Fixing bug in re-packing of stereo packets.
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
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@ -449,7 +449,7 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
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WebRtc_UWord16 s = payloadData[offsetBytes] << 8;
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WebRtc_UWord16 s = payloadData[offsetBytes] << 8;
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// check that we don't read outside the memory
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// check that we don't read outside the memory
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if(offsetBytes < (WebRtc_UWord32)payloadLength -2)
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if(offsetBytes < (WebRtc_UWord32)payloadLength - 1)
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{
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{
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s += payloadData[offsetBytes+1];
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s += payloadData[offsetBytes+1];
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}
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}
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@ -463,8 +463,8 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
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offsetSamples += audioSpecific.bitsPerSample;
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offsetSamples += audioSpecific.bitsPerSample;
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if(readShift <= audioSpecific.bitsPerSample)
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if(readShift <= audioSpecific.bitsPerSample)
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{
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{
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// next does not fitt
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// next does not fit
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// or fitt exactly
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// or fit exactly
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offsetSamples -= 8;
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offsetSamples -= 8;
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offsetBytes++;
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offsetBytes++;
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}
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}
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@ -485,8 +485,8 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
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offsetSamplesInsert += audioSpecific.bitsPerSample;
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offsetSamplesInsert += audioSpecific.bitsPerSample;
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if(insertShift <= audioSpecific.bitsPerSample)
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if(insertShift <= audioSpecific.bitsPerSample)
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{
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{
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// next does not fitt
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// next does not fit
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// or fitt exactly
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// or fit exactly
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offsetSamplesInsert -= 8;
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offsetSamplesInsert -= 8;
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offsetBytesInsert++;
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offsetBytesInsert++;
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}
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}
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