| // 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/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" |
| #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), |
| 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.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<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 == 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 |
