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

Issue 1316523002: Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Fix compile Created 4 years, 11 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index a3cee5e70720ba62e851858e2540276d5a03590d..1e96d17a67ddbfae7ab249a6b4d525ff05a7d1fb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -29,7 +29,7 @@ class RTPSenderAudio : public DTMFqueue {
int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int8_t payloadType,
uint32_t frequency,
- uint8_t channels,
+ size_t channels,
uint32_t rate,
RtpUtility::Payload** payload);
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