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Fix the encoder pause logic.

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BUG=1691

Review URL: https://webrtc-codereview.appspot.com/1352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3904 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2013-04-25 22:20:08 +00:00
parent 381da4be9c
commit 91563e42da
1 changed files with 17 additions and 10 deletions
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27
webrtc/video_engine/vie_encoder.cc
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@ -32,6 +32,10 @@ namespace webrtc {
// Pace in kbits/s until we receive first estimate.
static const int kInitialPace = 2000;
// Don't stop the encoder unless the delay is above this configured value.
static const int kMinPacingDelayMs = 200;
// Allow packets to be transmitted in up to 2 times max video bitrate if the
// bandwidth estimate allows it.
// TODO(holmer): Expose transmission start, min and max bitrates in the
@ -471,12 +475,19 @@ void ViEEncoder::TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
}
bool ViEEncoder::EncoderPaused() const {
// Pause video if paused by caller or as long as the network is down and the
// pacer queue has grown too large.
const bool max_send_buffer_reached =
paced_sender_->QueueInMs() >= target_delay_ms_;
return encoder_paused_ ||
(!network_is_transmitting_ && max_send_buffer_reached);
// Pause video if paused by caller or as long as the network is down or the
// pacer queue has grown too large in buffered mode.
if (encoder_paused_) {
return true;
}
if (target_delay_ms_ > 0) {
// Buffered mode.
// TODO(pwestin): Workaround until nack is configured as a time and not
// number of packets.
return paced_sender_->QueueInMs() >=
std::max(target_delay_ms_, kMinPacingDelayMs);
}
return !network_is_transmitting_;
}
RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
@ -775,10 +786,6 @@ int32_t ViEEncoder::SendData(
const RTPVideoHeader* rtp_video_hdr) {
{
CriticalSectionScoped cs(data_cs_.get());
if (EncoderPaused()) {
// Paused, don't send this packet.
return 0;
}
TRACE_EVENT2("webrtc", "VE::SendData",
"timestamp", time_stamp,
"capture_time_ms", capture_time_ms);