NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate
Well, in fact we need to return both. But return codec sample rate
separately and let the SdpAudioFormat contain the RTP clockrate,
otherwise we're essentially lying to our callers.
Bug: webrtc:11028
Change-Id: I40f36cb9db6b9824404ade6b0515a8312ff97009
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156307
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29444}
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 8ecb9b6..842869f 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -182,7 +182,7 @@
int last_output_sample_rate_hz() const override;
- absl::optional<SdpAudioFormat> GetDecoderFormat(
+ absl::optional<DecoderFormat> GetDecoderFormat(
int payload_type) const override;
// Flushes both the packet buffer and the sync buffer.