Make AudioEncoder::GetFrameLengthRange() pure virtual.
In order for WebRTC to be able to include packet overhead in its bitrate calculations, the AudioEncoder::GetFrameLengthRange() function must be implemented by all audio encoders. Making this member function pure virtual as per the following PSA: https://groups.google.com/forum/#!topic/discuss-webrtc/qscwYr38je0 Bug: webrtc:11427 Change-Id: I30d297ef05f57453bfc257624729559057cad118 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171517 Commit-Queue: Ali Tofigh <alito@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31127}
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@ -110,9 +110,4 @@ ANAStats AudioEncoder::GetANAStats() const {
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return ANAStats();
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return ANAStats();
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}
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}
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absl::optional<std::pair<TimeDelta, TimeDelta>>
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AudioEncoder::GetFrameLengthRange() const {
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return absl::nullopt;
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}
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} // namespace webrtc
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} // namespace webrtc
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@ -244,7 +244,7 @@ class AudioEncoder {
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// information. This is used to calculated the full bitrate range, including
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// information. This is used to calculated the full bitrate range, including
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// overhead.
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// overhead.
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virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
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virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
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const;
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const = 0;
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protected:
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protected:
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// Subclasses implement this to perform the actual encoding. Called by
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// Subclasses implement this to perform the actual encoding. Called by
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