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/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index c6af751..b53b5ad 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -143,6 +143,13 @@
enum ReturnCodes { kOK = 0, kFail = -1 };
+ // Return type for GetDecoderFormat.
+ struct DecoderFormat {
+ int sample_rate_hz;
+ int num_channels;
+ SdpAudioFormat sdp_format;
+ };
+
// Creates a new NetEq object, with parameters set in |config|. The |config|
// object will only have to be valid for the duration of the call to this
// method.
@@ -265,7 +272,7 @@
// Returns the decoder info for the given payload type. Returns empty if no
// such payload type was registered.
- virtual absl::optional<SdpAudioFormat> GetDecoderFormat(
+ virtual absl::optional<DecoderFormat> GetDecoderFormat(
int payload_type) const = 0;
// Flushes both the packet buffer and the sync buffer.