blob: cf050b7f10c5fae99d16a097129898f59bed2605 [file] [log] [blame]
// Copyright 2013 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/video_sender/video_sender.h"
#include <algorithm>
#include <cstring>
#include "base/bind.h"
#include "base/debug/trace_event.h"
#include "base/logging.h"
#include "base/message_loop/message_loop.h"
#include "media/cast/cast_defines.h"
#include "media/cast/rtcp/rtcp_defines.h"
#include "media/cast/transport/cast_transport_config.h"
#include "media/cast/video_sender/external_video_encoder.h"
#include "media/cast/video_sender/video_encoder_impl.h"
namespace media {
namespace cast {
const int kNumAggressiveReportsSentAtStart = 100;
const int kMinSchedulingDelayMs = 1;
VideoSender::VideoSender(
scoped_refptr<CastEnvironment> cast_environment,
const VideoSenderConfig& video_config,
const CreateVideoEncodeAcceleratorCallback& create_vea_cb,
const CreateVideoEncodeMemoryCallback& create_video_encode_mem_cb,
transport::CastTransportSender* const transport_sender)
: cast_environment_(cast_environment),
target_playout_delay_(base::TimeDelta::FromMilliseconds(
video_config.rtp_config.max_delay_ms)),
transport_sender_(transport_sender),
max_unacked_frames_(
std::min(kMaxUnackedFrames,
1 + static_cast<int>(target_playout_delay_ *
video_config.max_frame_rate /
base::TimeDelta::FromSeconds(1)))),
rtcp_(cast_environment_,
this,
transport_sender_,
NULL, // paced sender.
NULL,
video_config.rtcp_mode,
base::TimeDelta::FromMilliseconds(video_config.rtcp_interval),
video_config.rtp_config.ssrc,
video_config.incoming_feedback_ssrc,
video_config.rtcp_c_name,
VIDEO_EVENT),
rtp_timestamp_helper_(kVideoFrequency),
num_aggressive_rtcp_reports_sent_(0),
frames_in_encoder_(0),
last_sent_frame_id_(0),
latest_acked_frame_id_(0),
duplicate_ack_counter_(0),
congestion_control_(cast_environment->Clock(),
video_config.max_bitrate,
video_config.min_bitrate,
max_unacked_frames_),
cast_initialization_status_(STATUS_VIDEO_UNINITIALIZED),
weak_factory_(this) {
VLOG(1) << "max_unacked_frames " << max_unacked_frames_;
DCHECK_GT(max_unacked_frames_, 0);
if (video_config.use_external_encoder) {
video_encoder_.reset(new ExternalVideoEncoder(cast_environment,
video_config,
create_vea_cb,
create_video_encode_mem_cb));
} else {
video_encoder_.reset(new VideoEncoderImpl(
cast_environment, video_config, max_unacked_frames_));
}
cast_initialization_status_ = STATUS_VIDEO_INITIALIZED;
media::cast::transport::CastTransportVideoConfig transport_config;
transport_config.codec = video_config.codec;
transport_config.rtp.config = video_config.rtp_config;
transport_config.rtp.max_outstanding_frames = max_unacked_frames_;
transport_sender_->InitializeVideo(transport_config);
rtcp_.SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
memset(frame_id_to_rtp_timestamp_, 0, sizeof(frame_id_to_rtp_timestamp_));
}
VideoSender::~VideoSender() {
}
void VideoSender::InsertRawVideoFrame(
const scoped_refptr<media::VideoFrame>& video_frame,
const base::TimeTicks& capture_time) {
DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
if (cast_initialization_status_ != STATUS_VIDEO_INITIALIZED) {
NOTREACHED();
return;
}
DCHECK(video_encoder_.get()) << "Invalid state";
RtpTimestamp rtp_timestamp = GetVideoRtpTimestamp(capture_time);
cast_environment_->Logging()->InsertFrameEvent(
capture_time, FRAME_CAPTURE_BEGIN, VIDEO_EVENT,
rtp_timestamp, kFrameIdUnknown);
cast_environment_->Logging()->InsertFrameEvent(
cast_environment_->Clock()->NowTicks(),
FRAME_CAPTURE_END, VIDEO_EVENT,
rtp_timestamp,
kFrameIdUnknown);
// Used by chrome/browser/extension/api/cast_streaming/performance_test.cc
TRACE_EVENT_INSTANT2(
"cast_perf_test", "InsertRawVideoFrame",
TRACE_EVENT_SCOPE_THREAD,
"timestamp", capture_time.ToInternalValue(),
"rtp_timestamp", rtp_timestamp);
if (AreTooManyFramesInFlight()) {
VLOG(1) << "Dropping frame due to too many frames currently in-flight.";
return;
}
uint32 bitrate = congestion_control_.GetBitrate(
capture_time + target_playout_delay_, target_playout_delay_);
video_encoder_->SetBitRate(bitrate);
if (video_encoder_->EncodeVideoFrame(
video_frame,
capture_time,
base::Bind(&VideoSender::SendEncodedVideoFrame,
weak_factory_.GetWeakPtr(),
bitrate))) {
frames_in_encoder_++;
} else {
VLOG(1) << "Encoder rejected a frame. Skipping...";
}
}
void VideoSender::SendEncodedVideoFrame(
int requested_bitrate_before_encode,
scoped_ptr<transport::EncodedFrame> encoded_frame) {
DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
DCHECK_GT(frames_in_encoder_, 0);
frames_in_encoder_--;
const uint32 frame_id = encoded_frame->frame_id;
const bool is_first_frame_to_be_sent = last_send_time_.is_null();
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, encoded_frame->dependency == transport::EncodedFrame::KEY)
<< "Send encoded key frame; frame_id: " << frame_id;
cast_environment_->Logging()->InsertEncodedFrameEvent(
last_send_time_, FRAME_ENCODED, VIDEO_EVENT, encoded_frame->rtp_timestamp,
frame_id, static_cast<int>(encoded_frame->data.size()),
encoded_frame->dependency == transport::EncodedFrame::KEY,
requested_bitrate_before_encode);
// Only use lowest 8 bits as key.
frame_id_to_rtp_timestamp_[frame_id & 0xff] = encoded_frame->rtp_timestamp;
// Used by chrome/browser/extension/api/cast_streaming/performance_test.cc
TRACE_EVENT_INSTANT1(
"cast_perf_test", "VideoFrameEncoded",
TRACE_EVENT_SCOPE_THREAD,
"rtp_timestamp", encoded_frame->rtp_timestamp);
DCHECK(!encoded_frame->reference_time.is_null());
rtp_timestamp_helper_.StoreLatestTime(encoded_frame->reference_time,
encoded_frame->rtp_timestamp);
// 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) << "Sending last aggressive report.";
SendRtcpReport(is_last_aggressive_report);
}
congestion_control_.SendFrameToTransport(
frame_id, encoded_frame->data.size() * 8, last_send_time_);
transport_sender_->InsertCodedVideoFrame(*encoded_frame);
}
void VideoSender::IncomingRtcpPacket(scoped_ptr<Packet> packet) {
DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
rtcp_.IncomingRtcpPacket(&packet->front(), packet->size());
}
void VideoSender::ScheduleNextRtcpReport() {
DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
base::TimeDelta time_to_next = rtcp_.TimeToSendNextRtcpReport() -
cast_environment_->Clock()->NowTicks();
time_to_next = std::max(
time_to_next, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
cast_environment_->PostDelayedTask(
CastEnvironment::MAIN,
FROM_HERE,
base::Bind(&VideoSender::SendRtcpReport,
weak_factory_.GetWeakPtr(),
true),
time_to_next);
}
void VideoSender::SendRtcpReport(bool schedule_future_reports) {
DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
uint32 now_as_rtp_timestamp = 0;
if (rtp_timestamp_helper_.GetCurrentTimeAsRtpTimestamp(
now, &now_as_rtp_timestamp)) {
rtcp_.SendRtcpFromRtpSender(now, now_as_rtp_timestamp);
} else {
// |rtp_timestamp_helper_| should have stored a mapping by this point.
NOTREACHED();
}
if (schedule_future_reports)
ScheduleNextRtcpReport();
}
void VideoSender::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() +
target_playout_delay_;
time_to_next = std::max(
time_to_next, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
cast_environment_->PostDelayedTask(
CastEnvironment::MAIN,
FROM_HERE,
base::Bind(&VideoSender::ResendCheck, weak_factory_.GetWeakPtr()),
time_to_next);
}
void VideoSender::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 > target_playout_delay_) {
if (latest_acked_frame_id_ == last_sent_frame_id_) {
// Last frame acked, no point in doing anything
} else {
VLOG(1) << "ACK timeout; last acked frame: " << latest_acked_frame_id_;
ResendForKickstart();
}
}
ScheduleNextResendCheck();
}
void VideoSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
base::TimeDelta rtt;
base::TimeDelta avg_rtt;
base::TimeDelta min_rtt;
base::TimeDelta max_rtt;
if (rtcp_.Rtt(&rtt, &avg_rtt, &min_rtt, &max_rtt)) {
congestion_control_.UpdateRtt(rtt);
// Don't use a RTT lower than our average.
rtt = std::max(rtt, avg_rtt);
// Having the RTT values 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) << "No longer a need to send reports aggressively (sent "
<< num_aggressive_rtcp_reports_sent_ << ").";
num_aggressive_rtcp_reports_sent_ = kNumAggressiveReportsSentAtStart;
ScheduleNextRtcpReport();
}
} else {
// We have no measured value use default.
rtt = base::TimeDelta::FromMilliseconds(kStartRttMs);
}
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()) {
video_encoder_->LatestFrameIdToReference(cast_feedback.ack_frame_id_);
// 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) {
VLOG(1) << "Received duplicate ACK for frame " << latest_acked_frame_id_;
ResendForKickstart();
}
} else {
// Only count duplicated ACKs if there is no NACK request in between.
// This is to avoid aggresive resend.
duplicate_ack_counter_ = 0;
// A NACK is also used to cancel pending re-transmissions.
transport_sender_->ResendPackets(
false, cast_feedback.missing_frames_and_packets_, true, rtt);
}
base::TimeTicks now = cast_environment_->Clock()->NowTicks();
congestion_control_.AckFrame(cast_feedback.ack_frame_id_, now);
RtpTimestamp rtp_timestamp =
frame_id_to_rtp_timestamp_[cast_feedback.ack_frame_id_ & 0xff];
cast_environment_->Logging()->InsertFrameEvent(now,
FRAME_ACK_RECEIVED,
VIDEO_EVENT,
rtp_timestamp,
cast_feedback.ack_frame_id_);
const bool is_acked_out_of_order =
static_cast<int32>(cast_feedback.ack_frame_id_ -
latest_acked_frame_id_) < 0;
VLOG(2) << "Received ACK" << (is_acked_out_of_order ? " out-of-order" : "")
<< " for frame " << cast_feedback.ack_frame_id_;
if (!is_acked_out_of_order) {
// Cancel resends of acked frames.
MissingFramesAndPacketsMap missing_frames_and_packets;
PacketIdSet missing;
while (latest_acked_frame_id_ != cast_feedback.ack_frame_id_) {
latest_acked_frame_id_++;
missing_frames_and_packets[latest_acked_frame_id_] = missing;
}
transport_sender_->ResendPackets(
false, missing_frames_and_packets, true, rtt);
latest_acked_frame_id_ = cast_feedback.ack_frame_id_;
}
}
bool VideoSender::AreTooManyFramesInFlight() const {
DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
int frames_in_flight = frames_in_encoder_;
if (!last_send_time_.is_null()) {
frames_in_flight +=
static_cast<int32>(last_sent_frame_id_ - latest_acked_frame_id_);
}
VLOG(2) << frames_in_flight
<< " frames in flight; last sent: " << last_sent_frame_id_
<< " latest acked: " << latest_acked_frame_id_
<< " frames in encoder: " << frames_in_encoder_;
return frames_in_flight >= max_unacked_frames_;
}
void VideoSender::ResendForKickstart() {
DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
DCHECK(!last_send_time_.is_null());
VLOG(1) << "Resending last packet of frame " << last_sent_frame_id_
<< " to kick-start.";
// Send the first packet of the last encoded frame to kick start
// retransmission. This gives enough information to the receiver what
// packets and frames are missing.
MissingFramesAndPacketsMap missing_frames_and_packets;
PacketIdSet missing;
missing.insert(kRtcpCastLastPacket);
missing_frames_and_packets.insert(
std::make_pair(last_sent_frame_id_, missing));
last_send_time_ = cast_environment_->Clock()->NowTicks();
base::TimeDelta rtt;
base::TimeDelta avg_rtt;
base::TimeDelta min_rtt;
base::TimeDelta max_rtt;
rtcp_.Rtt(&rtt, &avg_rtt, &min_rtt, &max_rtt);
// Sending this extra packet is to kick-start the session. There is
// no need to optimize re-transmission for this case.
transport_sender_->ResendPackets(false, missing_frames_and_packets,
false, rtt);
}
} // namespace cast
} // namespace media