[asterisk-scf-commits] asterisk-scf/integration/media_operations_core.git branch "jitterbuffer" updated.

[Commits to the Asterisk SCF project code repositories]

branch "jitterbuffer" has been updated
       via  6905d722c0c98250cb314b56d544102fcb40f2b3 (commit)
      from  9ba0da005d0586fc14b49a664b906420cf21ffa3 (commit)

Summary of changes:
 src/JitterBufferOperation.cpp |   55 +++++++++++++++++++---------------------
 1 files changed, 26 insertions(+), 29 deletions(-)


- Log -----------------------------------------------------------------
commit 6905d722c0c98250cb314b56d544102fcb40f2b3
Author: Joshua Colp <jcolp at digium.com>
Date:   Mon Oct 17 15:47:31 2011 -0300

    Add working code for placing payloads into the jitterbuffer and for scheduling.

diff --git a/src/JitterBufferOperation.cpp b/src/JitterBufferOperation.cpp
index 3e2a570..bf7c058 100644
--- a/src/JitterBufferOperation.cpp
+++ b/src/JitterBufferOperation.cpp
@@ -15,6 +15,7 @@
  */
 
 #include <boost/thread.hpp>
+#include <boost/asio/detail/socket_ops.hpp>
 
 #include <IceUtil/UUID.h>
 #include <IceUtil/Timer.h>
@@ -54,12 +55,14 @@ private:
                              const AudioFormatPtr& inFormat,
                              const AudioFormatPtr& outFormat,
                              boost::shared_mutex& mutex,
-                             int numFramesToPush) 
+                             int numFramesToPush,
+                             const NetEqWrapperPtr& netEqWrapper) 
                          : mSource(source),
                            mOutFormat(outFormat),
                            mInFormat(inFormat),
                            mLock(mutex),
-                           mNumFramesToPush(numFramesToPush)
+                           mNumFramesToPush(numFramesToPush),
+                           mNetEq(netEqWrapper)
         { 
         }
 
@@ -103,12 +106,12 @@ private:
 
     private:
         TranslatorSourcePtr mSource;
-        NetEqWrapperPtr mNetEq;  
         AudioFormatPtr mOutFormat;
         AudioFormatPtr mInFormat;
         long mSeqNumber;
         boost::shared_mutex &mLock;
         int mNumFramesToPush;
+        NetEqWrapperPtr mNetEq;
     };
 
     class JitterBuffer : public BufferedTranslator
@@ -122,8 +125,8 @@ private:
             : BufferedTranslator(source, inputFormat, outputFormat, logger),
               mNetEq(netEqWrapper),
               mSsrc(rand()),
-              mFrameCount(0),
-              mBufferSending(false)
+              mBufferSending(false),
+              mTimer(new IceUtil::Timer())
         {
             mInAudioFormat = AudioFormatPtr::dynamicCast(inputFormat);
             mOutAudioFormat = AudioFormatPtr::dynamicCast(outputFormat);
@@ -158,7 +161,12 @@ private:
             rtpInfo.timeStamp = (WebRtc_UWord32)streamFrame->timestamp;
             //rtpInfo.markerBit = // Assuming this one is safe to leave unset. 
 
+            // WebRTC expects the payload to come straight from the network so we need to put it back in that state
             ShortSeqPayloadPtr inPayload = ShortSeqPayloadPtr::dynamicCast(inFrame->payload);
+            std::transform(inPayload->payload.begin(), inPayload->payload.end(),
+                           inPayload->payload.begin(), boost::asio::detail::socket_ops::host_to_network_short);
+            Ice::ByteSeq bytes((Ice::Byte*) &inPayload->payload.front(),
+                               (Ice::Byte*) &inPayload->payload[inPayload->payload.size()]);
 
             {
                 boost::unique_lock<boost::shared_mutex> lock(mLock);
@@ -166,33 +174,23 @@ private:
                 // Buffer the frame. 
                 WebRtcNetEQ_RecInRTPStruct(netEqInstance, 
                                            &rtpInfo, 
-                                           (WebRtc_UWord8*)&inPayload->payload.front(),
-                                           (WebRtc_Word16)mInAudioFormat->frameSize,
+                                           &bytes.front(),
+                                           (WebRtc_Word16)bytes.size(),
                                            (WebRtc_UWord32)IceUtil::Time::now().toMilliSeconds());
             }
-            mFrameCount++;
 
-            if (!mBufferSending)
+            if (mBufferSending == false)
             {
-                int framesToBuffer = mNetEq->getBufferSizeInBytes() / mInAudioFormat->frameSize;
-                if (mFrameCount >= framesToBuffer)
-                {
-                    mBufferSending = true;
-
-                    // Compute the timer period, and use a configurable number of frames so
-                    // that the timer task isn't pushing single frames.
-                    const int framesToPush(10); // TBD... config item
-                    
-                    int frameDuration = mOutAudioFormat->sampleRate / mOutAudioFormat->frameSize;
-
-                    // Start a timer to push buffered frames.
-                    IceUtil::TimerTaskPtr sendTask = new JitterBufferSendTask(mSource, 
-                                                                              mInAudioFormat, 
-                                                                              mOutAudioFormat, 
-                                                                              mLock,
-                                                                              framesToPush);
-                    mTimer->scheduleRepeated(sendTask, IceUtil::Time::milliSeconds(frameDuration * framesToPush));
-                }
+                mBufferSending = true;
+
+                // Start a timer to push buffered frames.
+                IceUtil::TimerTaskPtr sendTask = new JitterBufferSendTask(mSource, 
+                                                                          mInAudioFormat, 
+                                                                          mOutAudioFormat, 
+                                                                          mLock,
+                                                                          1,
+                                                                          mNetEq);
+                mTimer->scheduleRepeated(sendTask, IceUtil::Time::milliSeconds(mOutAudioFormat->frameSize / (mOutAudioFormat->sampleRate / 1000)));
             }
         }
 
@@ -202,7 +200,6 @@ private:
         AudioFormatPtr mOutAudioFormat;
         NetEqWrapperPtr mNetEq;
         WebRtc_UWord32 mSsrc;
-        int mFrameCount;
         bool mBufferSending;
         IceUtil::TimerPtr mTimer;
     };

-----------------------------------------------------------------------


-- 
asterisk-scf/integration/media_operations_core.git