blob: 2b0ef812775da1650332bab12631046c815f0690 [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 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/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 "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"
#include "rtc_base/task_utils/pending_task_safety_flag.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.
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);
// 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;
};
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;
// 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;
bool typing_noise_detected = false;
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;
double jitter_buffer_target_delay_seconds = 0.0;
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
uint64_t total_decode_time_ms = 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_