WebRTC Opus C interface: Add support for non-48 kHz encode sample rate
Plus tests fo 16 kHz.
Bug: webrtc:10631
Change-Id: I162c40b6120d7e308e535faba7501e437b0b5dc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137047
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28029}
diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h
index f5bcc45..54ecadd 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.h
+++ b/modules/audio_coding/codecs/opus/opus_interface.h
@@ -35,6 +35,7 @@
* Favor speech intelligibility.
* 1 - Audio applications.
* Favor faithfulness to the original input.
+ * - sample_rate_hz : sample rate of input audio
*
* Output:
* - inst : a pointer to Encoder context that is created
@@ -45,7 +46,8 @@
*/
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
size_t channels,
- int32_t application);
+ int32_t application,
+ int sample_rate_hz);
/****************************************************************************
* WebRtcOpus_MultistreamEncoderCreate(...)