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Wire up pading.

Browse Source 
Review URL: https://webrtc-codereview.appspot.com/509002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2094 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
2012-04-23 14:52:15 +00:00
parent 11654c2344
commit ddab60be56
2 changed files with 33 additions and 10 deletions
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37
src/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -516,12 +516,34 @@ RTPSender::SendOutgoingData(const FrameType frameType,
return _audio->SendAudio(frameType, payloadType, captureTimeStamp, payloadData, payloadSize,fragmentation);
} else
{
// assert audio frameTypes
assert(frameType == kVideoFrameKey ||
frameType == kVideoFrameDelta ||
frameType == kVideoFrameGolden ||
frameType == kVideoFrameAltRef);
// Assert on audio frameTypes.
assert(frameType != kAudioFrameSpeech &&
frameType != kAudioFrameCN);
// If the encoder generate an empty frame send pading.
if (frameType == kFrameEmpty) {
// Current bitrate since last estimate(1 second) averaged with the
// estimate since then, to get the most up to date bitrate.
uint32_t current_bitrate = BitrateNow();
int bitrate_diff = _targetSendBitrate * 1000 - current_bitrate;
if (bitrate_diff > 0) {
int bytes = 0;
if (current_bitrate == 0) {
// Start up phase. Send one 33.3 ms batch to start with.
bytes = (bitrate_diff / 8) / 30;
} else {
bytes = (bitrate_diff / 8);
// Cap at 200 ms of target send data.
int bytes_cap = _targetSendBitrate * 25; // 1000 / 8 / 5
if (bytes_cap > bytes) {
bytes = bytes_cap;
}
}
// Send pading data.
return SendPadData(payloadType, captureTimeStamp, bytes);
}
return 0;
}
return _video->SendVideo(videoType,
frameType,
payloadType,
@ -818,9 +840,8 @@ void RTPSender::UpdateNACKBitRate(const WebRtc_UWord32 bytes,
}
}
// Function triggered by timer.
void RTPSender::ProcessSendToNetwork() {
// triggered by timer
WebRtc_UWord32 delta_time_ms;
{
CriticalSectionScoped cs(_sendCritsect);
@ -828,12 +849,10 @@ void RTPSender::ProcessSendToNetwork() {
if (!_transmissionSmoothing) {
return;
}
WebRtc_UWord32 now = _clock.GetTimeInMS();
delta_time_ms = now - _timeLastSendToNetworkUpdate;
_timeLastSendToNetworkUpdate = now;
}
_sendBucket.UpdateBytesPerInterval(delta_time_ms, _targetSendBitrate);
while (!_sendBucket.Empty()) {

6
src/modules/video_coding/main/source/encoded_frame.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -216,6 +216,10 @@ webrtc::FrameType VCMEncodedFrame::ConvertFrameType(VideoFrameType frameType)
{
return kVideoFrameAltRef;
}
case kSkipFrame:
{
return kFrameEmpty;
}
default:
{
return kVideoFrameDelta;