Files
platform-external-webrtc/webrtc/media/base/audiorenderer.h
Taylor Brandstetter 1a018dcda3 Prevent a voice channel from sending data before a source is set.
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}
2016-03-08 20:37:48 +00:00

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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MEDIA_BASE_AUDIORENDERER_H_
#define WEBRTC_MEDIA_BASE_AUDIORENDERER_H_
// TODO(deadbeef): Remove this once not included by Chromium.
#endif // WEBRTC_MEDIA_BASE_AUDIORENDERER_H_