media/cast/sender/frame_sender.cc - cobalt - Git at Google

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.

 #include "media/cast/sender/frame_sender.h"

 #include <algorithm>
 #include <limits>
 #include <memory>
 #include <utility>
 #include <vector>

 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/macros.h"
 #include "base/numerics/safe_conversions.h"
 #include "base/trace_event/trace_event.h"
 #include "media/cast/constants.h"
 #include "media/cast/sender/sender_encoded_frame.h"

 namespace media {
 namespace cast {
 namespace {

 constexpr int kNumAggressiveReportsSentAtStart = 100;
 constexpr base::TimeDelta kMinSchedulingDelay = base::Milliseconds(1);
 constexpr base::TimeDelta kReceiverProcessTime = base::Milliseconds(250);

 // The additional number of frames that can be in-flight when input exceeds the
 // maximum frame rate.
 constexpr int kMaxFrameBurst = 5;

 }  // namespace

 // Convenience macro used in logging statements throughout this file.
 #define SENDER_SSRC (is_audio_ ? "AUDIO[" : "VIDEO[") << ssrc_ << "] "

 FrameSender::RtcpClient::RtcpClient(base::WeakPtr<FrameSender> frame_sender)
     : frame_sender_(frame_sender) {}

 FrameSender::RtcpClient::~RtcpClient() = default;

 void FrameSender::RtcpClient::OnReceivedCastMessage(
     const RtcpCastMessage& cast_message) {
   if (frame_sender_)
     frame_sender_->OnReceivedCastFeedback(cast_message);
 }

 void FrameSender::RtcpClient::OnReceivedRtt(base::TimeDelta round_trip_time) {
   if (frame_sender_)
     frame_sender_->OnMeasuredRoundTripTime(round_trip_time);
 }

 void FrameSender::RtcpClient::OnReceivedPli() {
   if (frame_sender_)
     frame_sender_->OnReceivedPli();
 }

 FrameSender::FrameSender(scoped_refptr<CastEnvironment> cast_environment,
                          CastTransport* const transport_sender,
                          const FrameSenderConfig& config,
                          CongestionControl* congestion_control)
     : cast_environment_(cast_environment),
       transport_sender_(transport_sender),
       ssrc_(config.sender_ssrc),
       min_playout_delay_(config.min_playout_delay.is_zero()
                              ? config.max_playout_delay
                              : config.min_playout_delay),
       max_playout_delay_(config.max_playout_delay),
       animated_playout_delay_(config.animated_playout_delay.is_zero()
                                   ? config.max_playout_delay
                                   : config.animated_playout_delay),
       send_target_playout_delay_(false),
       max_frame_rate_(config.max_frame_rate),
       num_aggressive_rtcp_reports_sent_(0),
       duplicate_ack_counter_(0),
       congestion_control_(congestion_control),
       picture_lost_at_receiver_(false),
       rtp_timebase_(config.rtp_timebase),
       is_audio_(config.rtp_payload_type <= RtpPayloadType::AUDIO_LAST),
       max_ack_delay_(config.max_playout_delay) {
   DCHECK(transport_sender_);
   DCHECK_GT(rtp_timebase_, 0);
   DCHECK(congestion_control_);
   // We assume animated content to begin with since that is the common use
   // case today.
   VLOG(1) << SENDER_SSRC << "min latency "
           << min_playout_delay_.InMilliseconds() << "max latency "
           << max_playout_delay_.InMilliseconds() << "animated latency "
           << animated_playout_delay_.InMilliseconds();
   SetTargetPlayoutDelay(animated_playout_delay_);

   CastTransportRtpConfig transport_config;
   transport_config.ssrc = config.sender_ssrc;
   transport_config.feedback_ssrc = config.receiver_ssrc;
   transport_config.rtp_payload_type = config.rtp_payload_type;
   transport_config.aes_key = config.aes_key;
   transport_config.aes_iv_mask = config.aes_iv_mask;

   transport_sender->InitializeStream(
       transport_config,
       std::make_unique<FrameSender::RtcpClient>(weak_factory_.GetWeakPtr()));
 }

 FrameSender::~FrameSender() = default;

 void FrameSender::ScheduleNextRtcpReport() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

   cast_environment_->PostDelayedTask(
       CastEnvironment::MAIN, FROM_HERE,
       base::BindOnce(&FrameSender::SendRtcpReport, weak_factory_.GetWeakPtr(),
                      true),
       base::Milliseconds(kRtcpReportIntervalMs));
 }

 void FrameSender::SendRtcpReport(bool schedule_future_reports) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

   // Sanity-check: We should have sent at least the first frame by this point.
   DCHECK(!last_send_time_.is_null());

   // Create lip-sync info for the sender report.  The last sent frame's
   // reference time and RTP timestamp are used to estimate an RTP timestamp in
   // terms of "now."  Note that |now| is never likely to be precise to an exact
   // frame boundary; and so the computation here will result in a
   // |now_as_rtp_timestamp| value that is rarely equal to any one emitted by the
   // encoder.
   const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
   const base::TimeDelta time_delta =
       now - GetRecordedReferenceTime(last_sent_frame_id_);
   const RtpTimeDelta rtp_delta =
       RtpTimeDelta::FromTimeDelta(time_delta, rtp_timebase_);
   const RtpTimeTicks now_as_rtp_timestamp =
       GetRecordedRtpTimestamp(last_sent_frame_id_) + rtp_delta;
   transport_sender_->SendSenderReport(ssrc_, now, now_as_rtp_timestamp);

   if (schedule_future_reports)
     ScheduleNextRtcpReport();
 }

 void FrameSender::OnMeasuredRoundTripTime(base::TimeDelta round_trip_time) {
   DCHECK_GT(round_trip_time, base::TimeDelta());
   current_round_trip_time_ = round_trip_time;
   max_ack_delay_ = 2 * std::max(current_round_trip_time_, base::TimeDelta()) +
                    kReceiverProcessTime;
   max_ack_delay_ = std::min(max_ack_delay_, target_playout_delay_);
 }

 void FrameSender::SetTargetPlayoutDelay(
     base::TimeDelta new_target_playout_delay) {
   if (send_target_playout_delay_ &&
       target_playout_delay_ == new_target_playout_delay) {
     return;
   }
   new_target_playout_delay = std::max(new_target_playout_delay,
                                       min_playout_delay_);
   new_target_playout_delay = std::min(new_target_playout_delay,
                                       max_playout_delay_);
   VLOG(2) << SENDER_SSRC << "Target playout delay changing from "
           << target_playout_delay_.InMilliseconds() << " ms to "
           << new_target_playout_delay.InMilliseconds() << " ms.";
   target_playout_delay_ = new_target_playout_delay;
   max_ack_delay_ = std::min(max_ack_delay_, target_playout_delay_);
   send_target_playout_delay_ = true;
   congestion_control_->UpdateTargetPlayoutDelay(target_playout_delay_);
 }

 void FrameSender::ResendCheck() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   DCHECK(!last_send_time_.is_null());
   const base::TimeDelta time_since_last_send =
       cast_environment_->Clock()->NowTicks() - last_send_time_;
   if (time_since_last_send > max_ack_delay_) {
     if (latest_acked_frame_id_ == last_sent_frame_id_) {
       // Last frame acked, no point in doing anything
     } else {
       VLOG(1) << SENDER_SSRC << "ACK timeout; last acked frame: "
               << latest_acked_frame_id_;
       ResendForKickstart();
     }
   }
   ScheduleNextResendCheck();
 }

 void FrameSender::ScheduleNextResendCheck() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   DCHECK(!last_send_time_.is_null());
   base::TimeDelta time_to_next =
       last_send_time_ - cast_environment_->Clock()->NowTicks() + max_ack_delay_;
   time_to_next = std::max(time_to_next, kMinSchedulingDelay);
   cast_environment_->PostDelayedTask(
       CastEnvironment::MAIN, FROM_HERE,
       base::BindOnce(&FrameSender::ResendCheck, weak_factory_.GetWeakPtr()),
       time_to_next);
 }

 void FrameSender::ResendForKickstart() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   DCHECK(!last_send_time_.is_null());
   VLOG(1) << SENDER_SSRC << "Resending last packet of frame "
           << last_sent_frame_id_ << " to kick-start.";
   last_send_time_ = cast_environment_->Clock()->NowTicks();
   transport_sender_->ResendFrameForKickstart(ssrc_, last_sent_frame_id_);
 }

 void FrameSender::RecordLatestFrameTimestamps(FrameId frame_id,
                                               base::TimeTicks reference_time,
                                               RtpTimeTicks rtp_timestamp) {
   DCHECK(!reference_time.is_null());
   frame_reference_times_[frame_id.lower_8_bits()] = reference_time;
   frame_rtp_timestamps_[frame_id.lower_8_bits()] = rtp_timestamp;
 }

 base::TimeTicks FrameSender::GetRecordedReferenceTime(FrameId frame_id) const {
   return frame_reference_times_[frame_id.lower_8_bits()];
 }

 RtpTimeTicks FrameSender::GetRecordedRtpTimestamp(FrameId frame_id) const {
   return frame_rtp_timestamps_[frame_id.lower_8_bits()];
 }

 int FrameSender::GetUnacknowledgedFrameCount() const {
   if (last_send_time_.is_null())
     return 0;
   const int count = last_sent_frame_id_ - latest_acked_frame_id_;
   DCHECK_GE(count, 0);
   return count;
 }

 base::TimeDelta FrameSender::GetAllowedInFlightMediaDuration() const {
   // The total amount allowed in-flight media should equal the amount that fits
   // within the entire playout delay window, plus the amount of time it takes to
   // receive an ACK from the receiver.
   // TODO(miu): Research is needed, but there is likely a better formula.
   return target_playout_delay_ + (current_round_trip_time_ / 2);
 }

 void FrameSender::SendEncodedFrame(
     int requested_bitrate_before_encode,
     std::unique_ptr<SenderEncodedFrame> encoded_frame) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

   VLOG(2) << SENDER_SSRC << "About to send another frame: last_sent="
           << last_sent_frame_id_ << ", latest_acked=" << latest_acked_frame_id_;

   const FrameId frame_id = encoded_frame->frame_id;
   const bool is_first_frame_to_be_sent = last_send_time_.is_null();

   if (picture_lost_at_receiver_ &&
       (encoded_frame->dependency == EncodedFrame::KEY)) {
     picture_lost_at_receiver_ = false;
     DCHECK(frame_id > latest_acked_frame_id_);
     // Cancel sending remaining frames.
     std::vector<FrameId> cancel_sending_frames;
     for (FrameId id = latest_acked_frame_id_ + 1; id < frame_id; ++id) {
       cancel_sending_frames.push_back(id);
     }
     transport_sender_->CancelSendingFrames(ssrc_, cancel_sending_frames);
     OnCancelSendingFrames();
   }

   last_send_time_ = cast_environment_->Clock()->NowTicks();
   last_sent_frame_id_ = frame_id;
   // If this is the first frame about to be sent, fake the value of
   // |latest_acked_frame_id_| to indicate the receiver starts out all caught up.
   // Also, schedule the periodic frame re-send checks.
   if (is_first_frame_to_be_sent) {
     latest_acked_frame_id_ = frame_id - 1;
     ScheduleNextResendCheck();
   }

   VLOG_IF(1, !is_audio_ && encoded_frame->dependency == EncodedFrame::KEY)
       << SENDER_SSRC << "Sending encoded key frame, id=" << frame_id;

   std::unique_ptr<FrameEvent> encode_event(new FrameEvent());
   encode_event->timestamp = encoded_frame->encode_completion_time;
   encode_event->type = FRAME_ENCODED;
   encode_event->media_type = is_audio_ ? AUDIO_EVENT : VIDEO_EVENT;
   encode_event->rtp_timestamp = encoded_frame->rtp_timestamp;
   encode_event->frame_id = frame_id;
   encode_event->size = base::checked_cast<uint32_t>(encoded_frame->data.size());
   encode_event->key_frame = encoded_frame->dependency == EncodedFrame::KEY;
   encode_event->target_bitrate = requested_bitrate_before_encode;
   encode_event->encoder_cpu_utilization = encoded_frame->encoder_utilization;
   encode_event->idealized_bitrate_utilization =
       encoded_frame->lossy_utilization;
   cast_environment_->logger()->DispatchFrameEvent(std::move(encode_event));

   RecordLatestFrameTimestamps(frame_id,
                               encoded_frame->reference_time,
                               encoded_frame->rtp_timestamp);

   if (!is_audio_) {
     // Used by chrome/browser/media/cast_mirroring_performance_browsertest.cc
     TRACE_EVENT_INSTANT1(
         "cast_perf_test", "VideoFrameEncoded",
         TRACE_EVENT_SCOPE_THREAD,
         "rtp_timestamp", encoded_frame->rtp_timestamp.lower_32_bits());
   }

   // At the start of the session, it's important to send reports before each
   // frame so that the receiver can properly compute playout times.  The reason
   // more than one report is sent is because transmission is not guaranteed,
   // only best effort, so send enough that one should almost certainly get
   // through.
   if (num_aggressive_rtcp_reports_sent_ < kNumAggressiveReportsSentAtStart) {
     // SendRtcpReport() will schedule future reports to be made if this is the
     // last "aggressive report."
     ++num_aggressive_rtcp_reports_sent_;
     const bool is_last_aggressive_report =
         (num_aggressive_rtcp_reports_sent_ == kNumAggressiveReportsSentAtStart);
     VLOG_IF(1, is_last_aggressive_report)
         << SENDER_SSRC << "Sending last aggressive report.";
     SendRtcpReport(is_last_aggressive_report);
   }

   congestion_control_->SendFrameToTransport(
       frame_id, encoded_frame->data.size() * 8, last_send_time_);

   if (send_target_playout_delay_) {
     encoded_frame->new_playout_delay_ms =
         target_playout_delay_.InMilliseconds();
   }

   const char* name = is_audio_ ? "Audio Transport" : "Video Transport";
   TRACE_EVENT_NESTABLE_ASYNC_BEGIN1(
       "cast.stream", name, TRACE_ID_WITH_SCOPE(name, frame_id.lower_32_bits()),
       "rtp_timestamp", encoded_frame->rtp_timestamp.lower_32_bits());
   transport_sender_->InsertFrame(ssrc_, *encoded_frame);
 }

 void FrameSender::OnCancelSendingFrames() {}

 void FrameSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

   const bool have_valid_rtt = current_round_trip_time_ > base::TimeDelta();
   if (have_valid_rtt) {
     congestion_control_->UpdateRtt(current_round_trip_time_);

     // Having the RTT value implies the receiver sent back a receiver report
     // based on it having received a report from here.  Therefore, ensure this
     // sender stops aggressively sending reports.
     if (num_aggressive_rtcp_reports_sent_ < kNumAggressiveReportsSentAtStart) {
       VLOG(1) << SENDER_SSRC
               << "No longer a need to send reports aggressively (sent "
               << num_aggressive_rtcp_reports_sent_ << ").";
       num_aggressive_rtcp_reports_sent_ = kNumAggressiveReportsSentAtStart;
       ScheduleNextRtcpReport();
     }
   }

   if (last_send_time_.is_null())
     return;  // Cannot get an ACK without having first sent a frame.

   if (cast_feedback.missing_frames_and_packets.empty() &&
       cast_feedback.received_later_frames.empty()) {
     if (latest_acked_frame_id_ == cast_feedback.ack_frame_id) {
       VLOG(1) << SENDER_SSRC << "Received duplicate ACK for frame "
               << latest_acked_frame_id_;
       TRACE_EVENT_INSTANT2(
           "cast.stream", "Duplicate ACK", TRACE_EVENT_SCOPE_THREAD,
           "ack_frame_id", cast_feedback.ack_frame_id.lower_32_bits(),
           "last_sent_frame_id", last_sent_frame_id_.lower_32_bits());
     }
     // We only count duplicate ACKs when we have sent newer frames.
     if (latest_acked_frame_id_ == cast_feedback.ack_frame_id &&
         latest_acked_frame_id_ != last_sent_frame_id_) {
       duplicate_ack_counter_++;
     } else {
       duplicate_ack_counter_ = 0;
     }
     // TODO(miu): The values "2" and "3" should be derived from configuration.
     if (duplicate_ack_counter_ >= 2 && duplicate_ack_counter_ % 3 == 2) {
       ResendForKickstart();
     }
   } else {
     // Only count duplicated ACKs if there is no NACK request in between.
     // This is to avoid aggresive resend.
     duplicate_ack_counter_ = 0;
   }

   base::TimeTicks now = cast_environment_->Clock()->NowTicks();
   congestion_control_->AckFrame(cast_feedback.ack_frame_id, now);
   if (!cast_feedback.received_later_frames.empty()) {
     // Ack the received frames.
     congestion_control_->AckLaterFrames(cast_feedback.received_later_frames,
                                         now);
   }

   std::unique_ptr<FrameEvent> ack_event(new FrameEvent());
   ack_event->timestamp = now;
   ack_event->type = FRAME_ACK_RECEIVED;
   ack_event->media_type = is_audio_ ? AUDIO_EVENT : VIDEO_EVENT;
   ack_event->rtp_timestamp =
       GetRecordedRtpTimestamp(cast_feedback.ack_frame_id);
   ack_event->frame_id = cast_feedback.ack_frame_id;
   cast_environment_->logger()->DispatchFrameEvent(std::move(ack_event));

   const bool is_acked_out_of_order =
       cast_feedback.ack_frame_id < latest_acked_frame_id_;
   VLOG(2) << SENDER_SSRC
           << "Received ACK" << (is_acked_out_of_order ? " out-of-order" : "")
           << " for frame " << cast_feedback.ack_frame_id;
   if (is_acked_out_of_order) {
     TRACE_EVENT_INSTANT2(
         "cast.stream", "ACK out of order", TRACE_EVENT_SCOPE_THREAD,
         "ack_frame_id", cast_feedback.ack_frame_id.lower_32_bits(),
         "latest_acked_frame_id", latest_acked_frame_id_.lower_32_bits());
   } else if (latest_acked_frame_id_ < cast_feedback.ack_frame_id) {
     // Cancel resends of acked frames.
     std::vector<FrameId> frames_to_cancel;
     frames_to_cancel.reserve(cast_feedback.ack_frame_id -
                              latest_acked_frame_id_);
     do {
       ++latest_acked_frame_id_;
       frames_to_cancel.push_back(latest_acked_frame_id_);
       // This is a good place to match the trace for frame ids
       // since this ensures we not only track frame ids that are
       // implicitly ACKed, but also handles duplicate ACKs
       const char* name = is_audio_ ? "Audio Transport" : "Video Transport";
       TRACE_EVENT_NESTABLE_ASYNC_END1(
           "cast.stream", name,
           TRACE_ID_WITH_SCOPE(name, latest_acked_frame_id_.lower_32_bits()),
           "RTT_usecs", current_round_trip_time_.InMicroseconds());
     } while (latest_acked_frame_id_ < cast_feedback.ack_frame_id);
     transport_sender_->CancelSendingFrames(ssrc_, frames_to_cancel);
     OnCancelSendingFrames();
   }
 }

 void FrameSender::OnReceivedPli() {
   picture_lost_at_receiver_ = true;
 }

 bool FrameSender::ShouldDropNextFrame(base::TimeDelta frame_duration) const {
   // Check that accepting the next frame won't cause more frames to become
   // in-flight than the system's design limit.
   const int count_frames_in_flight =
       GetUnacknowledgedFrameCount() + GetNumberOfFramesInEncoder();
   if (count_frames_in_flight >= kMaxUnackedFrames) {
     VLOG(1) << SENDER_SSRC << "Dropping: Too many frames would be in-flight.";
     return true;
   }

   // Check that accepting the next frame won't exceed the configured maximum
   // frame rate, allowing for short-term bursts.
   base::TimeDelta duration_in_flight = GetInFlightMediaDuration();
   const double max_frames_in_flight =
       max_frame_rate_ * duration_in_flight.InSecondsF();
   if (count_frames_in_flight >= max_frames_in_flight + kMaxFrameBurst) {
     VLOG(1) << SENDER_SSRC << "Dropping: Burst threshold would be exceeded.";
     return true;
   }

   // Check that accepting the next frame won't exceed the allowed in-flight
   // media duration.
   const base::TimeDelta duration_would_be_in_flight =
       duration_in_flight + frame_duration;
   const base::TimeDelta allowed_in_flight = GetAllowedInFlightMediaDuration();
   if (VLOG_IS_ON(1)) {
     const int64_t percent =
         allowed_in_flight > base::TimeDelta()
             ? base::ClampRound<int64_t>(duration_would_be_in_flight /
                                         allowed_in_flight * 100)
             : std::numeric_limits<int64_t>::max();
     VLOG_IF(1, percent > 50)
         << SENDER_SSRC
         << duration_in_flight.InMicroseconds() << " usec in-flight + "
         << frame_duration.InMicroseconds() << " usec for next frame --> "
         << percent << "% of allowed in-flight.";
   }
   if (duration_would_be_in_flight > allowed_in_flight) {
     VLOG(1) << SENDER_SSRC << "Dropping: In-flight duration would be too high.";
     return true;
   }

   // Next frame is accepted.
   return false;
 }

 }  // namespace cast
 }  // namespace media
	// Copyright 2014 The Chromium Authors. All rights reserved.
	// Use of this source code is governed by a BSD-style license that can be
	// found in the LICENSE file.

	#include "media/cast/sender/frame_sender.h"

	#include <algorithm>
	#include <limits>
	#include <memory>
	#include <utility>
	#include <vector>

	#include "base/bind.h"
	#include "base/logging.h"
	#include "base/macros.h"
	#include "base/numerics/safe_conversions.h"
	#include "base/trace_event/trace_event.h"
	#include "media/cast/constants.h"
	#include "media/cast/sender/sender_encoded_frame.h"

	namespace media {
	namespace cast {
	namespace {

	constexpr int kNumAggressiveReportsSentAtStart = 100;
	constexpr base::TimeDelta kMinSchedulingDelay = base::Milliseconds(1);
	constexpr base::TimeDelta kReceiverProcessTime = base::Milliseconds(250);

	// The additional number of frames that can be in-flight when input exceeds the
	// maximum frame rate.
	constexpr int kMaxFrameBurst = 5;

	} // namespace

	// Convenience macro used in logging statements throughout this file.
	#define SENDER_SSRC (is_audio_ ? "AUDIO[" : "VIDEO[") << ssrc_ << "] "

	FrameSender::RtcpClient::RtcpClient(base::WeakPtr<FrameSender> frame_sender)
	: frame_sender_(frame_sender) {}

	FrameSender::RtcpClient::~RtcpClient() = default;

	void FrameSender::RtcpClient::OnReceivedCastMessage(
	const RtcpCastMessage& cast_message) {
	if (frame_sender_)
	frame_sender_->OnReceivedCastFeedback(cast_message);
	}

	void FrameSender::RtcpClient::OnReceivedRtt(base::TimeDelta round_trip_time) {
	if (frame_sender_)
	frame_sender_->OnMeasuredRoundTripTime(round_trip_time);
	}

	void FrameSender::RtcpClient::OnReceivedPli() {
	if (frame_sender_)
	frame_sender_->OnReceivedPli();
	}

	FrameSender::FrameSender(scoped_refptr<CastEnvironment> cast_environment,
	CastTransport* const transport_sender,
	const FrameSenderConfig& config,
	CongestionControl* congestion_control)
	: cast_environment_(cast_environment),
	transport_sender_(transport_sender),
	ssrc_(config.sender_ssrc),
	min_playout_delay_(config.min_playout_delay.is_zero()
	? config.max_playout_delay
	: config.min_playout_delay),
	max_playout_delay_(config.max_playout_delay),
	animated_playout_delay_(config.animated_playout_delay.is_zero()
	? config.max_playout_delay
	: config.animated_playout_delay),
	send_target_playout_delay_(false),
	max_frame_rate_(config.max_frame_rate),
	num_aggressive_rtcp_reports_sent_(0),
	duplicate_ack_counter_(0),
	congestion_control_(congestion_control),
	picture_lost_at_receiver_(false),
	rtp_timebase_(config.rtp_timebase),
	is_audio_(config.rtp_payload_type <= RtpPayloadType::AUDIO_LAST),
	max_ack_delay_(config.max_playout_delay) {
	DCHECK(transport_sender_);
	DCHECK_GT(rtp_timebase_, 0);
	DCHECK(congestion_control_);
	// We assume animated content to begin with since that is the common use
	// case today.
	VLOG(1) << SENDER_SSRC << "min latency "
	<< min_playout_delay_.InMilliseconds() << "max latency "
	<< max_playout_delay_.InMilliseconds() << "animated latency "
	<< animated_playout_delay_.InMilliseconds();
	SetTargetPlayoutDelay(animated_playout_delay_);

	CastTransportRtpConfig transport_config;
	transport_config.ssrc = config.sender_ssrc;
	transport_config.feedback_ssrc = config.receiver_ssrc;
	transport_config.rtp_payload_type = config.rtp_payload_type;
	transport_config.aes_key = config.aes_key;
	transport_config.aes_iv_mask = config.aes_iv_mask;

	transport_sender->InitializeStream(
	transport_config,
	std::make_unique<FrameSender::RtcpClient>(weak_factory_.GetWeakPtr()));
	}

	FrameSender::~FrameSender() = default;

	void FrameSender::ScheduleNextRtcpReport() {
	DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

	cast_environment_->PostDelayedTask(
	CastEnvironment::MAIN, FROM_HERE,
	base::BindOnce(&FrameSender::SendRtcpReport, weak_factory_.GetWeakPtr(),
	true),
	base::Milliseconds(kRtcpReportIntervalMs));
	}

	void FrameSender::SendRtcpReport(bool schedule_future_reports) {
	DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

	// Sanity-check: We should have sent at least the first frame by this point.
	DCHECK(!last_send_time_.is_null());

	// Create lip-sync info for the sender report. The last sent frame's
	// reference time and RTP timestamp are used to estimate an RTP timestamp in
	// terms of "now." Note that \|now\| is never likely to be precise to an exact
	// frame boundary; and so the computation here will result in a
	// \|now_as_rtp_timestamp\| value that is rarely equal to any one emitted by the
	// encoder.
	const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
	const base::TimeDelta time_delta =
	now - GetRecordedReferenceTime(last_sent_frame_id_);
	const RtpTimeDelta rtp_delta =
	RtpTimeDelta::FromTimeDelta(time_delta, rtp_timebase_);
	const RtpTimeTicks now_as_rtp_timestamp =
	GetRecordedRtpTimestamp(last_sent_frame_id_) + rtp_delta;
	transport_sender_->SendSenderReport(ssrc_, now, now_as_rtp_timestamp);

	if (schedule_future_reports)
	ScheduleNextRtcpReport();
	}

	void FrameSender::OnMeasuredRoundTripTime(base::TimeDelta round_trip_time) {
	DCHECK_GT(round_trip_time, base::TimeDelta());
	current_round_trip_time_ = round_trip_time;
	max_ack_delay_ = 2 * std::max(current_round_trip_time_, base::TimeDelta()) +
	kReceiverProcessTime;
	max_ack_delay_ = std::min(max_ack_delay_, target_playout_delay_);
	}

	void FrameSender::SetTargetPlayoutDelay(
	base::TimeDelta new_target_playout_delay) {
	if (send_target_playout_delay_ &&
	target_playout_delay_ == new_target_playout_delay) {
	return;
	}
	new_target_playout_delay = std::max(new_target_playout_delay,
	min_playout_delay_);
	new_target_playout_delay = std::min(new_target_playout_delay,
	max_playout_delay_);
	VLOG(2) << SENDER_SSRC << "Target playout delay changing from "
	<< target_playout_delay_.InMilliseconds() << " ms to "
	<< new_target_playout_delay.InMilliseconds() << " ms.";
	target_playout_delay_ = new_target_playout_delay;
	max_ack_delay_ = std::min(max_ack_delay_, target_playout_delay_);
	send_target_playout_delay_ = true;
	congestion_control_->UpdateTargetPlayoutDelay(target_playout_delay_);
	}

	void FrameSender::ResendCheck() {
	DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
	DCHECK(!last_send_time_.is_null());
	const base::TimeDelta time_since_last_send =
	cast_environment_->Clock()->NowTicks() - last_send_time_;
	if (time_since_last_send > max_ack_delay_) {
	if (latest_acked_frame_id_ == last_sent_frame_id_) {
	// Last frame acked, no point in doing anything
	} else {
	VLOG(1) << SENDER_SSRC << "ACK timeout; last acked frame: "
	<< latest_acked_frame_id_;
	ResendForKickstart();
	}
	}
	ScheduleNextResendCheck();
	}

	void FrameSender::ScheduleNextResendCheck() {
	DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
	DCHECK(!last_send_time_.is_null());
	base::TimeDelta time_to_next =
	last_send_time_ - cast_environment_->Clock()->NowTicks() + max_ack_delay_;
	time_to_next = std::max(time_to_next, kMinSchedulingDelay);
	cast_environment_->PostDelayedTask(
	CastEnvironment::MAIN, FROM_HERE,
	base::BindOnce(&FrameSender::ResendCheck, weak_factory_.GetWeakPtr()),
	time_to_next);
	}

	void FrameSender::ResendForKickstart() {
	DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
	DCHECK(!last_send_time_.is_null());
	VLOG(1) << SENDER_SSRC << "Resending last packet of frame "
	<< last_sent_frame_id_ << " to kick-start.";
	last_send_time_ = cast_environment_->Clock()->NowTicks();
	transport_sender_->ResendFrameForKickstart(ssrc_, last_sent_frame_id_);
	}

	void FrameSender::RecordLatestFrameTimestamps(FrameId frame_id,
	base::TimeTicks reference_time,
	RtpTimeTicks rtp_timestamp) {
	DCHECK(!reference_time.is_null());
	frame_reference_times_[frame_id.lower_8_bits()] = reference_time;
	frame_rtp_timestamps_[frame_id.lower_8_bits()] = rtp_timestamp;
	}

	base::TimeTicks FrameSender::GetRecordedReferenceTime(FrameId frame_id) const {
	return frame_reference_times_[frame_id.lower_8_bits()];
	}

	RtpTimeTicks FrameSender::GetRecordedRtpTimestamp(FrameId frame_id) const {
	return frame_rtp_timestamps_[frame_id.lower_8_bits()];
	}

	int FrameSender::GetUnacknowledgedFrameCount() const {
	if (last_send_time_.is_null())
	return 0;
	const int count = last_sent_frame_id_ - latest_acked_frame_id_;
	DCHECK_GE(count, 0);
	return count;
	}

	base::TimeDelta FrameSender::GetAllowedInFlightMediaDuration() const {
	// The total amount allowed in-flight media should equal the amount that fits
	// within the entire playout delay window, plus the amount of time it takes to
	// receive an ACK from the receiver.
	// TODO(miu): Research is needed, but there is likely a better formula.
	return target_playout_delay_ + (current_round_trip_time_ / 2);
	}

	void FrameSender::SendEncodedFrame(
	int requested_bitrate_before_encode,
	std::unique_ptr<SenderEncodedFrame> encoded_frame) {
	DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

	VLOG(2) << SENDER_SSRC << "About to send another frame: last_sent="
	<< last_sent_frame_id_ << ", latest_acked=" << latest_acked_frame_id_;

	const FrameId frame_id = encoded_frame->frame_id;
	const bool is_first_frame_to_be_sent = last_send_time_.is_null();

	if (picture_lost_at_receiver_ &&
	(encoded_frame->dependency == EncodedFrame::KEY)) {
	picture_lost_at_receiver_ = false;
	DCHECK(frame_id > latest_acked_frame_id_);
	// Cancel sending remaining frames.
	std::vector<FrameId> cancel_sending_frames;
	for (FrameId id = latest_acked_frame_id_ + 1; id < frame_id; ++id) {
	cancel_sending_frames.push_back(id);
	}
	transport_sender_->CancelSendingFrames(ssrc_, cancel_sending_frames);
	OnCancelSendingFrames();
	}

	last_send_time_ = cast_environment_->Clock()->NowTicks();
	last_sent_frame_id_ = frame_id;
	// If this is the first frame about to be sent, fake the value of
	// \|latest_acked_frame_id_\| to indicate the receiver starts out all caught up.
	// Also, schedule the periodic frame re-send checks.
	if (is_first_frame_to_be_sent) {
	latest_acked_frame_id_ = frame_id - 1;
	ScheduleNextResendCheck();
	}

	VLOG_IF(1, !is_audio_ && encoded_frame->dependency == EncodedFrame::KEY)
	<< SENDER_SSRC << "Sending encoded key frame, id=" << frame_id;

	std::unique_ptr<FrameEvent> encode_event(new FrameEvent());
	encode_event->timestamp = encoded_frame->encode_completion_time;
	encode_event->type = FRAME_ENCODED;
	encode_event->media_type = is_audio_ ? AUDIO_EVENT : VIDEO_EVENT;
	encode_event->rtp_timestamp = encoded_frame->rtp_timestamp;
	encode_event->frame_id = frame_id;
	encode_event->size = base::checked_cast<uint32_t>(encoded_frame->data.size());
	encode_event->key_frame = encoded_frame->dependency == EncodedFrame::KEY;
	encode_event->target_bitrate = requested_bitrate_before_encode;
	encode_event->encoder_cpu_utilization = encoded_frame->encoder_utilization;
	encode_event->idealized_bitrate_utilization =
	encoded_frame->lossy_utilization;
	cast_environment_->logger()->DispatchFrameEvent(std::move(encode_event));

	RecordLatestFrameTimestamps(frame_id,
	encoded_frame->reference_time,
	encoded_frame->rtp_timestamp);

	if (!is_audio_) {
	// Used by chrome/browser/media/cast_mirroring_performance_browsertest.cc
	TRACE_EVENT_INSTANT1(
	"cast_perf_test", "VideoFrameEncoded",
	TRACE_EVENT_SCOPE_THREAD,
	"rtp_timestamp", encoded_frame->rtp_timestamp.lower_32_bits());
	}

	// At the start of the session, it's important to send reports before each
	// frame so that the receiver can properly compute playout times. The reason
	// more than one report is sent is because transmission is not guaranteed,
	// only best effort, so send enough that one should almost certainly get
	// through.
	if (num_aggressive_rtcp_reports_sent_ < kNumAggressiveReportsSentAtStart) {
	// SendRtcpReport() will schedule future reports to be made if this is the
	// last "aggressive report."
	++num_aggressive_rtcp_reports_sent_;
	const bool is_last_aggressive_report =
	(num_aggressive_rtcp_reports_sent_ == kNumAggressiveReportsSentAtStart);
	VLOG_IF(1, is_last_aggressive_report)
	<< SENDER_SSRC << "Sending last aggressive report.";
	SendRtcpReport(is_last_aggressive_report);
	}

	congestion_control_->SendFrameToTransport(
	frame_id, encoded_frame->data.size() * 8, last_send_time_);

	if (send_target_playout_delay_) {
	encoded_frame->new_playout_delay_ms =
	target_playout_delay_.InMilliseconds();
	}

	const char* name = is_audio_ ? "Audio Transport" : "Video Transport";
	TRACE_EVENT_NESTABLE_ASYNC_BEGIN1(
	"cast.stream", name, TRACE_ID_WITH_SCOPE(name, frame_id.lower_32_bits()),
	"rtp_timestamp", encoded_frame->rtp_timestamp.lower_32_bits());
	transport_sender_->InsertFrame(ssrc_, *encoded_frame);
	}

	void FrameSender::OnCancelSendingFrames() {}

	void FrameSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
	DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

	const bool have_valid_rtt = current_round_trip_time_ > base::TimeDelta();
	if (have_valid_rtt) {
	congestion_control_->UpdateRtt(current_round_trip_time_);

	// Having the RTT value implies the receiver sent back a receiver report
	// based on it having received a report from here. Therefore, ensure this
	// sender stops aggressively sending reports.
	if (num_aggressive_rtcp_reports_sent_ < kNumAggressiveReportsSentAtStart) {
	VLOG(1) << SENDER_SSRC
	<< "No longer a need to send reports aggressively (sent "
	<< num_aggressive_rtcp_reports_sent_ << ").";
	num_aggressive_rtcp_reports_sent_ = kNumAggressiveReportsSentAtStart;
	ScheduleNextRtcpReport();
	}
	}

	if (last_send_time_.is_null())
	return; // Cannot get an ACK without having first sent a frame.

	if (cast_feedback.missing_frames_and_packets.empty() &&
	cast_feedback.received_later_frames.empty()) {
	if (latest_acked_frame_id_ == cast_feedback.ack_frame_id) {
	VLOG(1) << SENDER_SSRC << "Received duplicate ACK for frame "
	<< latest_acked_frame_id_;
	TRACE_EVENT_INSTANT2(
	"cast.stream", "Duplicate ACK", TRACE_EVENT_SCOPE_THREAD,
	"ack_frame_id", cast_feedback.ack_frame_id.lower_32_bits(),
	"last_sent_frame_id", last_sent_frame_id_.lower_32_bits());
	}
	// We only count duplicate ACKs when we have sent newer frames.
	if (latest_acked_frame_id_ == cast_feedback.ack_frame_id &&
	latest_acked_frame_id_ != last_sent_frame_id_) {
	duplicate_ack_counter_++;
	} else {
	duplicate_ack_counter_ = 0;
	}
	// TODO(miu): The values "2" and "3" should be derived from configuration.
	if (duplicate_ack_counter_ >= 2 && duplicate_ack_counter_ % 3 == 2) {
	ResendForKickstart();
	}
	} else {
	// Only count duplicated ACKs if there is no NACK request in between.
	// This is to avoid aggresive resend.
	duplicate_ack_counter_ = 0;
	}

	base::TimeTicks now = cast_environment_->Clock()->NowTicks();
	congestion_control_->AckFrame(cast_feedback.ack_frame_id, now);
	if (!cast_feedback.received_later_frames.empty()) {
	// Ack the received frames.
	congestion_control_->AckLaterFrames(cast_feedback.received_later_frames,
	now);
	}

	std::unique_ptr<FrameEvent> ack_event(new FrameEvent());
	ack_event->timestamp = now;
	ack_event->type = FRAME_ACK_RECEIVED;
	ack_event->media_type = is_audio_ ? AUDIO_EVENT : VIDEO_EVENT;
	ack_event->rtp_timestamp =
	GetRecordedRtpTimestamp(cast_feedback.ack_frame_id);
	ack_event->frame_id = cast_feedback.ack_frame_id;
	cast_environment_->logger()->DispatchFrameEvent(std::move(ack_event));

	const bool is_acked_out_of_order =
	cast_feedback.ack_frame_id < latest_acked_frame_id_;
	VLOG(2) << SENDER_SSRC
	<< "Received ACK" << (is_acked_out_of_order ? " out-of-order" : "")
	<< " for frame " << cast_feedback.ack_frame_id;
	if (is_acked_out_of_order) {
	TRACE_EVENT_INSTANT2(
	"cast.stream", "ACK out of order", TRACE_EVENT_SCOPE_THREAD,
	"ack_frame_id", cast_feedback.ack_frame_id.lower_32_bits(),
	"latest_acked_frame_id", latest_acked_frame_id_.lower_32_bits());
	} else if (latest_acked_frame_id_ < cast_feedback.ack_frame_id) {
	// Cancel resends of acked frames.
	std::vector<FrameId> frames_to_cancel;
	frames_to_cancel.reserve(cast_feedback.ack_frame_id -
	latest_acked_frame_id_);
	do {
	++latest_acked_frame_id_;
	frames_to_cancel.push_back(latest_acked_frame_id_);
	// This is a good place to match the trace for frame ids
	// since this ensures we not only track frame ids that are
	// implicitly ACKed, but also handles duplicate ACKs
	const char* name = is_audio_ ? "Audio Transport" : "Video Transport";
	TRACE_EVENT_NESTABLE_ASYNC_END1(
	"cast.stream", name,
	TRACE_ID_WITH_SCOPE(name, latest_acked_frame_id_.lower_32_bits()),
	"RTT_usecs", current_round_trip_time_.InMicroseconds());
	} while (latest_acked_frame_id_ < cast_feedback.ack_frame_id);
	transport_sender_->CancelSendingFrames(ssrc_, frames_to_cancel);
	OnCancelSendingFrames();
	}
	}

	void FrameSender::OnReceivedPli() {
	picture_lost_at_receiver_ = true;
	}

	bool FrameSender::ShouldDropNextFrame(base::TimeDelta frame_duration) const {
	// Check that accepting the next frame won't cause more frames to become
	// in-flight than the system's design limit.
	const int count_frames_in_flight =
	GetUnacknowledgedFrameCount() + GetNumberOfFramesInEncoder();
	if (count_frames_in_flight >= kMaxUnackedFrames) {
	VLOG(1) << SENDER_SSRC << "Dropping: Too many frames would be in-flight.";
	return true;
	}

	// Check that accepting the next frame won't exceed the configured maximum
	// frame rate, allowing for short-term bursts.
	base::TimeDelta duration_in_flight = GetInFlightMediaDuration();
	const double max_frames_in_flight =
	max_frame_rate_ * duration_in_flight.InSecondsF();
	if (count_frames_in_flight >= max_frames_in_flight + kMaxFrameBurst) {
	VLOG(1) << SENDER_SSRC << "Dropping: Burst threshold would be exceeded.";
	return true;
	}

	// Check that accepting the next frame won't exceed the allowed in-flight
	// media duration.
	const base::TimeDelta duration_would_be_in_flight =
	duration_in_flight + frame_duration;
	const base::TimeDelta allowed_in_flight = GetAllowedInFlightMediaDuration();
	if (VLOG_IS_ON(1)) {
	const int64_t percent =
	allowed_in_flight > base::TimeDelta()
	? base::ClampRound<int64_t>(duration_would_be_in_flight /
	allowed_in_flight * 100)
	: std::numeric_limits<int64_t>::max();
	VLOG_IF(1, percent > 50)
	<< SENDER_SSRC
	<< duration_in_flight.InMicroseconds() << " usec in-flight + "
	<< frame_duration.InMicroseconds() << " usec for next frame --> "
	<< percent << "% of allowed in-flight.";
	}
	if (duration_would_be_in_flight > allowed_in_flight) {
	VLOG(1) << SENDER_SSRC << "Dropping: In-flight duration would be too high.";
	return true;
	}

	// Next frame is accepted.
	return false;
	}

	} // namespace cast
	} // namespace media