API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1384005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/video_engine/stream_synchronization.cc b/webrtc/video_engine/stream_synchronization.cc
index 6ad579c..9490d10 100644
--- a/webrtc/video_engine/stream_synchronization.cc
+++ b/webrtc/video_engine/stream_synchronization.cc
@@ -29,12 +29,14 @@
extra_video_delay_ms = 0;
last_video_delay_ms = 0;
extra_audio_delay_ms = 0;
+ last_audio_delay_ms = 0;
network_delay = 120;
}
int extra_video_delay_ms;
int last_video_delay_ms;
int extra_audio_delay_ms;
+ int last_audio_delay_ms;
int network_delay;
};
@@ -87,9 +89,9 @@
bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
int current_audio_delay_ms,
- int* extra_audio_delay_ms,
+ int* total_audio_delay_target_ms,
int* total_video_delay_target_ms) {
- assert(extra_audio_delay_ms && total_video_delay_target_ms);
+ assert(total_audio_delay_target_ms && total_video_delay_target_ms);
int current_video_delay_ms = *total_video_delay_target_ms;
WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, video_channel_id_,
@@ -173,17 +175,26 @@
new_video_delay_ms =
std::min(new_video_delay_ms, base_target_delay_ms_ + kMaxDeltaDelayMs);
- // Make sure that audio is never below our target.
- channel_delay_->extra_audio_delay_ms =
- std::max(base_target_delay_ms_, channel_delay_->extra_audio_delay_ms);
+ int new_audio_delay_ms;
+ if (channel_delay_->extra_audio_delay_ms > base_target_delay_ms_) {
+ new_audio_delay_ms = channel_delay_->extra_audio_delay_ms;
+ } else {
+ // No change to the audio delay. We are changing video and we only
+ // allow to change one at the time.
+ new_audio_delay_ms = channel_delay_->last_audio_delay_ms;
+ }
+
+ // Make sure that we don't go below the extra audio delay.
+ new_audio_delay_ms = std::max(
+ new_audio_delay_ms, channel_delay_->extra_audio_delay_ms);
// Verify we don't go above the maximum allowed audio delay.
- channel_delay_->extra_audio_delay_ms = std::min(
- channel_delay_->extra_audio_delay_ms,
- base_target_delay_ms_ + kMaxDeltaDelayMs);
+ new_audio_delay_ms =
+ std::min(new_audio_delay_ms, base_target_delay_ms_ + kMaxDeltaDelayMs);
- // Remember our last video delay.
+ // Remember our last audio and video delays.
channel_delay_->last_video_delay_ms = new_video_delay_ms;
+ channel_delay_->last_audio_delay_ms = new_audio_delay_ms;
WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, video_channel_id_,
"Sync video delay %d ms for video channel and audio delay %d for audio "
@@ -192,8 +203,8 @@
audio_channel_id_);
// Return values.
- *extra_audio_delay_ms = channel_delay_->extra_audio_delay_ms;
*total_video_delay_target_ms = new_video_delay_ms;
+ *total_audio_delay_target_ms = new_audio_delay_ms;
return true;
}
@@ -201,6 +212,8 @@
// Initial extra delay for audio (accounting for existing extra delay).
channel_delay_->extra_audio_delay_ms +=
target_delay_ms - base_target_delay_ms_;
+ channel_delay_->last_audio_delay_ms +=
+ target_delay_ms - base_target_delay_ms_;
// The video delay is compared to the last value (and how much we can update
// is limited by that as well).