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The level controller complexity tests have lately been

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flaky, with many false results and with a huge
variance.

This CL addresses that by changing the way the
API call durations are measured, using a warmup
period and a longer interval for computing the
timing estimates.

Furthermore, this CL reduces the number of tests
to compensate for the fact that the tests now are
more expensive, as well as to reduce the number
of regressions further.

BUG=webrtc:6614,webrtc:6685,666725

Review-Url: https://codereview.webrtc.org/2549403002
Cr-Commit-Position: refs/heads/master@{#15440}
This commit is contained in:
peah 2016-12-06 04:11:25 -08:00 committed by Commit bot
parent c9badd52c8
commit dbc960c045
1 changed files with 25 additions and 17 deletions
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42
webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
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@ -27,7 +27,10 @@
namespace webrtc {
namespace {
const size_t kNumFramesToProcess = 100;
const size_t kNumFramesToProcess = 300;
const size_t kNumFramesToProcessAtWarmup = 300;
const size_t kToTalNumFrames =
kNumFramesToProcess + kNumFramesToProcessAtWarmup;
std::string FormPerformanceMeasureString(const test::PerformanceTimer& timer) {
std::string s = std::to_string(timer.GetDurationAverage());
@ -45,12 +48,16 @@ void RunStandaloneSubmodule(int sample_rate_hz, size_t num_channels) {
LevelController level_controller;
level_controller.Initialize(sample_rate_hz);
for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) {
buffers.UpdateInputBuffers();
timer.StartTimer();
if (frame_no >= kNumFramesToProcessAtWarmup) {
timer.StartTimer();
}
level_controller.Process(buffers.capture_input_buffer.get());
timer.StopTimer();
if (frame_no >= kNumFramesToProcessAtWarmup) {
timer.StopTimer();
}
}
webrtc::test::PrintResultMeanAndError(
"level_controller_call_durations",
@ -120,27 +127,34 @@ void RunTogetherWithApm(std::string test_description,
StreamConfig capture_output_config(capture_output_sample_rate_hz,
num_channels, false);
for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) {
buffers.UpdateInputBuffers();
total_timer.StartTimer();
render_timer.StartTimer();
if (frame_no >= kNumFramesToProcessAtWarmup) {
total_timer.StartTimer();
render_timer.StartTimer();
}
ASSERT_EQ(AudioProcessing::kNoError,
apm->ProcessReverseStream(
&buffers.render_input[0], render_input_config,
render_output_config, &buffers.render_output[0]));
render_timer.StopTimer();
if (frame_no >= kNumFramesToProcessAtWarmup) {
render_timer.StopTimer();
capture_timer.StartTimer();
}
capture_timer.StartTimer();
ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0));
ASSERT_EQ(
AudioProcessing::kNoError,
apm->ProcessStream(&buffers.capture_input[0], capture_input_config,
capture_output_config, &buffers.capture_output[0]));
capture_timer.StopTimer();
total_timer.StopTimer();
if (frame_no >= kNumFramesToProcessAtWarmup) {
capture_timer.StopTimer();
total_timer.StopTimer();
}
}
webrtc::test::PrintResultMeanAndError(
@ -189,15 +203,9 @@ void TestSomeSampleRatesWithApm(const std::string& test_name,
bool include_default_apm_processing) {
// Test some stereo combinations first.
size_t num_channels = 2;
RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate8kHz,
AudioProcessing::kSampleRate48kHz, num_channels,
use_mobile_agc, include_default_apm_processing);
RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate16kHz,
AudioProcessing::kSampleRate32kHz, num_channels,
use_mobile_agc, include_default_apm_processing);
RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate32kHz,
AudioProcessing::kSampleRate16kHz, num_channels,
use_mobile_agc, include_default_apm_processing);
RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz,
AudioProcessing::kSampleRate8kHz, num_channels,
use_mobile_agc, include_default_apm_processing);