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Check current buffer time span instead of number of samples in postpone decoding after expand.

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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}
This commit is contained in:
Jakob Ivarsson
2019-03-12 15:12:08 +01:00
committed by Commit Bot
parent 075e7fdd00
commit 1b4254ada5
3 changed files with 24 additions and 3 deletions
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14
modules/audio_coding/neteq/packet_buffer.cc
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@ -287,6 +287,20 @@ size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
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);