[Webkit-unassigned] [Bug 153828] [Gstreamer] Add volume and mute support to the WebRTC mediaplayer

[bugzilla-daemon at webkit.org]

https://bugs.webkit.org/show_bug.cgi?id=153828

Darin Adler <darin at apple.com> changed:

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--- Comment #4 from Darin Adler <darin at apple.com> ---
Comment on attachment 294703
  --> https://bugs.webkit.org/attachment.cgi?id=294703
[OpenWebRTC] audio mute/volume support

View in context: https://bugs.webkit.org/attachment.cgi?id=294703&action=review

> Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp:108
> +    if (!m_audioTrack)
> +        return;
> +
> +

Extra blank line here.

> Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp:109
> +    auto realTimeMediaSource = reinterpret_cast<RealtimeMediaSourceOwr*>(&m_audioTrack->source());

Having to use reinterpret_cast is usually a major danger sign; is there a way to use static_cast in the three places in this file instead?

I suggest making this be a reference instead of a pointer:

    auto& realTimeMediaSource = *some_kind_of_cast<RealtimeMediaSourceOwr&>(m_audioTrack->source());

> Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.h:77
> +    void setVolume(float) override;
> +    void setMuted(bool) override;

final instead of override?

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