Chromium Code Reviews| Index: webrtc/audio/audio_send_stream.h |
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h |
| index 8b96350590b90cf5e659d5d805cf6be62573130f..391cd8def7cf60d3700bfdfbbc6116c5ff2fa4ba 100644 |
| --- a/webrtc/audio/audio_send_stream.h |
| +++ b/webrtc/audio/audio_send_stream.h |
| @@ -40,7 +40,7 @@ class AudioSendStream final : public webrtc::AudioSendStream { |
| // webrtc::AudioSendStream implementation. |
| bool SendTelephoneEvent(int payload_type, uint8_t event, |
| - uint32_t duration_ms) override; |
| + uint16_t duration_ms) override; |
|
hlundin-webrtc
2016/02/24 10:33:20
Why not generalize and make it an int? This goes f
the sun
2016/03/08 15:20:48
Done.
|
| webrtc::AudioSendStream::Stats GetStats() const override; |
| const webrtc::AudioSendStream::Config& config() const; |