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Remove cast to LocalAudioSource from AudioRtpSender.

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We can't assume that the audio source implementation will be our own internal one and we shouldn't apply local audio options to a remote audio track this way either.

BUG=5423
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1576913002 .

Cr-Commit-Position: refs/heads/master@{#11341}
This commit is contained in:
Tommi 2016-01-21 16:12:17 +01:00
parent 32be07bc36
commit 3c16978c84
1 changed files with 5 additions and 0 deletions
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5
talk/app/webrtc/rtpsender.cc
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@ -199,12 +199,17 @@ void AudioRtpSender::Stop() {
void AudioRtpSender::SetAudioSend() {
RTC_DCHECK(!stopped_ && can_send_track());
cricket::AudioOptions options;
#if !defined(WEBRTC_CHROMIUM_BUILD)
// TODO(tommi): Remove this hack when we move CreateAudioSource out of
// PeerConnection. This is a bit of a strange way to apply local audio
// options since it is also applied to all streams/channels, local or remote.
if (track_->enabled() && track_->GetSource() &&
!track_->GetSource()->remote()) {
// TODO(xians): Remove this static_cast since we should be able to connect
// a remote audio track to a peer connection.
options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
}
#endif
// Use the renderer if the audio track has one, otherwise use the sink
// adapter owned by this class.