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Enabling bufffering mode with no sync module or VoE

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BUG= 1454

Review URL: https://webrtc-codereview.appspot.com/1149006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3625 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mikhal@webrtc.org
2013-03-06 23:29:33 +00:00
parent 488d4c9493
commit efe4edb6da
4 changed files with 29 additions and 15 deletions
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30
webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
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@ -686,6 +686,24 @@ void ViEAutoTest::ViERtpRtcpAPITest()
tbChannel.videoChannel, false));
// Buffering mode - sender side.
EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
invalid_channel_id, 0));
int invalid_delay = -1;
EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
tbChannel.videoChannel, invalid_delay));
invalid_delay = 15000;
EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
tbChannel.videoChannel, invalid_delay));
EXPECT_EQ(0, ViE.rtp_rtcp->SetSenderBufferingMode(
tbChannel.videoChannel, 5000));
// Buffering mode - receiver side.
// Run without VoE to verify it that does not crash, but return an error.
EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
tbChannel.videoChannel, 0));
EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
tbChannel.videoChannel, 2000));
// Set VoE (required to set up stream-sync).
webrtc::VoiceEngine* voice_engine = webrtc::VoiceEngine::Create();
EXPECT_TRUE(NULL != voice_engine);
@ -698,17 +716,6 @@ void ViEAutoTest::ViERtpRtcpAPITest()
EXPECT_EQ(0, ViE.base->ConnectAudioChannel(tbChannel.videoChannel,
audio_channel));
EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
invalid_channel_id, 0));
int invalid_delay = -1;
EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
tbChannel.videoChannel, invalid_delay));
invalid_delay = 15000;
EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
tbChannel.videoChannel, invalid_delay));
EXPECT_EQ(0, ViE.rtp_rtcp->SetSenderBufferingMode(
tbChannel.videoChannel, 5000));
// Buffering mode - receiver side.
EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
invalid_channel_id, 0));
EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
@ -718,6 +725,7 @@ void ViEAutoTest::ViERtpRtcpAPITest()
tbChannel.videoChannel, invalid_delay));
EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiverBufferingMode(
tbChannel.videoChannel, 5000));
// Real-time mode - sender side.
EXPECT_EQ(0, ViE.rtp_rtcp->SetSenderBufferingMode(
tbChannel.videoChannel, 0));

3
webrtc/video_engine/vie_channel.cc
View File

@ -799,7 +799,8 @@ int ViEChannel::SetReceiverBufferingMode(int target_delay_ms) {
}
vcm_.SetNackSettings(max_nack_list_size, max_nack_reordering_threshold_);
vcm_.SetMinReceiverDelay(target_delay_ms);
vie_sync_.SetTargetBufferingDelay(target_delay_ms);
if (vie_sync_.SetTargetBufferingDelay(target_delay_ms) < 0)
return -1;
return 0;
}

9
webrtc/video_engine/vie_sync_module.cc
View File

@ -172,14 +172,19 @@ WebRtc_Word32 ViESyncModule::Process() {
return 0;
}
void ViESyncModule::SetTargetBufferingDelay(int target_delay_ms) {
int ViESyncModule::SetTargetBufferingDelay(int target_delay_ms) {
CriticalSectionScoped cs(data_cs_.get());
if (!voe_sync_interface_) {
WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, vie_channel_->Id(),
"voe_sync_interface_ NULL, can't set playout delay.");
return -1;
}
sync_->SetTargetBufferingDelay(target_delay_ms);
// Setting initial playout delay to voice engine (video engine is updated via
// the VCM interface).
assert(voe_sync_interface_ != NULL);
voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_,
target_delay_ms);
return 0;
}
} // namespace webrtc

2
webrtc/video_engine/vie_sync_module.h
View File

@ -41,7 +41,7 @@ class ViESyncModule : public Module {
int VoiceChannel();
// Set target delay for buffering mode (0 = real-time mode).
void SetTargetBufferingDelay(int target_delay_ms);
int SetTargetBufferingDelay(int target_delay_ms);
// Implements Module.
virtual WebRtc_Word32 TimeUntilNextProcess();