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Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.

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This implementation will be replaced by a faster one and sparse will be removed.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1530913002

Cr-Commit-Position: refs/heads/master@{#11099}
This commit is contained in:
asapersson
2015-12-21 01:46:20 -08:00
committed by Commit bot
parent 9fca7e18c3
commit 53805324c0
12 changed files with 149 additions and 130 deletions
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17
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -1346,8 +1346,9 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
capture_nonlocked_.stream_delay_ms - capture_.last_stream_delay_ms;
if (diff_stream_delay_ms > kMinDiffDelayMs &&
capture_.last_stream_delay_ms != 0) {
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.PlatformReportedStreamDelayJump",
diff_stream_delay_ms, kMinDiffDelayMs, 1000, 100);
RTC_HISTOGRAM_COUNTS_SPARSE(
"WebRTC.Audio.PlatformReportedStreamDelayJump", diff_stream_delay_ms,
kMinDiffDelayMs, 1000, 100);
if (capture_.stream_delay_jumps == -1) {
capture_.stream_delay_jumps = 0; // Activate counter if needed.
}
@ -1364,9 +1365,9 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
aec_system_delay_ms - capture_.last_aec_system_delay_ms;
if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
capture_.last_aec_system_delay_ms != 0) {
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
100);
RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.AecSystemDelayJump",
diff_aec_system_delay_ms, kMinDiffDelayMs,
1000, 100);
if (capture_.aec_system_delay_jumps == -1) {
capture_.aec_system_delay_jumps = 0; // Activate counter if needed.
}
@ -1382,7 +1383,7 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
rtc::CritScope cs_capture(&crit_capture_);
if (capture_.stream_delay_jumps > -1) {
RTC_HISTOGRAM_ENUMERATION(
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
capture_.stream_delay_jumps, 51);
}
@ -1390,8 +1391,8 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
capture_.last_stream_delay_ms = 0;
if (capture_.aec_system_delay_jumps > -1) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
capture_.aec_system_delay_jumps, 51);
RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.Audio.NumOfAecSystemDelayJumps",
capture_.aec_system_delay_jumps, 51);
}
capture_.aec_system_delay_jumps = -1;
capture_.last_aec_system_delay_ms = 0;