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Fixing bug in re-packing of stereo packets.

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BUG=341
TEST=voe_cmd_test, run G.722. First modify /src/modules/audio_coding_main/source/acm_codec_database.cc
@@ -149,7 +149,7 @@ const CodecInst ACMCodecDB::database_[] = {
   {kDynamicPayloadtypes[count_database++], "CELT", 32000, 320, 2, 64000},
 #endif
 #ifdef WEBRTC_CODEC_G722
-  {9, "G722", 16000, 320, 1, 64000},
+  {9, "G722", 16000, 320, 2, 64000},
 #endif
 #ifdef WEBRTC_CODEC_G722_1

Review URL: https://webrtc-codereview.appspot.com/454001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1930 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org 2012-03-23 10:01:11 +00:00
parent 3e820e5109
commit c231e4cb03
1 changed files with 5 additions and 5 deletions
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10
src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
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@ -449,7 +449,7 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
WebRtc_UWord16 s = payloadData[offsetBytes] << 8;
// check that we don't read outside the memory
if(offsetBytes < (WebRtc_UWord32)payloadLength -2)
if(offsetBytes < (WebRtc_UWord32)payloadLength - 1)
{
s += payloadData[offsetBytes+1];
}
@ -463,8 +463,8 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
offsetSamples += audioSpecific.bitsPerSample;
if(readShift <= audioSpecific.bitsPerSample)
{
// next does not fitt
// or fitt exactly
// next does not fit
// or fit exactly
offsetSamples -= 8;
offsetBytes++;
}
@ -485,8 +485,8 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
offsetSamplesInsert += audioSpecific.bitsPerSample;
if(insertShift <= audioSpecific.bitsPerSample)
{
// next does not fitt
// or fitt exactly
// next does not fit
// or fit exactly
offsetSamplesInsert -= 8;
offsetBytesInsert++;
}