AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate
A later change will allow them to differ. Bug: webrtc:10631 Change-Id: I4e13f41980261990b3bbbc6897cd754369265ca0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137046 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27991}
This commit is contained in:
@ -88,6 +88,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
|
||||
|
||||
int SampleRateHz() const override;
|
||||
size_t NumChannels() const override;
|
||||
int RtpTimestampRateHz() const override;
|
||||
size_t Num10MsFramesInNextPacket() const override;
|
||||
size_t Max10MsFramesInAPacket() const override;
|
||||
int GetTargetBitrate() const override;
|
||||
|
||||
Reference in New Issue
Block a user