Wire up pading.
Review URL: https://webrtc-codereview.appspot.com/509002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2094 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.cc b/src/modules/rtp_rtcp/source/rtp_sender.cc
index 7d12a70..3136b27 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -516,12 +516,34 @@
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 @@
}
}
+// Function triggered by timer.
void RTPSender::ProcessSendToNetwork() {
-
- // triggered by timer
WebRtc_UWord32 delta_time_ms;
{
CriticalSectionScoped cs(_sendCritsect);
@@ -828,12 +849,10 @@
if (!_transmissionSmoothing) {
return;
}
-
WebRtc_UWord32 now = _clock.GetTimeInMS();
delta_time_ms = now - _timeLastSendToNetworkUpdate;
_timeLastSendToNetworkUpdate = now;
}
-
_sendBucket.UpdateBytesPerInterval(delta_time_ms, _targetSendBitrate);
while (!_sendBucket.Empty()) {
diff --git a/src/modules/video_coding/main/source/encoded_frame.cc b/src/modules/video_coding/main/source/encoded_frame.cc
index 8e5d745..dff9df3 100644
--- a/src/modules/video_coding/main/source/encoded_frame.cc
+++ b/src/modules/video_coding/main/source/encoded_frame.cc
@@ -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 @@
{
return kVideoFrameAltRef;
}
+ case kSkipFrame:
+ {
+ return kFrameEmpty;
+ }
default:
{
return kVideoFrameDelta;
