| Index: webrtc/audio/audio_send_stream.h
|
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
|
| index d463b3da30fadefcf507f7682e797659338642e3..cf0a19ca4be36cd31f60855689a21dc7f895348e 100644
|
| --- a/webrtc/audio/audio_send_stream.h
|
| +++ b/webrtc/audio/audio_send_stream.h
|
| @@ -40,8 +40,8 @@
|
| bool DeliverRtcp(const uint8_t* packet, size_t length) override;
|
|
|
| // webrtc::AudioSendStream implementation.
|
| - bool SendTelephoneEvent(int payload_type, int event,
|
| - int duration_ms) override;
|
| + bool SendTelephoneEvent(int payload_type, uint8_t event,
|
| + uint32_t duration_ms) override;
|
| webrtc::AudioSendStream::Stats GetStats() const override;
|
|
|
| const webrtc::AudioSendStream::Config& config() const;
|
|
|