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Adds event traces and counters for WebRTC receive side.

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Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
edjee@google.com
2013-04-04 19:43:34 +00:00
parent 835dbf4516
commit 79b0289bfc
10 changed files with 345 additions and 4 deletions
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3
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
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@@ -22,6 +22,7 @@
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
@@ -2231,6 +2232,8 @@ AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const {
// Automatic resample to the requested frequency.
WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
WebRtc_Word32 desired_freq_hz, AudioFrame* audio_frame) {
TRACE_EVENT0("webrtc_voe", "ACM::PlayoutData10Ms");
bool stereo_mode;
if (GetSilence(desired_freq_hz, audio_frame))