Index: webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc |
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc |
index eeaec865898dd5cd48798cf38a3d40f848c935ba..5a5fa1ac0a3058b12532a50c8713600e7ee5c51a 100644 |
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc |
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc |
@@ -44,6 +44,7 @@ VideoSender::VideoSender(PacketProcessorListener* listener, |
VideoSource* source, |
BandwidthEstimatorType estimator_type) |
: PacketSender(listener, source->flow_id()), |
+ running_(true), |
source_(source), |
bwe_(CreateBweSender(estimator_type, |
source_->bits_per_second() / 1000, |
@@ -76,16 +77,22 @@ void VideoSender::ProcessFeedbackAndGeneratePackets( |
std::max<int64_t>(std::min(time_ms, time_until_feedback_ms), 0); |
} |
Packets generated; |
+ |
source_->RunFor(time_to_run_ms, &generated); |
bwe_->OnPacketsSent(generated); |
+ |
packets->merge(generated, DereferencingComparator<Packet>); |
+ |
clock_.AdvanceTimeMilliseconds(time_to_run_ms); |
+ |
if (!feedbacks->empty()) { |
bwe_->GiveFeedback(*feedbacks->front()); |
delete feedbacks->front(); |
feedbacks->pop_front(); |
} |
+ |
bwe_->Process(); |
+ |
time_ms -= time_to_run_ms; |
} while (time_ms > 0); |
assert(feedbacks->empty()); |
@@ -101,6 +108,22 @@ void VideoSender::OnNetworkChanged(uint32_t target_bitrate_bps, |
source_->SetBitrateBps(target_bitrate_bps); |
} |
+void VideoSender::Pause() { |
+ running_ = false; |
+ source_->Pause(); |
+ bwe_->Pause(); |
+} |
+ |
+void VideoSender::Resume() { |
+ running_ = true; |
+ source_->Resume(); |
+ bwe_->Resume(); |
+} |
+ |
+uint32_t VideoSender::TargetBitrateKbps() { |
+ return (source_->bits_per_second() + 500) / 1000; |
+} |
+ |
PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener, |
VideoSource* source, |
BandwidthEstimatorType estimator) |
@@ -262,11 +285,56 @@ void PacedVideoSender::OnNetworkChanged(uint32_t target_bitrate_bps, |
PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000, 0); |
} |
+const int kNoLimit = std::numeric_limits<int>::max(); |
+const int kPacketSizeBytes = 1200; |
+ |
+TcpSender::TcpSender(PacketProcessorListener* listener, |
+ int flow_id, |
+ int64_t offset_ms) |
+ : PacketSender(listener, flow_id), |
+ cwnd_(10), |
+ ssthresh_(kNoLimit), |
+ ack_received_(false), |
+ last_acked_seq_num_(0), |
+ next_sequence_number_(0), |
+ offset_ms_(offset_ms), |
+ last_reduction_time_ms_(-1), |
+ last_rtt_ms_(0), |
+ total_sent_bytes_(0), |
+ send_limit_bytes_(kNoLimit), |
+ running_(true), |
+ last_generated_packets_ms_(0), |
+ num_recent_sent_packets_(0), |
+ bitrate_kbps_(0) { |
+} |
+ |
+TcpSender::TcpSender(PacketProcessorListener* listener, |
+ int flow_id, |
+ int64_t offset_ms, |
+ int send_limit_bytes) |
+ : TcpSender(listener, flow_id, offset_ms) { |
+ send_limit_bytes_ = send_limit_bytes; |
+} |
+ |
+void TcpSender::set_choke_filter(ChokeFilter* choke_filter) { |
+ choke_filter_ = choke_filter; |
+} |
+ |
void TcpSender::RunFor(int64_t time_ms, Packets* in_out) { |
if (clock_.TimeInMilliseconds() + time_ms < offset_ms_) { |
clock_.AdvanceTimeMilliseconds(time_ms); |
+ if (running_) { |
+ choke_filter_->PauseFlow(*flow_ids().begin()); |
+ running_ = false; |
+ } |
return; |
} |
+ |
+ if (!running_) { |
+ choke_filter_->ResumeFlow(*flow_ids().begin()); |
stefan-webrtc
2015/07/02 11:03:42
I don't like that the TcpSender can access chokes.
magalhaesc
2015/07/02 17:17:02
Removed
|
+ running_ = true; |
+ } |
+ |
int64_t start_time_ms = clock_.TimeInMilliseconds(); |
BWE_TEST_LOGGING_CONTEXT("Sender"); |
BWE_TEST_LOGGING_CONTEXT(*flow_ids().begin()); |
@@ -359,15 +427,47 @@ void TcpSender::HandleLoss() { |
Packets TcpSender::GeneratePackets(size_t num_packets) { |
Packets generated; |
+ |
+ UpdateSendBitrateEstimate(num_packets); |
+ |
for (size_t i = 0; i < num_packets; ++i) { |
- generated.push_back(new MediaPacket(*flow_ids().begin(), |
- 1000 * clock_.TimeInMilliseconds(), |
- 1200, next_sequence_number_++)); |
+ if ((total_sent_bytes_ + kPacketSizeBytes) > send_limit_bytes_) { |
+ if (running_) { |
+ choke_filter_->PauseFlow(*flow_ids().begin()); |
+ running_ = false; |
+ } |
+ break; |
+ } |
+ generated.push_back( |
+ new MediaPacket(*flow_ids().begin(), 1000 * clock_.TimeInMilliseconds(), |
+ kPacketSizeBytes, next_sequence_number_++)); |
generated.back()->set_sender_timestamp_us( |
1000 * clock_.TimeInMilliseconds()); |
+ |
+ total_sent_bytes_ += kPacketSizeBytes; |
} |
+ |
return generated; |
} |
+ |
+void TcpSender::UpdateSendBitrateEstimate(size_t num_packets) { |
+ const int kTimeWindowMs = 500; |
+ num_recent_sent_packets_ += num_packets; |
+ |
+ int64_t delta_ms = clock_.TimeInMilliseconds() - last_generated_packets_ms_; |
+ if (delta_ms >= kTimeWindowMs) { |
+ bitrate_kbps_ = |
+ static_cast<uint32_t>(8 * num_recent_sent_packets_ * kPacketSizeBytes) / |
+ delta_ms; |
+ last_generated_packets_ms_ = clock_.TimeInMilliseconds(); |
+ num_recent_sent_packets_ = 0; |
+ } |
+} |
+ |
+uint32_t TcpSender::TargetBitrateKbps() { |
+ return bitrate_kbps_; |
+} |
+ |
} // namespace bwe |
} // namespace testing |
} // namespace webrtc |