Cast: Refactor RTCP handling

media/cast/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/sender/audio_sender.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/net/cast_transport_config.h"
 #include "media/cast/net/rtcp/rtcp_defines.h"

[miu, 2014/07/16 19:58:03]

Remove this include.

[Alpha Left Google, 2014/07/17 01:01:46]

Done.

 #include "media/cast/sender/audio_encoder.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,
                          CastTransportSender* const transport_sender)
-    : cast_environment_(cast_environment),
+    : FrameSender(
+        cast_environment,
+        transport_sender,
+        base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval),
+        audio_config.frequency,
+        true),
       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::CastTransportRtpConfig transport_config;
   transport_config.ssrc = audio_config.ssrc;
+  transport_config.feedback_ssrc = audio_config.incoming_feedback_ssrc;
+  transport_config.c_name = audio_config.rtcp_c_name;
   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);
-
+  transport_sender->InitializeAudio(
+      transport_config,
+      base::Bind(&AudioSender::OnReceivedCastFeedback,
+                 weak_factory_.GetWeakPtr()),
+      base::Bind(&AudioSender::OnReceivedRtt, weak_factory_.GetWeakPtr()));
   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();
@@ skipping 49 matching lines @@
     ++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,
@@ skipping 14 matching lines @@
       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()) {
+  if (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();
     }
   }
@@ skipping 12 matching lines @@
     // 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);
+        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,
@@ skipping 39 matching lines @@
   // 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);
+      true, missing_frames_and_packets, false, min_rtt_);
 }
 
 }  // namespace cast
 }  // namespace media