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Diffstat (limited to 'third_party/libwebrtc/media/base/media_channel.h')
| -rw-r--r-- | third_party/libwebrtc/media/base/media_channel.h | 945 |
1 files changed, 945 insertions, 0 deletions
diff --git a/third_party/libwebrtc/media/base/media_channel.h b/third_party/libwebrtc/media/base/media_channel.h new file mode 100644 index 0000000000..57b488479b --- /dev/null +++ b/third_party/libwebrtc/media/base/media_channel.h @@ -0,0 +1,945 @@ +/* + * 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 MEDIA_BASE_MEDIA_CHANNEL_H_ +#define MEDIA_BASE_MEDIA_CHANNEL_H_ + +#include <map> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_options.h" +#include "api/crypto/frame_decryptor_interface.h" +#include "api/crypto/frame_encryptor_interface.h" +#include "api/frame_transformer_interface.h" +#include "api/media_stream_interface.h" +#include "api/rtc_error.h" +#include "api/rtp_parameters.h" +#include "api/task_queue/pending_task_safety_flag.h" +#include "api/transport/data_channel_transport_interface.h" +#include "api/transport/rtp/rtp_source.h" +#include "api/units/time_delta.h" +#include "api/video/video_content_type.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "api/video/video_timing.h" +#include "api/video_codecs/video_encoder_config.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "call/video_receive_stream.h" +#include "common_video/include/quality_limitation_reason.h" +#include "media/base/codec.h" +#include "media/base/delayable.h" +#include "media/base/media_constants.h" +#include "media/base/stream_params.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" +#include "modules/rtp_rtcp/include/report_block_data.h" +#include "rtc_base/async_packet_socket.h" +#include "rtc_base/buffer.h" +#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/dscp.h" +#include "rtc_base/logging.h" +#include "rtc_base/network_route.h" +#include "rtc_base/socket.h" +#include "rtc_base/string_encode.h" +#include "rtc_base/strings/string_builder.h" + +namespace rtc { +class Timing; +} + +namespace webrtc { +class AudioSinkInterface; +class VideoFrame; +} // namespace webrtc + +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 absl::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) { + rtc::StringBuilder ost; // no-presubmit-check TODO(webrtc:8982) + ost << "["; + for (size_t i = 0; i < vals.size(); ++i) { + if (i > 0) { + ost << ", "; + } + ost << vals[i].ToString(); + } + ost << "]"; + return ost.Release(); +} + +// 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 { + VideoOptions(); + ~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 { + rtc::StringBuilder 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.Release(); + } + + // Enable denoising? This flag comes from the getUserMedia + // constraint 'googNoiseReduction', and WebRtcVideoEngine passes it + // on to the codec options. Disabled by default. + absl::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. + // TODO(https://crbug.com/1315155): Remove the ability to set it in Chromium + // and delete this flag (it should default to 100 kbps). + absl::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. + absl::optional<bool> is_screencast; + webrtc::VideoTrackInterface::ContentHint content_hint; + + private: + template <typename T> + static void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) { + if (o) { + *s = o; + } + } +}; + +class MediaChannel { + 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(webrtc::TaskQueueBase* network_thread, + bool enable_dscp = false); + virtual ~MediaChannel(); + + virtual cricket::MediaType media_type() const = 0; + + // Sets the abstract interface class for sending RTP/RTCP data. + virtual void SetInterface(NetworkInterface* iface); + // Called on the network when an RTP packet is received. + virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet, + int64_t packet_time_us) = 0; + // Called on the network thread after a transport has finished sending a + // packet. + virtual void OnPacketSent(const rtc::SentPacket& sent_packet) = 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( + absl::string_view transport_name, + const rtc::NetworkRoute& network_route) = 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. In the case of an ssrc of 0, the possibly cached + // StreamParams is removed. + virtual bool RemoveSendStream(uint32_t ssrc) = 0; + // Creates a new incoming media stream with SSRCs, CNAME as described + // by sp. In the case of a sp without SSRCs, the unsignaled sp is cached + // to be used later for unsignaled streams received. + 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; + // Resets any cached StreamParams for an unsignaled RecvStream, and removes + // any existing unsignaled streams. + virtual void ResetUnsignaledRecvStream() = 0; + // This is currently a workaround because of the demuxer state being managed + // across two separate threads. Once the state is consistently managed on + // the same thread (network), this workaround can be removed. + // These two notifications inform the media channel when the transport's + // demuxer criteria is being updated. + // * OnDemuxerCriteriaUpdatePending() happens on the same thread that the + // channel's streams are added and removed (worker thread). + // * OnDemuxerCriteriaUpdateComplete() happens on the same thread. + // Because the demuxer is updated asynchronously, there is a window of time + // where packets are arriving to the channel for streams that have already + // been removed on the worker thread. It is important NOT to treat these as + // new unsignalled ssrcs. + virtual void OnDemuxerCriteriaUpdatePending() = 0; + virtual void OnDemuxerCriteriaUpdateComplete() = 0; + // Returns the absoulte sendtime extension id value from media channel. + virtual int GetRtpSendTimeExtnId() const; + // Set the frame encryptor to use on all outgoing frames. This is optional. + // This pointers lifetime is managed by the set of RtpSender it is attached + // to. + // TODO(benwright) make pure virtual once internal supports it. + virtual void SetFrameEncryptor( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor); + // Set the frame decryptor to use on all incoming frames. This is optional. + // This pointers lifetimes is managed by the set of RtpReceivers it is + // attached to. + // TODO(benwright) make pure virtual once internal supports it. + virtual void SetFrameDecryptor( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor); + + // Enable network condition based codec switching. + virtual void SetVideoCodecSwitchingEnabled(bool enabled); + + // note: The encoder_selector object must remain valid for the lifetime of the + // MediaChannel, unless replaced. + virtual void SetEncoderSelector( + uint32_t ssrc, + webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) { + } + + // Base method to send packet using NetworkInterface. + bool SendPacket(rtc::CopyOnWriteBuffer* packet, + const rtc::PacketOptions& options); + + bool SendRtcp(rtc::CopyOnWriteBuffer* packet, + const rtc::PacketOptions& options); + + int SetOption(NetworkInterface::SocketType type, + rtc::Socket::Option opt, + int option); + + // Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285. + // Set to true if it's allowed to mix one- and two-byte RTP header extensions + // in the same stream. The setter and getter must only be called from + // worker_thread. + void SetExtmapAllowMixed(bool extmap_allow_mixed); + bool ExtmapAllowMixed() const; + + // Returns `true` if a non-null NetworkInterface pointer is held. + // Must be called on the network thread. + bool HasNetworkInterface() const; + + virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0; + virtual webrtc::RTCError SetRtpSendParameters( + uint32_t ssrc, + const webrtc::RtpParameters& parameters) = 0; + + virtual void SetEncoderToPacketizerFrameTransformer( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer); + virtual void SetDepacketizerToDecoderFrameTransformer( + uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer); + + protected: + int SetOptionLocked(NetworkInterface::SocketType type, + rtc::Socket::Option opt, + int option) RTC_RUN_ON(network_thread_); + + bool DscpEnabled() const; + + // This is the DSCP value used for both RTP and RTCP channels if DSCP is + // enabled. It can be changed at any time via `SetPreferredDscp`. + rtc::DiffServCodePoint PreferredDscp() const; + void SetPreferredDscp(rtc::DiffServCodePoint new_dscp); + + rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> network_safety(); + + // Utility implementation for derived classes (video/voice) that applies + // the packet options and passes the data onwards to `SendPacket`. + void SendRtp(const uint8_t* data, + size_t len, + const webrtc::PacketOptions& options); + + void SendRtcp(const uint8_t* data, size_t len); + + private: + // Apply the preferred DSCP setting to the underlying network interface RTP + // and RTCP channels. If DSCP is disabled, then apply the default DSCP value. + void UpdateDscp() RTC_RUN_ON(network_thread_); + + bool DoSendPacket(rtc::CopyOnWriteBuffer* packet, + bool rtcp, + const rtc::PacketOptions& options); + + const bool enable_dscp_; + const rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> network_safety_ + RTC_PT_GUARDED_BY(network_thread_); + webrtc::TaskQueueBase* const network_thread_; + NetworkInterface* network_interface_ RTC_GUARDED_BY(network_thread_) = + nullptr; + rtc::DiffServCodePoint preferred_dscp_ RTC_GUARDED_BY(network_thread_) = + rtc::DSCP_DEFAULT; + bool extmap_allow_mixed_ = false; +}; + +// 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 { + uint32_t ssrc = 0; + double timestamp = 0.0; // NTP timestamp, represented as seconds since epoch. +}; + +struct SsrcReceiverInfo { + uint32_t ssrc = 0; + double timestamp = 0.0; +}; + +struct MediaSenderInfo { + MediaSenderInfo(); + ~MediaSenderInfo(); + 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; + } + // Returns true if the media has been connected. + bool connected() const { return local_stats.size() > 0; } + // Utility accessor for clients that make the assumption only one ssrc + // exists per media. + // This will eventually go away. + // Call sites that compare this to zero should use connected() instead. + // https://bugs.webrtc.org/8694 + uint32_t ssrc() const { + if (connected()) { + return local_stats[0].ssrc; + } else { + return 0; + } + } + // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-bytessent + int64_t payload_bytes_sent = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-headerbytessent + int64_t header_and_padding_bytes_sent = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent + uint64_t retransmitted_bytes_sent = 0; + int packets_sent = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent + uint64_t retransmitted_packets_sent = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-nackcount + uint32_t nacks_rcvd = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-targetbitrate + double target_bitrate = 0.0; + int packets_lost = 0; + float fraction_lost = 0.0f; + int64_t rtt_ms = 0; + std::string codec_name; + absl::optional<int> codec_payload_type; + std::vector<SsrcSenderInfo> local_stats; + std::vector<SsrcReceiverInfo> remote_stats; + // A snapshot of the most recent Report Block with additional data of interest + // to statistics. Used to implement RTCRemoteInboundRtpStreamStats. Within + // this list, the ReportBlockData::RTCPReportBlock::source_ssrc(), which is + // the SSRC of the corresponding outbound RTP stream, is unique. + std::vector<webrtc::ReportBlockData> report_block_datas; + absl::optional<bool> active; +}; + +struct MediaReceiverInfo { + MediaReceiverInfo(); + ~MediaReceiverInfo(); + 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; + } + // Returns true if the media has been connected. + bool connected() const { return local_stats.size() > 0; } + // Utility accessor for clients that make the assumption only one ssrc + // exists per media. + // This will eventually go away. + // Call sites that compare this to zero should use connected(); + // https://bugs.webrtc.org/8694 + uint32_t ssrc() const { + if (connected()) { + return local_stats[0].ssrc; + } else { + return 0; + } + } + + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-bytesreceived + int64_t payload_bytes_rcvd = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-headerbytesreceived + int64_t header_and_padding_bytes_rcvd = 0; + int packets_rcvd = 0; + int packets_lost = 0; + absl::optional<uint32_t> nacks_sent; + // Jitter (network-related) latency (cumulative). + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay + double jitter_buffer_delay_seconds = 0.0; + // Target delay for the jitter buffer (cumulative). + // TODO(crbug.com/webrtc/14244): This metric is only implemented for + // audio, it should be implemented for video as well. + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbuffertargetdelay + absl::optional<double> jitter_buffer_target_delay_seconds; + // Minimum obtainable delay for the jitter buffer (cumulative). + // TODO(crbug.com/webrtc/14244): This metric is only implemented for + // audio, it should be implemented for video as well. + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay + absl::optional<double> jitter_buffer_minimum_delay_seconds; + // Number of observations for cumulative jitter latency. + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferemittedcount + uint64_t jitter_buffer_emitted_count = 0; + // The timestamp at which the last packet was received, i.e. the time of the + // local clock when it was received - not the RTP timestamp of that packet. + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp + absl::optional<int64_t> last_packet_received_timestamp_ms; + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp + absl::optional<int64_t> estimated_playout_ntp_timestamp_ms; + std::string codec_name; + absl::optional<int> codec_payload_type; + std::vector<SsrcReceiverInfo> local_stats; + std::vector<SsrcSenderInfo> remote_stats; +}; + +struct VoiceSenderInfo : public MediaSenderInfo { + VoiceSenderInfo(); + ~VoiceSenderInfo(); + int jitter_ms = 0; + // Current audio level, expressed linearly [0,32767]. + int audio_level = 0; + // See description of "totalAudioEnergy" in the WebRTC stats spec: + // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy + double total_input_energy = 0.0; + double total_input_duration = 0.0; + webrtc::ANAStats ana_statistics; + webrtc::AudioProcessingStats apm_statistics; +}; + +struct VoiceReceiverInfo : public MediaReceiverInfo { + VoiceReceiverInfo(); + ~VoiceReceiverInfo(); + int jitter_ms = 0; + int jitter_buffer_ms = 0; + int jitter_buffer_preferred_ms = 0; + int delay_estimate_ms = 0; + int audio_level = 0; + // Stats below correspond to similarly-named fields in the WebRTC stats spec. + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats + double total_output_energy = 0.0; + uint64_t total_samples_received = 0; + double total_output_duration = 0.0; + uint64_t concealed_samples = 0; + uint64_t silent_concealed_samples = 0; + uint64_t concealment_events = 0; + uint64_t inserted_samples_for_deceleration = 0; + uint64_t removed_samples_for_acceleration = 0; + uint64_t fec_packets_received = 0; + uint64_t fec_packets_discarded = 0; + // Stats below correspond to similarly-named fields in the WebRTC stats spec. + // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats + uint64_t packets_discarded = 0; + // Stats below DO NOT correspond directly to anything in the WebRTC stats + // fraction of synthesized audio inserted through expansion. + float expand_rate = 0.0f; + // fraction of synthesized speech inserted through expansion. + float speech_expand_rate = 0.0f; + // fraction of data out of secondary decoding, including FEC and RED. + float secondary_decoded_rate = 0.0f; + // 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 = 0.0f; + // Fraction of data removed through time compression. + float accelerate_rate = 0.0f; + // Fraction of data inserted through time stretching. + float preemptive_expand_rate = 0.0f; + int decoding_calls_to_silence_generator = 0; + int decoding_calls_to_neteq = 0; + int decoding_normal = 0; + // TODO(alexnarest): Consider decoding_neteq_plc for consistency + int decoding_plc = 0; + int decoding_codec_plc = 0; + int decoding_cng = 0; + int decoding_plc_cng = 0; + int decoding_muted_output = 0; + // Estimated capture start time in NTP time in ms. + int64_t capture_start_ntp_time_ms = -1; + // Count of the number of buffer flushes. + uint64_t jitter_buffer_flushes = 0; + // Number of samples expanded due to delayed packets. + uint64_t delayed_packet_outage_samples = 0; + // Arrival delay of received audio packets. + double relative_packet_arrival_delay_seconds = 0.0; + // Count and total duration of audio interruptions (loss-concealement periods + // longer than 150 ms). + int32_t interruption_count = 0; + int32_t total_interruption_duration_ms = 0; + // Remote outbound stats derived by the received RTCP sender reports. + // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict* + absl::optional<int64_t> last_sender_report_timestamp_ms; + absl::optional<int64_t> last_sender_report_remote_timestamp_ms; + uint32_t sender_reports_packets_sent = 0; + uint64_t sender_reports_bytes_sent = 0; + uint64_t sender_reports_reports_count = 0; + absl::optional<webrtc::TimeDelta> round_trip_time; + webrtc::TimeDelta total_round_trip_time = webrtc::TimeDelta::Zero(); + int round_trip_time_measurements = 0; +}; + +struct VideoSenderInfo : public MediaSenderInfo { + VideoSenderInfo(); + ~VideoSenderInfo(); + std::vector<SsrcGroup> ssrc_groups; + std::string encoder_implementation_name; + int firs_rcvd = 0; + int plis_rcvd = 0; + int send_frame_width = 0; + int send_frame_height = 0; + int frames = 0; + double framerate_input = 0; + int framerate_sent = 0; + int aggregated_framerate_sent = 0; + int nominal_bitrate = 0; + int adapt_reason = 0; + int adapt_changes = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationreason + webrtc::QualityLimitationReason quality_limitation_reason = + webrtc::QualityLimitationReason::kNone; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations + std::map<webrtc::QualityLimitationReason, int64_t> + quality_limitation_durations_ms; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges + uint32_t quality_limitation_resolution_changes = 0; + int avg_encode_ms = 0; + int encode_usage_percent = 0; + uint32_t frames_encoded = 0; + uint32_t key_frames_encoded = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime + uint64_t total_encode_time_ms = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget + uint64_t total_encoded_bytes_target = 0; + uint64_t total_packet_send_delay_ms = 0; + bool has_entered_low_resolution = false; + absl::optional<uint64_t> qp_sum; + webrtc::VideoContentType content_type = webrtc::VideoContentType::UNSPECIFIED; + uint32_t frames_sent = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcvideosenderstats-hugeframessent + uint32_t huge_frames_sent = 0; + uint32_t aggregated_huge_frames_sent = 0; + absl::optional<std::string> rid; +}; + +struct VideoReceiverInfo : public MediaReceiverInfo { + VideoReceiverInfo(); + ~VideoReceiverInfo(); + std::vector<SsrcGroup> ssrc_groups; + std::string decoder_implementation_name; + int packets_concealed = 0; + int firs_sent = 0; + int plis_sent = 0; + int frame_width = 0; + int frame_height = 0; + int framerate_rcvd = 0; + int framerate_decoded = 0; + int framerate_output = 0; + // Framerate as sent to the renderer. + int framerate_render_input = 0; + // Framerate that the renderer reports. + int framerate_render_output = 0; + uint32_t frames_received = 0; + uint32_t frames_dropped = 0; + uint32_t frames_decoded = 0; + uint32_t key_frames_decoded = 0; + uint32_t frames_rendered = 0; + absl::optional<uint64_t> qp_sum; + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime + webrtc::TimeDelta total_decode_time = webrtc::TimeDelta::Zero(); + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay + webrtc::TimeDelta total_processing_delay = webrtc::TimeDelta::Zero(); + webrtc::TimeDelta total_assembly_time = webrtc::TimeDelta::Zero(); + uint32_t frames_assembled_from_multiple_packets = 0; + double total_inter_frame_delay = 0; + double total_squared_inter_frame_delay = 0; + int64_t interframe_delay_max_ms = -1; + uint32_t freeze_count = 0; + uint32_t pause_count = 0; + uint32_t total_freezes_duration_ms = 0; + uint32_t total_pauses_duration_ms = 0; + uint32_t total_frames_duration_ms = 0; + double sum_squared_frame_durations = 0.0; + uint32_t jitter_ms = 0; + + webrtc::VideoContentType content_type = webrtc::VideoContentType::UNSPECIFIED; + + // 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 = 0; + // Maximum observed frame decode latency. + int max_decode_ms = 0; + // Jitter (network-related) latency. + int jitter_buffer_ms = 0; + // Requested minimum playout latency. + int min_playout_delay_ms = 0; + // Requested latency to account for rendering delay. + int render_delay_ms = 0; + // Target overall delay: network+decode+render, accounting for + // min_playout_delay_ms. + int target_delay_ms = 0; + // Current overall delay, possibly ramping towards target_delay_ms. + int current_delay_ms = 0; + + // Estimated capture start time in NTP time in ms. + int64_t capture_start_ntp_time_ms = -1; + + // First frame received to first frame decoded latency. + int64_t first_frame_received_to_decoded_ms = -1; + + // Timing frame info: all important timestamps for a full lifetime of a + // single 'timing frame'. + absl::optional<webrtc::TimingFrameInfo> timing_frame_info; +}; + +struct BandwidthEstimationInfo { + int available_send_bandwidth = 0; + int available_recv_bandwidth = 0; + int target_enc_bitrate = 0; + int actual_enc_bitrate = 0; + int retransmit_bitrate = 0; + int transmit_bitrate = 0; + int64_t bucket_delay = 0; +}; + +// Maps from payload type to `RtpCodecParameters`. +typedef std::map<int, webrtc::RtpCodecParameters> RtpCodecParametersMap; + +struct VoiceMediaInfo { + VoiceMediaInfo(); + ~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; + int32_t device_underrun_count = 0; +}; + +struct VideoMediaInfo { + VideoMediaInfo(); + ~VideoMediaInfo(); + void Clear() { + senders.clear(); + aggregated_senders.clear(); + receivers.clear(); + send_codecs.clear(); + receive_codecs.clear(); + } + // Each sender info represents one "outbound-rtp" stream.In non - simulcast, + // this means one info per RtpSender but if simulcast is used this means + // one info per simulcast layer. + std::vector<VideoSenderInfo> senders; + // Used for legacy getStats() API's "ssrc" stats and modern getStats() API's + // "track" stats. If simulcast is used, instead of having one sender info per + // simulcast layer, the metrics of all layers of an RtpSender are aggregated + // into a single sender info per RtpSender. + std::vector<VideoSenderInfo> aggregated_senders; + std::vector<VideoReceiverInfo> receivers; + RtpCodecParametersMap send_codecs; + RtpCodecParametersMap receive_codecs; +}; + +struct RtcpParameters { + bool reduced_size = false; + bool remote_estimate = false; +}; + +template <class Codec> +struct RtpParameters { + virtual ~RtpParameters() = default; + + std::vector<Codec> codecs; + std::vector<webrtc::RtpExtension> extensions; + // For a send stream this is true if we've neogtiated a send direction, + // for a receive stream this is true if we've negotiated a receive direction. + bool is_stream_active = true; + + // TODO(pthatcher): Add streams. + RtcpParameters rtcp; + + std::string ToString() const { + rtc::StringBuilder ost; + ost << "{"; + const char* separator = ""; + for (const auto& entry : ToStringMap()) { + ost << separator << entry.first << ": " << entry.second; + separator = ", "; + } + ost << "}"; + return ost.Release(); + } + + protected: + virtual std::map<std::string, std::string> ToStringMap() const { + return {{"codecs", VectorToString(codecs)}, + {"extensions", VectorToString(extensions)}}; + } +}; + +// 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> { + int max_bandwidth_bps = -1; + // This is the value to be sent in the MID RTP header extension (if the header + // extension in included in the list of extensions). + std::string mid; + bool extmap_allow_mixed = false; + + protected: + std::map<std::string, std::string> ToStringMap() const override { + auto params = RtpParameters<Codec>::ToStringMap(); + params["max_bandwidth_bps"] = rtc::ToString(max_bandwidth_bps); + params["mid"] = (mid.empty() ? "<not set>" : mid); + params["extmap-allow-mixed"] = extmap_allow_mixed ? "true" : "false"; + return params; + } +}; + +struct AudioSendParameters : RtpSendParameters<AudioCodec> { + AudioSendParameters(); + ~AudioSendParameters() override; + AudioOptions options; + + protected: + std::map<std::string, std::string> ToStringMap() const override; +}; + +struct AudioRecvParameters : RtpParameters<AudioCodec> {}; + +class VoiceMediaChannel : public MediaChannel, public Delayable { + public: + VoiceMediaChannel(webrtc::TaskQueueBase* network_thread, + bool enable_dscp = false) + : MediaChannel(network_thread, enable_dscp) {} + ~VoiceMediaChannel() override {} + + cricket::MediaType media_type() const override; + virtual bool SetSendParameters(const AudioSendParameters& params) = 0; + virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0; + // Get the receive parameters for the incoming stream identified by `ssrc`. + virtual webrtc::RtpParameters GetRtpReceiveParameters( + uint32_t ssrc) const = 0; + // Retrieve the receive parameters for the default receive + // stream, which is used when SSRCs are not signaled. + virtual webrtc::RtpParameters GetDefaultRtpReceiveParameters() const = 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; + // Set speaker output volume of the specified ssrc. + virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0; + // Set speaker output volume for future unsignaled streams. + virtual bool SetDefaultOutputVolume(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, + bool get_and_clear_legacy_stats) = 0; + + virtual void SetRawAudioSink( + uint32_t ssrc, + std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0; + virtual void SetDefaultRawAudioSink( + 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> { + VideoSendParameters(); + ~VideoSendParameters() override; + // 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; + + protected: + std::map<std::string, std::string> ToStringMap() const override; +}; + +// 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 Delayable { + public: + explicit VideoMediaChannel(webrtc::TaskQueueBase* network_thread, + bool enable_dscp = false) + : MediaChannel(network_thread, enable_dscp) {} + ~VideoMediaChannel() override {} + + cricket::MediaType media_type() const override; + virtual bool SetSendParameters(const VideoSendParameters& params) = 0; + virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0; + // Get the receive parameters for the incoming stream identified by `ssrc`. + virtual webrtc::RtpParameters GetRtpReceiveParameters( + uint32_t ssrc) const = 0; + // Retrieve the receive parameters for the default receive + // stream, which is used when SSRCs are not signaled. + virtual webrtc::RtpParameters GetDefaultRtpReceiveParameters() const = 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, + const VideoOptions* options, + rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0; + // Sets the sink object to be used for the specified stream. + virtual bool SetSink(uint32_t ssrc, + rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0; + // The sink is used for the 'default' stream. + virtual void SetDefaultSink( + 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; + // Set recordable encoded frame callback for `ssrc` + virtual void SetRecordableEncodedFrameCallback( + uint32_t ssrc, + std::function<void(const webrtc::RecordableEncodedFrame&)> callback) = 0; + // Clear recordable encoded frame callback for `ssrc` + virtual void ClearRecordableEncodedFrameCallback(uint32_t ssrc) = 0; + // Cause generation of a keyframe for `ssrc` + virtual void GenerateKeyFrame(uint32_t ssrc) = 0; + + virtual std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const = 0; +}; + +// 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. + // SCTP data channels use SIDs. + int sid = 0; + // The type of message (binary, text, or control). + webrtc::DataMessageType type = webrtc::DataMessageType::kText; + // A per-stream value incremented per packet in the stream. + int seq_num = 0; +}; + +enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK }; + +} // namespace cricket + +#endif // MEDIA_BASE_MEDIA_CHANNEL_H_ |