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Unified Diff: webrtc/audio/audio_send_stream.h

Issue 1722253002: - Clean up unused voice engine DTMF code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@voe_dtmf_1
Patch Set: more remove Created 4 years, 10 months ago
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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;
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