.
void add_ssrc(uint32_t ssrc) {
SsrcReceiverInfo stat;
stat.ssrc = ssrc;
add_ssrc(stat);
}
std::vector<uint32_t> ssrcs() const {
std::vector<uint32_t> retval;
for (std::vector<SsrcReceiverInfo>::const_iterator it = local_stats.begin();
it != local_stats.end(); ++it) {
retval.push_back(it->ssrc);
}
return retval;
}
// Utility accessor for clients that make the assumption only one ssrc
// exists per media.
// This will eventually go away.
uint32_t ssrc() const {
if (local_stats.size() > 0) {
return local_stats[0].ssrc;
} else {
return 0;
}
}
int64_t bytes_rcvd;
int packets_rcvd;
int packets_lost;
float fraction_lost;
std::string codec_name;
rtc::Optional<int> codec_payload_type;
std::vector<SsrcReceiverInfo> local_stats;
std::vector<SsrcSenderInfo> remote_stats;
};
struct VoiceSenderInfo : public MediaSenderInfo {
VoiceSenderInfo()
: ext_seqnum(0),
jitter_ms(0),
audio_level(0),
total_input_energy(0.0),
total_input_duration(0.0),
aec_quality_min(0.0),
echo_delay_median_ms(0),
echo_delay_std_ms(0),
echo_return_loss(0),
echo_return_loss_enhancement(0),
residual_echo_likelihood(0.0f),
residual_echo_likelihood_recent_max(0.0f),
typing_noise_detected(false) {}
int ext_seqnum;
int jitter_ms;
int audio_level;
// See description of "totalAudioEnergy" in the WebRTC stats spec:
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
double total_input_energy;
double total_input_duration;
float aec_quality_min;
int echo_delay_median_ms;
int echo_delay_std_ms;
int echo_return_loss;
int echo_return_loss_enhancement;
float residual_echo_likelihood;
float residual_echo_likelihood_recent_max;
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),
total_output_energy(0.0),
total_samples_received(0),
total_output_duration(0.0),
concealed_samples(0),
expand_rate(0),
speech_expand_rate(0),
secondary_decoded_rate(0),
secondary_discarded_rate(0),
accelerate_rate(0),
preemptive_expand_rate(0),
decoding_calls_to_silence_generator(0),
decoding_calls_to_neteq(0),
decoding_normal(0),
decoding_plc(0),
decoding_cng(0),
decoding_plc_cng(0),
decoding_muted_output(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;
// See description of "totalAudioEnergy" in the WebRTC stats spec:
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
double total_output_energy;
// See description of "totalSamplesReceived" in the WebRTC stats spec:
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesreceived
uint64_t total_samples_received;
// See description of "totalSamplesDuration" in the WebRTC stats spec:
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesduration
double total_output_duration;
// See description of "concealedSamples" in the WebRTC stats spec:
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-concealedsamples
uint64_t concealed_samples;
// fraction of synthesized audio inserted through expansion.
float expand_rate;
// fraction of synthesized speech inserted through expansion.
float speech_expand_rate;
// fraction of data out of secondary decoding, including FEC and RED.
float secondary_decoded_rate;
// Fraction of secondary data, including FEC and RED, that is discarded.
// Discarding of secondary data can be caused by the reception of the primary
// data, obsoleting the secondary data. It can also be caused by early
// or late arrival of secondary data. This metric is the percentage of
// discarded secondary data since last query of receiver info.
float secondary_discarded_rate;
// Fraction of data removed through time compression.
float accelerate_rate;
// Fraction of data inserted through time stretching.
float preemptive_expand_rate;
int decoding_calls_to_silence_generator;
int decoding_calls_to_neteq;
int decoding_normal;
int decoding_plc;
int decoding_cng;
int decoding_plc_cng;
int decoding_muted_output;
// Estimated capture start time in NTP time in ms.
int64_t capture_start_ntp_time_ms;
};
struct VideoSenderInfo : public MediaSenderInfo {
VideoSenderInfo()
: packets_cached(0),
firs_rcvd(0),
plis_rcvd(0),
nacks_rcvd(0),
send_frame_width(0),
send_frame_height(0),
framerate_input(0),
framerate_sent(0),
nominal_bitrate(0),
preferred_bitrate(0),
adapt_reason(0),
adapt_changes(0),
avg_encode_ms(0),
encode_usage_percent(0),
frames_encoded(0) {}
std::vector<SsrcGroup> ssrc_groups;
// TODO(hbos): Move this to |VideoMediaInfo::send_codecs|?
std::string encoder_implementation_name;
int packets_cached;
int firs_rcvd;
int plis_rcvd;
int nacks_rcvd;
int send_frame_width;
int send_frame_height;
int framerate_input;
int framerate_sent;
int nominal_bitrate;
int preferred_bitrate;
int adapt_reason;
int adapt_changes;
int avg_encode_ms;
int encode_usage_percent;
uint32_t frames_encoded;
rtc::Optional<uint64_t> qp_sum;
};
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),
frames_received(0),
frames_decoded(0),
frames_rendered(0),
interframe_delay_max_ms(-1),
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;
// TODO(hbos): Move this to |VideoMediaInfo::receive_codecs|?
std::string decoder_implementation_name;
int packets_concealed;
int firs_sent;
int plis_sent;
int nacks_sent;
int frame_width;
int frame_height;
int framerate_rcvd;
int framerate_decoded;
int framerate_output;
// Framerate as sent to the renderer.
int framerate_render_input;
// Framerate that the renderer reports.
int framerate_render_output;
uint32_t frames_received;
uint32_t frames_decoded;
uint32_t frames_rendered;
rtc::Optional<uint64_t> qp_sum;
int64_t interframe_delay_max_ms;
// All stats below are gathered per-VideoReceiver, but some will be correlated
// across MediaStreamTracks. NOTE(hta): when sinking stats into per-SSRC
// structures, reflect this in the new layout.
// Current frame decode latency.
int decode_ms;
// Maximum observed frame decode latency.
int max_decode_ms;
// Jitter (network-related) latency.
int jitter_buffer_ms;
// Requested minimum playout latency.
int min_playout_delay_ms;
// Requested latency to account for rendering delay.
int render_delay_ms;
// Target overall delay: network+decode+render, accounting for
// min_playout_delay_ms.
int target_delay_ms;
// Current overall delay, possibly ramping towards target_delay_ms.
int current_delay_ms;
// Estimated capture start time in NTP time in ms.
int64_t capture_start_ntp_time_ms;
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
rtc::Optional<webrtc::TimingFrameInfo> timing_frame_info;
};
struct DataSenderInfo : public MediaSenderInfo {
DataSenderInfo()
: ssrc(0) {
}
uint32_t ssrc;
};
struct DataReceiverInfo : public MediaReceiverInfo {
DataReceiverInfo()
: ssrc(0) {
}
uint32_t ssrc;
};
struct BandwidthEstimationInfo {
BandwidthEstimationInfo()
: available_send_bandwidth(0),
available_recv_bandwidth(0),
target_enc_bitrate(0),
actual_enc_bitrate(0),
retransmit_bitrate(0),
transmit_bitrate(0),
bucket_delay(0) {
}
int available_send_bandwidth;
int available_recv_bandwidth;
int target_enc_bitrate;
int actual_enc_bitrate;
int retransmit_bitrate;
int transmit_bitrate;
int64_t bucket_delay;
};
// Maps from payload type to |RtpCodecParameters|.
typedef std::map<int, webrtc::RtpCodecParameters> RtpCodecParametersMap;
struct VoiceMediaInfo {
void Clear() {
senders.clear();
receivers.clear();
send_codecs.clear();
receive_codecs.clear();
}
std::vector<VoiceSenderInfo> senders;
std::vector<VoiceReceiverInfo> receivers;
RtpCodecParametersMap send_codecs;
RtpCodecParametersMap receive_codecs;
};
struct VideoMediaInfo {
void Clear() {
senders.clear();
receivers.clear();
bw_estimations.clear();
send_codecs.clear();
receive_codecs.clear();
}
std::vector<VideoSenderInfo> senders;
std::vector<VideoReceiverInfo> receivers;
// Deprecated.
// TODO(holmer): Remove once upstream projects no longer use this.
std::vector<BandwidthEstimationInfo> bw_estimations;
RtpCodecParametersMap send_codecs;
RtpCodecParametersMap receive_codecs;
};
struct DataMediaInfo {
void Clear() {
senders.clear();
receivers.clear();
}
std::vector<DataSenderInfo> senders;
std::vector<DataReceiverInfo> receivers;
};
struct RtcpParameters {
bool reduced_size = false;
};
template <class Codec>
struct RtpParameters {
virtual std::string ToString() const {
std::ostringstream ost;
ost << "{";
ost << "codecs: " << VectorToString(codecs) << ", ";
ost << "extensions: " << VectorToString(extensions);
ost << "}";
return ost.str();
}
std::vector<Codec> codecs;
std::vector<webrtc::RtpExtension> extensions;
// TODO(pthatcher): Add streams.
RtcpParameters rtcp;
virtual ~RtpParameters() = default;
};
// TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
// encapsulate all the parameters needed for an RtpSender.
template <class Codec>
struct RtpSendParameters : RtpParameters<Codec> {
std::string ToString() const override {
std::ostringstream ost;
ost << "{";
ost << "codecs: " << VectorToString(this->codecs) << ", ";
ost << "extensions: " << VectorToString(this->extensions) << ", ";
ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
ost << "}";
return ost.str();
}
int max_bandwidth_bps = -1;
};
struct AudioSendParameters : RtpSendParameters<AudioCodec> {
std::string ToString() const override {
std::ostringstream ost;
ost << "{";
ost << "codecs: " << VectorToString(this->codecs) << ", ";
ost << "extensions: " << VectorToString(this->extensions) << ", ";
ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
ost << "options: " << options.ToString();
ost << "}";
return ost.str();
}
AudioOptions options;
};
struct AudioRecvParameters : RtpParameters<AudioCodec> {
};
class VoiceMediaChannel : public MediaChannel {
public:
enum Error {
ERROR_NONE = 0, // No error.
ERROR_OTHER, // Other errors.
ERROR_REC_DEVICE_OPEN_FAILED = 100, // Could not open mic.
ERROR_REC_DEVICE_MUTED, // Mic was muted by OS.
ERROR_REC_DEVICE_SILENT, // No background noise picked up.
ERROR_REC_DEVICE_SATURATION, // Mic input is clipping.
ERROR_REC_DEVICE_REMOVED, // Mic was removed while active.
ERROR_REC_RUNTIME_ERROR, // Processing is encountering errors.
ERROR_REC_SRTP_ERROR, // Generic SRTP failure.
ERROR_REC_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_REC_TYPING_NOISE_DETECTED, // Typing noise is detected.
ERROR_PLAY_DEVICE_OPEN_FAILED = 200, // Could not open playout.
ERROR_PLAY_DEVICE_MUTED, // Playout muted by OS.
ERROR_PLAY_DEVICE_REMOVED, // Playout removed while active.
ERROR_PLAY_RUNTIME_ERROR, // Errors in voice processing.
ERROR_PLAY_SRTP_ERROR, // Generic SRTP failure.
ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_PLAY_SRTP_REPLAY, // Packet replay detected.
};
VoiceMediaChannel() {}
explicit VoiceMediaChannel(const MediaConfig& config)
: MediaChannel(config) {}
virtual ~VoiceMediaChannel() {}
virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
virtual bool SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) = 0;
// Get the receive parameters for the incoming stream identified by |ssrc|.
// If |ssrc| is 0, retrieve the receive parameters for the default receive
// stream, which is used when SSRCs are not signaled. Note that calling with
// an |ssrc| of 0 will return encoding parameters with an unset |ssrc|
// member.
virtual webrtc::RtpParameters GetRtpReceiveParameters(
uint32_t ssrc) const = 0;
virtual bool SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) = 0;
// Starts or stops playout of received audio.
virtual void SetPlayout(bool playout) = 0;
// Starts or stops sending (and potentially capture) of local audio.
virtual void SetSend(bool send) = 0;
// Configure stream for sending.
virtual bool SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) = 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;
// Set speaker output volume of the specified ssrc.
virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0;
// Returns if the telephone-event has been negotiated.
virtual bool CanInsertDtmf() = 0;
// Send a DTMF |event|. The DTMF out-of-band signal will be used.
// The |ssrc| should be either 0 or a valid send stream ssrc.
// The valid value for the |event| are 0 to 15 which corresponding to
// DTMF event 0-9, *, #, A-D.
virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0;
virtual void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
virtual std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const = 0;
};
// TODO(deadbeef): Rename to VideoSenderParameters, since they're intended to
// encapsulate all the parameters needed for a video RtpSender.
struct VideoSendParameters : RtpSendParameters<VideoCodec> {
// Use conference mode? This flag comes from the remote
// description's SDP line 'a=x-google-flag:conference', copied over
// by VideoChannel::SetRemoteContent_w, and ultimately used by
// conference mode screencast logic in
// WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig.
// The special screencast behaviour is disabled by default.
bool conference_mode = false;
};
// TODO(deadbeef): Rename to VideoReceiverParameters, since they're intended to
// encapsulate all the parameters needed for a video RtpReceiver.
struct VideoRecvParameters : RtpParameters<VideoCodec> {
};
class VideoMediaChannel : public MediaChannel {
public:
enum Error {
ERROR_NONE = 0, // No error.
ERROR_OTHER, // Other errors.
ERROR_REC_DEVICE_OPEN_FAILED = 100, // Could not open camera.
ERROR_REC_DEVICE_NO_DEVICE, // No camera.
ERROR_REC_DEVICE_IN_USE, // Device is in already use.
ERROR_REC_DEVICE_REMOVED, // Device is removed.
ERROR_REC_SRTP_ERROR, // Generic sender SRTP failure.
ERROR_REC_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_REC_CPU_MAX_CANT_DOWNGRADE, // Can't downgrade capture anymore.
ERROR_PLAY_SRTP_ERROR = 200, // Generic receiver SRTP failure.
ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_PLAY_SRTP_REPLAY, // Packet replay detected.
};
VideoMediaChannel() {}
explicit VideoMediaChannel(const MediaConfig& config)
: MediaChannel(config) {}
virtual ~VideoMediaChannel() {}
virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
virtual bool SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) = 0;
// Get the receive parameters for the incoming stream identified by |ssrc|.
// If |ssrc| is 0, retrieve the receive parameters for the default receive
// stream, which is used when SSRCs are not signaled. Note that calling with
// an |ssrc| of 0 will return encoding parameters with an unset |ssrc|
// member.
virtual webrtc::RtpParameters GetRtpReceiveParameters(
uint32_t ssrc) const = 0;
virtual bool SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) = 0;
// Gets the currently set codecs/payload types to be used for outgoing media.
virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
// Starts or stops transmission (and potentially capture) of local video.
virtual bool SetSend(bool send) = 0;
// Configure stream for sending and register a source.
// The |ssrc| must correspond to a registered send stream.
virtual bool SetVideoSend(
uint32_t ssrc,
bool enable,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0;
// Sets the sink object to be used for the specified stream.
// If SSRC is 0, the sink is used for the 'default' stream.
virtual bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0;
// This fills the "bitrate parts" (rtx, video bitrate) of the
// BandwidthEstimationInfo, since that part that isn't possible to get
// through webrtc::Call::GetStats, as they are statistics of the send
// streams.
// TODO(holmer): We should change this so that either BWE graphs doesn't
// need access to bitrates of the streams, or change the (RTC)StatsCollector
// so that it's getting the send stream stats separately by calling
// GetStats(), and merges with BandwidthEstimationInfo by itself.
virtual void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VideoMediaInfo* info) = 0;
};
enum DataMessageType {
// Chrome-Internal use only. See SctpDataMediaChannel for the actual PPID
// values.
DMT_NONE = 0,
DMT_CONTROL = 1,
DMT_BINARY = 2,
DMT_TEXT = 3,
};
// Info about data received in DataMediaChannel. For use in
// DataMediaChannel::SignalDataReceived and in all of the signals that
// signal fires, on up the chain.
struct ReceiveDataParams {
// The in-packet stream indentifier.
// RTP data channels use SSRCs, SCTP data channels use SIDs.
union {
uint32_t ssrc;
int sid;
};
// 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() : sid(0), type(DMT_TEXT), seq_num(0), timestamp(0) {}
};
struct SendDataParams {
// The in-packet stream indentifier.
// RTP data channels use SSRCs, SCTP data channels use SIDs.
union {
uint32_t ssrc;
int sid;
};
// 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()
: sid(0),
type(DMT_TEXT),
// TODO(pthatcher): Make these true by default?
ordered(false),
reliable(false),
max_rtx_count(0),
max_rtx_ms(0) {}
};
enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
struct DataSendParameters : RtpSendParameters<DataCodec> {
std::string ToString() const {
std::ostringstream ost;
// Options and extensions aren't used.
ost << "{";
ost << "codecs: " << VectorToString(codecs) << ", ";
ost << "max_bandwidth_bps: " << max_bandwidth_bps;
ost << "}";
return ost.str();
}
};
struct DataRecvParameters : RtpParameters<DataCodec> {
};
class DataMediaChannel : public MediaChannel {
public:
enum Error {
ERROR_NONE = 0, // No error.
ERROR_OTHER, // Other errors.
ERROR_SEND_SRTP_ERROR = 200, // Generic SRTP failure.
ERROR_SEND_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_RECV_SRTP_ERROR, // Generic SRTP failure.
ERROR_RECV_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_RECV_SRTP_REPLAY, // Packet replay detected.
};
DataMediaChannel() {}
DataMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
virtual ~DataMediaChannel() {}
virtual bool SetSendParameters(const DataSendParameters& params) = 0;
virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
// TODO(pthatcher): Implement this.
virtual bool GetStats(DataMediaInfo* info) { return true; }
virtual bool SetSend(bool send) = 0;
virtual bool SetReceive(bool receive) = 0;
virtual void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) {}
virtual bool SendData(
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result = NULL) = 0;
// Signals when data is received (params, data, len)
sigslot::signal3<const ReceiveDataParams&,
const char*,
size_t> SignalDataReceived;
// Signal when the media channel is ready to send the stream. Arguments are:
// writable(bool)
sigslot::signal1<bool> SignalReadyToSend;
};
} // namespace cricket
#endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_