From 332274dfefdf1f7a106c9f035f83eb8b001b2e44 Mon Sep 17 00:00:00 2001
From: Minyue Li <minyue@webrtc.org>
Date: Wed, 13 Nov 2019 16:05:46 +0100
Subject: [PATCH] Adding GetInDtx to WebRTC Opus Interface.

Bug: webrtc:11085
Change-Id: Ie9152cbe3f3c70f6febafb877852d68a831bcae9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159708
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29834}
---
 modules/audio_coding/codecs/opus/opus_interface.cc | 12 ++++++++++++
 modules/audio_coding/codecs/opus/opus_interface.h  | 14 ++++++++++++++
 2 files changed, 26 insertions(+)

diff --git a/modules/audio_coding/codecs/opus/opus_interface.cc b/modules/audio_coding/codecs/opus/opus_interface.cc
index 2f475cbed1..f98afd792f 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.cc
+++ b/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -370,6 +370,18 @@ int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels) {
   }
 }
 
+int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst) {
+  if (!inst) {
+    return -1;
+  }
+  int32_t in_dtx;
+  if (ENCODER_CTL(inst, OPUS_GET_IN_DTX(&in_dtx)) == 0) {
+    return in_dtx;
+  } else {
+    return -1;
+  }
+}
+
 int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
                                  size_t channels,
                                  int sample_rate_hz) {
diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h
index ef62e0d04d..e8de973010 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.h
+++ b/modules/audio_coding/codecs/opus/opus_interface.h
@@ -306,6 +306,20 @@ int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst);
  */
 int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth);
 
+/*
+ * WebRtcOpus_GetInDtx(...)
+ *
+ * Gets the DTX state of the encoder.
+ *
+ * Input:
+ *      - inst   : Encoder context
+ *
+ * Return value  : -1 - Error.
+ *                 1  - Last encoded frame was comfort noise update during DTX.
+ *                 0  - Last encoded frame was encoded with encoder not in DTX.
+ */
+int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst);
+
 /*
  * WebRtcOpus_SetForceChannels(...)
  *