From efe4edb6da601b2e1f038ca4938432df3026e5c0 Mon Sep 17 00:00:00 2001
From: "mikhal@webrtc.org"
 <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Wed, 6 Mar 2013 23:29:33 +0000
Subject: [PATCH] Enabling bufffering mode with no sync module or VoE

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
---
 .../auto_test/source/vie_autotest_rtp_rtcp.cc | 30 ++++++++++++-------
 webrtc/video_engine/vie_channel.cc            |  3 +-
 webrtc/video_engine/vie_sync_module.cc        |  9 ++++--
 webrtc/video_engine/vie_sync_module.h         |  2 +-
 4 files changed, 29 insertions(+), 15 deletions(-)

diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
index e944c0c59f..a95eab3125 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
@@ -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));
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 70d1a373d5..9b505a239f 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -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;
 }
 
diff --git a/webrtc/video_engine/vie_sync_module.cc b/webrtc/video_engine/vie_sync_module.cc
index e01b0e2719..e973898681 100644
--- a/webrtc/video_engine/vie_sync_module.cc
+++ b/webrtc/video_engine/vie_sync_module.cc
@@ -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
diff --git a/webrtc/video_engine/vie_sync_module.h b/webrtc/video_engine/vie_sync_module.h
index 8f10c61124..94dae57f2f 100644
--- a/webrtc/video_engine/vie_sync_module.h
+++ b/webrtc/video_engine/vie_sync_module.h
@@ -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();