| Index: webrtc/voice_engine/channel_proxy.cc
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| diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
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| index 46bcef99423ed65d3ea0fe8616ced23b6975e740..4a0b3cc4b1e7dadc076220e3cdb4b437d85aeeef 100644
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| --- a/webrtc/voice_engine/channel_proxy.cc
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| +++ b/webrtc/voice_engine/channel_proxy.cc
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| @@ -15,6 +15,7 @@
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|  #include "webrtc/api/call/audio_sink.h"
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|  #include "webrtc/base/checks.h"
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|  #include "webrtc/base/logging.h"
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| +#include "webrtc/base/safe_minmax.h"
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|  #include "webrtc/call/rtp_transport_controller_send_interface.h"
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|  #include "webrtc/voice_engine/channel.h"
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|  
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| @@ -290,7 +291,7 @@ void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
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|    RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
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|    // Limit to range accepted by both VoE and ACM, so we're at least getting as
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|    // close as possible, instead of failing.
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| -  delay_ms = std::max(0, std::min(delay_ms, 10000));
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| +  delay_ms = rtc::SafeClamp(delay_ms, 0, 10000);
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|    int error = channel()->SetMinimumPlayoutDelay(delay_ms);
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|    if (0 != error) {
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|      LOG(LS_WARNING) << "Error setting minimum playout delay.";
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| 
 |