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Add new UMA metric for NetEq target buffer delay

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The UMA metric will log the same information that goes into the
googPreferredJitterBufferMs stat.

Bug: webrtc:8488
Change-Id: I4e4e1e362dd42377105d52d2c4cd49c1ecb1a90d
Reviewed-on: https://webrtc-review.googlesource.com/26740
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20923}
This commit is contained in:
Henrik Lundin
2017-11-29 09:14:04 +01:00
committed by Commit Bot
parent 4e70a72571
commit abbff89b29
8 changed files with 25 additions and 3 deletions
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6
modules/audio_coding/acm2/audio_coding_module.cc
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@ -164,6 +164,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
int FilteredCurrentDelayMs() const override;
int TargetDelayMs() const override;
// Get 10 milliseconds of raw audio data to play out, and
// automatic resample to the requested frequency if > 0.
int PlayoutData10Ms(int desired_freq_hz,
@ -1193,6 +1195,10 @@ int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
return receiver_.FilteredCurrentDelayMs();
}
int AudioCodingModuleImpl::TargetDelayMs() const {
return receiver_.TargetDelayMs();
}
bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
if (!encoder_stack_) {
RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";