webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h - Issue 2206743002: Revert of Add EncodedImageCallback::OnEncodedImage().

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

Issue 2206743002: Revert of Add EncodedImageCallback::OnEncodedImage(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 4 years, 4 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 d540593923b8f4a25e756adbb70b819afee0785e..cb3ddb2ad3b8e45920b9306aca6d6010f6aee940 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

@@ -34,12 +34,12 @@

uint32_t rate,

RtpUtility::Payload** payload);

- bool SendAudio(FrameType frame_type,

- int8_t payload_type,

- uint32_t capture_timestamp,

- const uint8_t* payload_data,

- size_t payload_size,

- const RTPFragmentationHeader* fragmentation);

+ int32_t SendAudio(FrameType frame_type,

+ int8_t payload_type,

+ uint32_t capture_timestamp,

+ const uint8_t* payload_data,

+ size_t payload_size,

+ const RTPFragmentationHeader* fragmentation);

// set audio packet size, used to determine when it's time to send a DTMF

// packet in silence (CNG)

@@ -62,7 +62,7 @@

int32_t RED(int8_t* payload_type) const;

protected:

- bool SendTelephoneEventPacket(

+ int32_t SendTelephoneEventPacket(

bool ended,

int8_t dtmf_payload_type,

uint32_t dtmf_timestamp,

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