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Unified Diff: webrtc/modules/audio_coding/neteq/neteq_impl.cc

Issue 1948663002: NetEq: Update stats earlier in the GetAudioInternal call (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@timescale-holdoff
Patch Set: Created 4 years, 8 months ago
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Index: webrtc/modules/audio_coding/neteq/neteq_impl.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 555d3c71bcc3ad238518de1de699ddcc24ca7155..945ac5ce161bd15ffb0fbf9cb20ea905fa533021 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -815,6 +815,7 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame) {
Operations operation;
bool play_dtmf;
tick_timer_->Increment();
+ stats_.IncreaseCounter(output_size_samples_, fs_hz_);
int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
&play_dtmf);
if (return_value != 0) {
@@ -1029,9 +1030,6 @@ int NetEqImpl::GetDecision(Operations* operation,
*play_dtmf = false;
*operation = kUndefined;
- // Increment time counters.
- stats_.IncreaseCounter(output_size_samples_, fs_hz_);
-
assert(sync_buffer_.get());
uint32_t end_timestamp = sync_buffer_->end_timestamp();
if (!new_codec_) {
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