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Fix jitter buffer delay reporting.

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Previously, if more than one packet is extracted in a GetAudio call then
an incorrect number of samples will be reported.

Bug: webrtc:10363
Change-Id: Ia1bcc87a0e0082060e4f746d37a4008735eec6b3
Reviewed-on: https://webrtc-review.googlesource.com/c/124829
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26903}
This commit is contained in:
Jakob Ivarsson
2019-02-28 09:55:49 +01:00
committed by Commit Bot
parent c58c01d6d4
commit 26c59ff6ca
2 changed files with 33 additions and 1 deletions
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2
modules/audio_coding/neteq/neteq_impl.cc
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@ -1915,7 +1915,7 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
}
extracted_samples = packet->timestamp - first_timestamp + packet_duration;
stats_.JitterBufferDelay(extracted_samples, waiting_time_ms);
stats_.JitterBufferDelay(packet_duration, waiting_time_ms);
packet_list->push_back(std::move(*packet)); // Store packet in list.
packet = absl::nullopt; // Ensure it's never used after the move.