Check current buffer time span instead of number of samples in postpone decoding after expand.
Bug: webrtc:10392
Change-Id: I2ad4d8c7a3f87cab32e2ea097b2e05aa179e0bc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126761
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27080}
diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc
index 343763b..e90fadc 100644
--- a/modules/audio_coding/neteq/packet_buffer.cc
+++ b/modules/audio_coding/neteq/packet_buffer.cc
@@ -287,6 +287,20 @@
return num_samples;
}
+size_t PacketBuffer::GetSpanSamples(size_t last_decoded_length) const {
+ if (buffer_.size() == 0) {
+ return 0;
+ }
+
+ size_t span = buffer_.back().timestamp - buffer_.front().timestamp;
+ if (buffer_.back().frame && buffer_.back().frame->Duration() > 0) {
+ span += buffer_.back().frame->Duration();
+ } else {
+ span += last_decoded_length;
+ }
+ return span;
+}
+
bool PacketBuffer::ContainsDtxOrCngPacket(
const DecoderDatabase* decoder_database) const {
RTC_DCHECK(decoder_database);