Returning correct duration estimate on Opus DTX packets.

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

Index: webrtc/modules/audio_coding/codecs/opus/opus_interface.c
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 4f6e22f77d021a7161f2977f8404c34af43208a3..2ac53736650907ca4d54cfb1e1ccc42646f1116a 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -359,6 +359,12 @@ int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
 int WebRtcOpus_DurationEst(OpusDecInst* inst,
                            const uint8_t* payload,
                            size_t payload_length_bytes) {
+  if (payload_length_bytes == 0) {
+    // WebRtcOpus_Decode calls PLC when payload length is zero. So we return
+    // PLC duration correspondingly.
+    return WebRtcOpus_PlcDuration(inst);
+  }
+
   int frames, samples;
   frames = opus_packet_get_nb_frames(payload, (opus_int32)payload_length_bytes);
   if (frames < 0) {
@@ -373,6 +379,15 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
   return samples;
 }
 
+int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
+  /* The number of samples we ask for is |number_of_lost_frames| times
+   * |prev_decoded_samples_|. Limit the number of samples to maximum
+   * |kWebRtcOpusMaxFrameSizePerChannel|. */
+  const int plc_samples = inst->prev_decoded_samples;
+  return (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
+      plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
+}
+
 int WebRtcOpus_FecDurationEst(const uint8_t* payload,
                               size_t payload_length_bytes) {
   int samples;