Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(658)

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

Issue 1316523002: Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Rebase onto cleanup change Created 5 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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 86407f9f40617555c4d2ba9162795f590311d9ee..a9eb5a47d30c6a3375df62bf5200459dc55a5b0e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -66,7 +66,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t frequency,
- const uint8_t channels,
+ const size_t channels,
const uint32_t rate,
RtpUtility::Payload** payload) {
if (RtpUtility::StringCompare(payloadName, "cn", 2)) {

Powered by Google App Engine
This is Rietveld 408576698