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authorfischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-02-20 23:13:46 +0000
committerfischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-02-20 23:13:46 +0000
commit59b3a4ca1db1bb16568f4d184c7cd13e9c1f5786 (patch)
tree7474642d95c13809250b327272fe55635a2e21a5 /voice_engine/voe_video_sync_impl.cc
parent3a4f3f1ce4f4576f8129bfe900d5cc4737936c4b (diff)
downloadwebrtc-59b3a4ca1db1bb16568f4d184c7cd13e9c1f5786.tar.gz
Misc cleanups to webrtc/android code:
- Replace some deprecated calls/enums with their more modern equivalents. - Clean up some usage of global data and/or hide it better - Catch specific exceptions instead of Exception, and log the exception instead of just its message. - Random log message cleanups - Added a build_with_libjingle gyp variable to mimic build_with_chromium for when webrtc is built as part of a libjingle project but not part of chromium. BUG=webrtc:1169 TEST=none Review URL: https://webrtc-codereview.appspot.com/1105010 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3554 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'voice_engine/voe_video_sync_impl.cc')
-rw-r--r--voice_engine/voe_video_sync_impl.cc8
1 files changed, 1 insertions, 7 deletions
diff --git a/voice_engine/voe_video_sync_impl.cc b/voice_engine/voe_video_sync_impl.cc
index 7df46031..59f5218b 100644
--- a/voice_engine/voe_video_sync_impl.cc
+++ b/voice_engine/voe_video_sync_impl.cc
@@ -51,7 +51,6 @@ int VoEVideoSyncImpl::GetPlayoutTimestamp(int channel, unsigned int& timestamp)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"GetPlayoutTimestamp(channel=%d, timestamp=?)", channel);
- ANDROID_NOT_SUPPORTED(_shared->statistics());
IPHONE_NOT_SUPPORTED(_shared->statistics());
if (!_shared->statistics().Initialized())
@@ -76,7 +75,6 @@ int VoEVideoSyncImpl::SetInitTimestamp(int channel,
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetInitTimestamp(channel=%d, timestamp=%lu)",
channel, timestamp);
- ANDROID_NOT_SUPPORTED(_shared->statistics());
IPHONE_NOT_SUPPORTED(_shared->statistics());
if (!_shared->statistics().Initialized())
@@ -101,7 +99,6 @@ int VoEVideoSyncImpl::SetInitSequenceNumber(int channel,
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetInitSequenceNumber(channel=%d, sequenceNumber=%hd)",
channel, sequenceNumber);
- ANDROID_NOT_SUPPORTED(_shared->statistics());
IPHONE_NOT_SUPPORTED(_shared->statistics());
if (!_shared->statistics().Initialized())
@@ -125,7 +122,6 @@ int VoEVideoSyncImpl::SetMinimumPlayoutDelay(int channel,int delayMs)
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetMinimumPlayoutDelay(channel=%d, delayMs=%d)",
channel, delayMs);
- ANDROID_NOT_SUPPORTED(_shared->statistics());
IPHONE_NOT_SUPPORTED(_shared->statistics());
if (!_shared->statistics().Initialized())
@@ -172,7 +168,6 @@ int VoEVideoSyncImpl::GetDelayEstimate(int channel, int& delayMs)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"GetDelayEstimate(channel=%d, delayMs=?)", channel);
- ANDROID_NOT_SUPPORTED(_shared->statistics());
IPHONE_NOT_SUPPORTED(_shared->statistics());
if (!_shared->statistics().Initialized())
@@ -195,7 +190,6 @@ int VoEVideoSyncImpl::GetPlayoutBufferSize(int& bufferMs)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"GetPlayoutBufferSize(bufferMs=?)");
- ANDROID_NOT_SUPPORTED(_shared->statistics());
IPHONE_NOT_SUPPORTED(_shared->statistics());
if (!_shared->statistics().Initialized())
@@ -223,7 +217,7 @@ int VoEVideoSyncImpl::GetRtpRtcp(int channel, RtpRtcp* &rtpRtcpModule)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"GetRtpRtcp(channel=%i)", channel);
-
+
if (!_shared->statistics().Initialized())
{
_shared->SetLastError(VE_NOT_INITED, kTraceError);