Cast: Reshuffle files under media/cast

media/cast/audio_sender/audio_sender.cc

-// 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/audio_sender/audio_sender.h"
-
-#include "base/bind.h"
-#include "base/logging.h"
-#include "base/message_loop/message_loop.h"
-#include "media/cast/audio_sender/audio_encoder.h"
-#include "media/cast/cast_defines.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/transport/cast_transport_config.h"
-
-namespace media {
-namespace cast {
-namespace {
-
-const int kNumAggressiveReportsSentAtStart = 100;
-const int kMinSchedulingDelayMs = 1;
-
-// TODO(miu): This should be specified in AudioSenderConfig, but currently it is
-// fixed to 100 FPS (i.e., 10 ms per frame), and AudioEncoder assumes this as
-// well.
-const int kAudioFrameRate = 100;
-
-// Helper function to compute the maximum unacked audio frames that is sent.
-int GetMaxUnackedFrames(base::TimeDelta target_delay) {
-  // As long as it doesn't go over |kMaxUnackedFrames|, it is okay to send more
-  // audio data than the target delay would suggest. Audio packets are tiny and
-  // receiver has the ability to drop any one of the packets.
-  // We send up to three times of the target delay of audio frames.
-  int frames =
-      1 + 3 * target_delay * kAudioFrameRate / base::TimeDelta::FromSeconds(1);
-  return std::min(kMaxUnackedFrames, frames);
-}
-}  // namespace
-
-AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
-                         const AudioSenderConfig& audio_config,
-                         transport::CastTransportSender* const transport_sender)
-    : cast_environment_(cast_environment),
-      target_playout_delay_(audio_config.target_playout_delay),
-      transport_sender_(transport_sender),
-      max_unacked_frames_(GetMaxUnackedFrames(target_playout_delay_)),
-      configured_encoder_bitrate_(audio_config.bitrate),
-      rtcp_(cast_environment,
-            this,
-            transport_sender_,
-            NULL,  // paced sender.
-            NULL,
-            audio_config.rtcp_mode,
-            base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval),
-            audio_config.ssrc,
-            audio_config.incoming_feedback_ssrc,
-            audio_config.rtcp_c_name,
-            AUDIO_EVENT),
-      rtp_timestamp_helper_(audio_config.frequency),
-      num_aggressive_rtcp_reports_sent_(0),
-      last_sent_frame_id_(0),
-      latest_acked_frame_id_(0),
-      duplicate_ack_counter_(0),
-      cast_initialization_status_(STATUS_AUDIO_UNINITIALIZED),
-      weak_factory_(this) {
-  VLOG(1) << "max_unacked_frames " << max_unacked_frames_;
-  DCHECK_GT(max_unacked_frames_, 0);
-
-  if (!audio_config.use_external_encoder) {
-    audio_encoder_.reset(
-        new AudioEncoder(cast_environment,
-                         audio_config.channels,
-                         audio_config.frequency,
-                         audio_config.bitrate,
-                         audio_config.codec,
-                         base::Bind(&AudioSender::SendEncodedAudioFrame,
-                                    weak_factory_.GetWeakPtr())));
-    cast_initialization_status_ = audio_encoder_->InitializationResult();
-  } else {
-    NOTREACHED();  // No support for external audio encoding.
-    cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED;
-  }
-
-  media::cast::transport::CastTransportRtpConfig transport_config;
-  transport_config.ssrc = audio_config.ssrc;
-  transport_config.rtp_payload_type = audio_config.rtp_payload_type;
-  // TODO(miu): AudioSender needs to be like VideoSender in providing an upper
-  // limit on the number of in-flight frames.
-  transport_config.stored_frames = max_unacked_frames_;
-  transport_config.aes_key = audio_config.aes_key;
-  transport_config.aes_iv_mask = audio_config.aes_iv_mask;
-  transport_sender_->InitializeAudio(transport_config);
-
-  rtcp_.SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
-
-  memset(frame_id_to_rtp_timestamp_, 0, sizeof(frame_id_to_rtp_timestamp_));
-}
-
-AudioSender::~AudioSender() {}
-
-void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus,
-                              const base::TimeTicks& recorded_time) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  if (cast_initialization_status_ != STATUS_AUDIO_INITIALIZED) {
-    NOTREACHED();
-    return;
-  }
-  DCHECK(audio_encoder_.get()) << "Invalid internal state";
-
-  if (AreTooManyFramesInFlight()) {
-    VLOG(1) << "Dropping frame due to too many frames currently in-flight.";
-    return;
-  }
-
-  audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time);
-}
-
-void AudioSender::SendEncodedAudioFrame(
-    scoped_ptr<transport::EncodedFrame> encoded_frame) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-
-  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();
-  }
-
-  cast_environment_->Logging()->InsertEncodedFrameEvent(
-      last_send_time_, FRAME_ENCODED, AUDIO_EVENT, encoded_frame->rtp_timestamp,
-      frame_id, static_cast<int>(encoded_frame->data.size()),
-      encoded_frame->dependency == transport::EncodedFrame::KEY,
-      configured_encoder_bitrate_);
-  // Only use lowest 8 bits as key.
-  frame_id_to_rtp_timestamp_[frame_id & 0xff] = 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 we 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);
-  }
-
-  transport_sender_->InsertCodedAudioFrame(*encoded_frame);
-}
-
-void AudioSender::IncomingRtcpPacket(scoped_ptr<Packet> packet) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  rtcp_.IncomingRtcpPacket(&packet->front(), packet->size());
-}
-
-void AudioSender::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(&AudioSender::SendRtcpReport,
-                 weak_factory_.GetWeakPtr(),
-                 true),
-      time_to_next);
-}
-
-void AudioSender::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 AudioSender::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(&AudioSender::ResendCheck, weak_factory_.GetWeakPtr()),
-      time_to_next);
-}
-
-void AudioSender::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 AudioSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-
-  if (rtcp_.is_rtt_available()) {
-    // 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();
-    }
-  }
-
-  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()) {
-    // 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;
-
-    base::TimeDelta rtt;
-    base::TimeDelta avg_rtt;
-    base::TimeDelta min_rtt;
-    base::TimeDelta max_rtt;
-    rtcp_.Rtt(&rtt, &avg_rtt, &min_rtt, &max_rtt);
-
-    // A NACK is also used to cancel pending re-transmissions.
-    transport_sender_->ResendPackets(
-        true, cast_feedback.missing_frames_and_packets_, false, min_rtt);
-  }
-
-  const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
-
-  const RtpTimestamp rtp_timestamp =
-      frame_id_to_rtp_timestamp_[cast_feedback.ack_frame_id_ & 0xff];
-  cast_environment_->Logging()->InsertFrameEvent(now,
-                                                 FRAME_ACK_RECEIVED,
-                                                 AUDIO_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(
-        true, missing_frames_and_packets, true, base::TimeDelta());
-    latest_acked_frame_id_ = cast_feedback.ack_frame_id_;
-  }
-}
-
-bool AudioSender::AreTooManyFramesInFlight() const {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  int frames_in_flight = 0;
-  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_;
-  return frames_in_flight >= max_unacked_frames_;
-}
-
-void AudioSender::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(
-      true, missing_frames_and_packets, false, min_rtt);
-}
-
-}  // namespace cast
-}  // namespace media