Index: media/cast/sender/audio_sender.cc |
diff --git a/media/cast/sender/audio_sender.cc b/media/cast/sender/audio_sender.cc |
index a36e6de882ccdf64c0b732927fda7eabadc764dd..0a458911358c5af323ff1320be4e81deed858665 100644 |
--- a/media/cast/sender/audio_sender.cc |
+++ b/media/cast/sender/audio_sender.cc |
@@ -15,8 +15,6 @@ namespace media { |
namespace cast { |
namespace { |
-const int kNumAggressiveReportsSentAtStart = 100; |
- |
// 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. |
@@ -29,13 +27,15 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment, |
CastTransportSender* const transport_sender) |
: FrameSender( |
cast_environment, |
+ true, |
transport_sender, |
base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval), |
audio_config.frequency, |
audio_config.ssrc, |
kAudioFrameRate * 2.0, // We lie to increase max outstanding frames. |
- audio_config.target_playout_delay), |
- configured_encoder_bitrate_(audio_config.bitrate), |
+ audio_config.target_playout_delay, |
+ NewFixedCongestionControl(audio_config.bitrate)), |
+ samples_sent_to_encoder_(0), |
weak_factory_(this) { |
cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED; |
VLOG(1) << "max_unacked_frames " << max_unacked_frames_; |
@@ -48,8 +48,9 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment, |
audio_config.frequency, |
audio_config.bitrate, |
audio_config.codec, |
- base::Bind(&AudioSender::SendEncodedAudioFrame, |
- weak_factory_.GetWeakPtr()))); |
+ base::Bind(&FrameSender::SendEncodedFrame, |
+ weak_factory_.GetWeakPtr(), |
+ audio_config.bitrate))); |
cast_initialization_status_ = audio_encoder_->InitializationResult(); |
} else { |
NOTREACHED(); // No support for external audio encoding. |
@@ -89,142 +90,17 @@ void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus, |
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_); |
- |
- RecordLatestFrameTimestamps(frame_id, |
- 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); |
- } |
- |
- if (send_target_playout_delay_) { |
- encoded_frame->new_playout_delay_ms = |
- target_playout_delay_.InMilliseconds(); |
- } |
- transport_sender_->InsertFrame(ssrc_, *encoded_frame); |
-} |
+ int64 old_frames_sent = |
+ samples_sent_to_encoder_ * kAudioFrameRate / rtp_timebase_; |
+ samples_sent_to_encoder_ += audio_bus->frames(); |
+ int64 new_frames_sent = |
+ samples_sent_to_encoder_ * kAudioFrameRate / rtp_timebase_; |
+ frames_in_encoder_ += new_frames_sent - old_frames_sent; |
-void AudioSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) { |
- DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
- |
- 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(); |
- } |
- } |
- |
- 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; |
- } |
- |
- cast_environment_->Logging()->InsertFrameEvent( |
- cast_environment_->Clock()->NowTicks(), |
- FRAME_ACK_RECEIVED, |
- AUDIO_EVENT, |
- GetRecordedRtpTimestamp(cast_feedback.ack_frame_id), |
- 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. |
- std::vector<uint32> cancel_sending_frames; |
- while (latest_acked_frame_id_ != cast_feedback.ack_frame_id) { |
- latest_acked_frame_id_++; |
- cancel_sending_frames.push_back(latest_acked_frame_id_); |
- } |
- transport_sender_->CancelSendingFrames(ssrc_, cancel_sending_frames); |
- latest_acked_frame_id_ = cast_feedback.ack_frame_id; |
- } |
+ audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time); |
} |
-bool AudioSender::ShouldDropNextFrame(base::TimeTicks capture_time) const { |
- DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
- int frames_in_flight = 0; |
- base::TimeDelta duration_in_flight; |
- if (!last_send_time_.is_null()) { |
- frames_in_flight = |
- static_cast<int32>(last_sent_frame_id_ - latest_acked_frame_id_); |
- if (frames_in_flight > 0) { |
- const uint32 oldest_unacked_frame_id = latest_acked_frame_id_ + 1; |
- duration_in_flight = |
- capture_time - GetRecordedReferenceTime(oldest_unacked_frame_id); |
- } |
- } |
- VLOG(2) << frames_in_flight |
- << " frames in flight; last sent: " << last_sent_frame_id_ |
- << "; latest acked: " << latest_acked_frame_id_ |
- << "; duration in flight: " |
- << duration_in_flight.InMicroseconds() << " usec (" |
- << (target_playout_delay_ > base::TimeDelta() ? |
- 100 * duration_in_flight / target_playout_delay_ : |
- kint64max) << "%)"; |
- return frames_in_flight >= max_unacked_frames_ || |
- duration_in_flight >= target_playout_delay_; |
+void AudioSender::OnAck(uint32 frame_id) { |
} |
} // namespace cast |