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Coverity report: fixes an issue where the returnvalue of a function is not checked.

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Review URL: https://webrtc-codereview.appspot.com/347013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1542 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org 2012-01-24 23:43:54 +00:00
parent f5d8c3bc3b
commit 567b99be5f
8 changed files with 28 additions and 48 deletions
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6
src/modules/rtp_rtcp/interface/rtp_rtcp.h
View File

@ -1010,9 +1010,9 @@ public:
*
* return -1 on failure else 0
*/
virtual WebRtc_Word32 SetSendBitrate(const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit) = 0;
virtual void SetSendBitrate(const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit) = 0;
/*
* Turn on/off generic FEC

2
src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -243,7 +243,7 @@ class MockRtpRtcp : public RtpRtcp {
MOCK_METHOD1(SetCameraDelay,
WebRtc_Word32(const WebRtc_Word32 delayMS));
MOCK_METHOD3(SetSendBitrate,
WebRtc_Word32(const WebRtc_UWord32 startBitrate, const WebRtc_UWord16 minBitrateKbit, const WebRtc_UWord16 maxBitrateKbit));
void(const WebRtc_UWord32 startBitrate, const WebRtc_UWord16 minBitrateKbit, const WebRtc_UWord16 maxBitrateKbit));
MOCK_METHOD3(SetGenericFECStatus,
WebRtc_Word32(const bool enable, const WebRtc_UWord8 payloadTypeRED, const WebRtc_UWord8 payloadTypeFEC));
MOCK_METHOD3(GenericFECStatus,

3
src/modules/rtp_rtcp/source/bandwidth_management.cc
View File

@ -42,7 +42,7 @@ BandwidthManagement::~BandwidthManagement()
delete _critsect;
}
WebRtc_Word32
void
BandwidthManagement::SetSendBitrate(const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit)
@ -59,7 +59,6 @@ BandwidthManagement::SetSendBitrate(const WebRtc_UWord32 startBitrate,
{
_maxBitRateConfigured = maxBitrateKbit*1000;
}
return 0;
}
WebRtc_Word32

6
src/modules/rtp_rtcp/source/bandwidth_management.h
View File

@ -43,9 +43,9 @@ public:
WebRtc_Word32 AvailableBandwidth(WebRtc_UWord16* bandwidthKbit) const;
WebRtc_Word32 SetSendBitrate(const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit);
void SetSendBitrate(const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit);
WebRtc_Word32 MaxConfiguredBitrate(WebRtc_UWord16* maxBitrateKbit);

14
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -2192,10 +2192,9 @@ RtpVideoCodecTypes ModuleRtpRtcpImpl::SendVideoCodec() const {
return _rtpSender.VideoCodecType();
}
WebRtc_Word32 ModuleRtpRtcpImpl::SetSendBitrate(
const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit) {
void ModuleRtpRtcpImpl::SetSendBitrate(const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit) {
WEBRTC_TRACE(kTraceModuleCall,
kTraceRtpRtcp,
@ -2220,11 +2219,12 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetSendBitrate(
it++;
}
}
// TODO(henrike): this function also returns a value. It never fails so
// make it return void.
_rtpSender.SetTargetSendBitrate(startBitrate);
return _bandwidthManagement.SetSendBitrate(startBitrate,
minBitrateKbit,
maxBitrateKbit);
_bandwidthManagement.SetSendBitrate(startBitrate, minBitrateKbit,
maxBitrateKbit);
}
WebRtc_Word32 ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(

6
src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -460,9 +460,9 @@ public:
virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
virtual WebRtc_Word32 SetSendBitrate(const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit);
virtual void SetSendBitrate(const WebRtc_UWord32 startBitrate,
const WebRtc_UWord16 minBitrateKbit,
const WebRtc_UWord16 maxBitrateKbit);
virtual WebRtc_Word32 SetGenericFECStatus(const bool enable,
const WebRtc_UWord8 payloadTypeRED,

28
src/video_engine/vie_channel.cc
View File

@ -251,13 +251,8 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
std::min(start_bitrate, video_codec.simulcastStream[0].maxBitrate);
start_bitrate -= stream_bitrate;
// Set correct bitrate to base layer.
if (rtp_rtcp_.SetSendBitrate(
stream_bitrate, video_codec.minBitrate,
video_codec.simulcastStream[0].maxBitrate) != 0) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s: could not set send bitrates", __FUNCTION__);
return -1;
}
rtp_rtcp_.SetSendBitrate(stream_bitrate, video_codec.minBitrate,
video_codec.simulcastStream[0].maxBitrate);
// Create our simulcast RTP modules.
for (int i = simulcast_rtp_rtcp_.size();
i < video_codec.numberOfSimulcastStreams - 1;
@ -333,13 +328,8 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
const WebRtc_UWord32 stream_bitrate =
std::min(start_bitrate, video_codec.simulcastStream[idx].maxBitrate);
start_bitrate -= stream_bitrate;
if (rtp_rtcp->SetSendBitrate(
stream_bitrate, video_codec.minBitrate,
video_codec.simulcastStream[idx].maxBitrate) != 0) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s: could not set send bitrates", __FUNCTION__);
return -1;
}
rtp_rtcp->SetSendBitrate(stream_bitrate, video_codec.minBitrate,
video_codec.simulcastStream[idx].maxBitrate);
}
vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
} else {
@ -357,13 +347,9 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
// Clear any previous modules.
vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
if (rtp_rtcp_.SetSendBitrate(video_codec.startBitrate * 1000,
video_codec.minBitrate,
video_codec.maxBitrate) != 0) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s: could not set send bitrates", __FUNCTION__);
return -1;
}
rtp_rtcp_.SetSendBitrate(video_codec.startBitrate * 1000,
video_codec.minBitrate,
video_codec.maxBitrate);
}
// Enable this if H264 is available.
// This sets the wanted packetization mode.

11
src/video_engine/vie_encoder.cc
View File

@ -273,14 +273,9 @@ WebRtc_Word32 ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
video_codec.codecType, video_codec.width, video_codec.height);
// Convert from kbps to bps.
if (default_rtp_rtcp_.SetSendBitrate(video_codec.startBitrate * 1000,
video_codec.minBitrate,
video_codec.maxBitrate) != 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
ViEId(engine_id_, channel_id_),
"Could not set RTP module bitrates");
return -1;
}
default_rtp_rtcp_.SetSendBitrate(video_codec.startBitrate * 1000,
video_codec.minBitrate,
video_codec.maxBitrate);
// Setting target width and height for VPM.
if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,