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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Issue 1513303003: [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 3bc861ccdabf5ae8406321e3691a1c8984beb86b..86407f9f40617555c4d2ba9162795f590311d9ee 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -68,7 +68,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
const uint32_t frequency,
const uint8_t channels,
const uint32_t rate,
- RtpUtility::Payload*& payload) {
+ RtpUtility::Payload** payload) {
if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
CriticalSectionScoped cs(_sendAudioCritsect.get());
// we can have multiple CNG payload types
@@ -96,13 +96,13 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
return 0;
// The default timestamp rate is 8000 Hz, but other rates may be defined.
}
- payload = new RtpUtility::Payload;
- payload->typeSpecific.Audio.frequency = frequency;
- payload->typeSpecific.Audio.channels = channels;
- payload->typeSpecific.Audio.rate = rate;
- payload->audio = true;
- payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
- strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ *payload = new RtpUtility::Payload;
+ (*payload)->typeSpecific.Audio.frequency = frequency;
+ (*payload)->typeSpecific.Audio.channels = channels;
+ (*payload)->typeSpecific.Audio.rate = rate;
+ (*payload)->audio = true;
+ (*payload)->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
+ strncpy((*payload)->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
return 0;
}
@@ -384,13 +384,13 @@ int32_t RTPSenderAudio::SetRED(int8_t payloadType) {
}
// Get payload type for Redundant Audio Data RFC 2198
-int32_t RTPSenderAudio::RED(int8_t& payloadType) const {
+int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
CriticalSectionScoped cs(_sendAudioCritsect.get());
if (_REDPayloadType == -1) {
// not configured
return -1;
}
- payloadType = _REDPayloadType;
+ *payloadType = _REDPayloadType;
return 0;
}
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