From a4527c89e77ae6c8c07c935e960bc8d9ddff35b3 Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Thu, 3 Dec 2015 13:06:20 +0100 Subject: [PATCH] Add comments about the Audio parts of the public Call API being WIP. BUG=webrtc:4690 R=kjellander@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1493933003 . Cr-Commit-Position: refs/heads/master@{#10882} --- webrtc/audio_receive_stream.h | 5 +++++ webrtc/audio_send_stream.h | 5 +++++ webrtc/audio_state.h | 5 +++++ 3 files changed, 15 insertions(+) diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h index 3e5a518a7d..356a3a3928 100644 --- a/webrtc/audio_receive_stream.h +++ b/webrtc/audio_receive_stream.h @@ -24,6 +24,11 @@ namespace webrtc { class AudioDecoder; +// WORK IN PROGRESS +// This class is under development and is not yet intended for for use outside +// of WebRtc/Libjingle. Please use the VoiceEngine API instead. +// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 + class AudioReceiveStream : public ReceiveStream { public: struct Stats { diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h index c5db82b91b..7069c377d3 100644 --- a/webrtc/audio_send_stream.h +++ b/webrtc/audio_send_stream.h @@ -23,6 +23,11 @@ namespace webrtc { +// WORK IN PROGRESS +// This class is under development and is not yet intended for for use outside +// of WebRtc/Libjingle. Please use the VoiceEngine API instead. +// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 + class AudioSendStream : public SendStream { public: struct Stats { diff --git a/webrtc/audio_state.h b/webrtc/audio_state.h index c6168237a9..fa5784c844 100644 --- a/webrtc/audio_state.h +++ b/webrtc/audio_state.h @@ -18,6 +18,11 @@ namespace webrtc { class AudioDeviceModule; class VoiceEngine; +// WORK IN PROGRESS +// This class is under development and is not yet intended for for use outside +// of WebRtc/Libjingle. Please use the VoiceEngine API instead. +// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 + // AudioState holds the state which must be shared between multiple instances of // webrtc::Call for audio processing purposes. class AudioState : public rtc::RefCountInterface {