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This CL solves three remaining Coverity warnings.

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A few more members were left uninitialized, and one more size mismatch in a multiplication.

Review URL: https://webrtc-codereview.appspot.com/367001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1558 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org 2012-01-27 09:00:46 +00:00
parent 4bcd1776a6
commit cbe1de9aa6
3 changed files with 25 additions and 24 deletions
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29
src/modules/audio_coding/main/source/acm_g7291.cc
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@ -1,5 +1,5 @@
/* /*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source * that can be found in the LICENSE file in the root of the source
@ -44,7 +44,10 @@ namespace webrtc {
ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */) ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */)
: _encoderInstPtr(NULL), : _encoderInstPtr(NULL),
_decoderInstPtr(NULL) { _decoderInstPtr(NULL),
_myRate(32000),
_flag8kHz(0),
_flagG729mode(0) {
return; return;
} }
@ -163,21 +166,15 @@ ACMG729_1::SetBitRateSafe(
struct G729_1_inst_t_; struct G729_1_inst_t_;
ACMG729_1::ACMG729_1( ACMG729_1::ACMG729_1(WebRtc_Word16 codecID)
WebRtc_Word16 codecID): : _encoderInstPtr(NULL),
_encoderInstPtr(NULL), _decoderInstPtr(NULL),
_decoderInstPtr(NULL) _myRate(32000), // Default rate.
{ _flag8kHz(0),
_flagG729mode(0) {
// TODO(tlegrand): We should add codecID as a input variable to the
// constructor of ACMGenericCodec.
_codecID = codecID; _codecID = codecID;
// Our current G729.1 does not support Annex C
// which is DTX.
_hasInternalDTX = false;
// Default rate
_myRate = 32000;
_flag8kHz = 0;
_flagG729mode = 0;
return; return;
} }

9
src/modules/audio_coding/main/source/acm_speex.cc
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@ -1,5 +1,5 @@
/* /*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source * that can be found in the LICENSE file in the root of the source
@ -54,7 +54,12 @@ namespace webrtc {
#ifndef WEBRTC_CODEC_SPEEX #ifndef WEBRTC_CODEC_SPEEX
ACMSPEEX::ACMSPEEX(WebRtc_Word16 /* codecID*/) ACMSPEEX::ACMSPEEX(WebRtc_Word16 /* codecID*/)
: _encoderInstPtr(NULL), : _encoderInstPtr(NULL),
_decoderInstPtr(NULL) { _decoderInstPtr(NULL),
_complMode(0),
_vbrEnabled(false),
_encodingRate(-1),
_samplingFrequency(-1),
_samplesIn20MsAudio(-1) {
return; return;
} }

7
src/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@ -1,5 +1,5 @@
/* /*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source * that can be found in the LICENSE file in the root of the source
@ -1191,9 +1191,8 @@ match");
} else { } else {
// Copy payload data for future use. // Copy payload data for future use.
size_t length = static_cast<size_t>( size_t length = static_cast<size_t>(
audioFrame._payloadDataLengthInSamples * audio_channels * audioFrame._payloadDataLengthInSamples * audio_channels);
sizeof(WebRtc_UWord16)); memcpy(audio, audioFrame._payloadData, length * sizeof(WebRtc_UWord16));
memcpy(audio, audioFrame._payloadData, length);
} }
WebRtc_UWord32 currentTimestamp; WebRtc_UWord32 currentTimestamp;