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Unified Diff: media/cast/sender/audio_sender.cc

Issue 542883004: Cast: Merge common functionality from audio/video sender into frame_sender. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: merge Created 6 years, 3 months ago
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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
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