| /* |
| * 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/call/video_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), |
| content_type(webrtc::VideoContentType::UNSPECIFIED), |
| 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; |
| |
| webrtc::VideoContentType content_type; |
| |
| // 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_ |