At the top level, setting a track on an RtpSender is equivalent to setting a source (previously called a renderer) on a voice send stream. An RtpSender without a track is not supposed to send data (not even muted data), so a send stream without a source shouldn't send data. Also replacing SendFlags with a boolean and implementing "Start" and "Stop" methods on AudioSendStream, which was planned anyway and simplifies this CL. R=pthatcher@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1741933002 . Cr-Commit-Position: refs/heads/master@{#11918}
17 lines
610 B
C
17 lines
610 B
C
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MEDIA_BASE_AUDIORENDERER_H_
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#define WEBRTC_MEDIA_BASE_AUDIORENDERER_H_
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// TODO(deadbeef): Remove this once not included by Chromium.
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#endif // WEBRTC_MEDIA_BASE_AUDIORENDERER_H_
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