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Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )

Browse Source 
Reason for revert:
Appears to cause a regression to RampUpTest.SendSideAudioOnlyUpDownUpRtx:

https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus6%29/builds/626

Original issue's description:
> Use different restrictions of acked bitrate lag depending on operating point.
>
> Before this the BWE was allowed to operate freely up to 100 kbps. This isn't a good idea for audio BWE.
>
> BUG=webrtc:5079
>
> Committed: https://crrev.com/5932149c9aeaa7679ad0bc3183047766832ca907
> Cr-Commit-Position: refs/heads/master@{#15389}

TBR=terelius@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2547113002
Cr-Commit-Position: refs/heads/master@{#15394}
This commit is contained in:
deadbeef 2016-12-02 11:29:33 -08:00 committed by Commit bot
parent ef0cd5621d
commit 2e59a02dd4
2 changed files with 5 additions and 46 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

13
webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
View File

@ -228,14 +228,11 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t current_bitrate_bps,
default:
assert(false);
}
// Don't change the bit rate if the send side is too far off.
// We allow a bit more lag at very low rates to not too easily get stuck if
// the encoder produces uneven outputs.
const uint32_t max_bitrate_bps =
static_cast<uint32_t>(1.5f * incoming_bitrate_bps) + 10000;
if (current_bitrate_bps > current_bitrate_bps_ &&
current_bitrate_bps > max_bitrate_bps) {
current_bitrate_bps = std::max(current_bitrate_bps_, max_bitrate_bps);
if ((incoming_bitrate_bps > 100000 || current_bitrate_bps > 150000) &&
current_bitrate_bps > 1.5 * incoming_bitrate_bps) {
// Allow changing the bit rate if we are operating at very low rates
// Don't change the bit rate if the send side is too far off
current_bitrate_bps = current_bitrate_bps_;
time_last_bitrate_change_ = now_ms;
}
return current_bitrate_bps;

38
webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
View File

@ -89,42 +89,4 @@ TEST(AimdRateControlTest, GetLastBitrateDecrease) {
states.aimd_rate_control->GetLastBitrateDecreaseBps());
}
TEST(AimdRateControlTest, BweLimitedByAckedBitrate) {
auto states = CreateAimdRateControlStates();
constexpr int kAckedBitrate = 10000;
InitBitrate(states, kAckedBitrate,
states.simulated_clock->TimeInMilliseconds());
while (states.simulated_clock->TimeInMilliseconds() - kClockInitialTime <
20000) {
UpdateRateControl(states, kBwNormal, kAckedBitrate,
states.simulated_clock->TimeInMilliseconds());
states.simulated_clock->AdvanceTimeMilliseconds(100);
}
ASSERT_TRUE(states.aimd_rate_control->ValidEstimate());
EXPECT_EQ(static_cast<uint32_t>(1.5 * kAckedBitrate + 10000),
states.aimd_rate_control->LatestEstimate());
}
TEST(AimdRateControlTest, BweNotLimitedByDecreasingAckedBitrate) {
auto states = CreateAimdRateControlStates();
constexpr int kAckedBitrate = 100000;
InitBitrate(states, kAckedBitrate,
states.simulated_clock->TimeInMilliseconds());
while (states.simulated_clock->TimeInMilliseconds() - kClockInitialTime <
20000) {
UpdateRateControl(states, kBwNormal, kAckedBitrate,
states.simulated_clock->TimeInMilliseconds());
states.simulated_clock->AdvanceTimeMilliseconds(100);
}
ASSERT_TRUE(states.aimd_rate_control->ValidEstimate());
// If the acked bitrate decreases the BWE shouldn't be reduced to 1.5x
// what's being acked, but also shouldn't get to increase more.
uint32_t prev_estimate = states.aimd_rate_control->LatestEstimate();
UpdateRateControl(states, kBwNormal, kAckedBitrate / 2,
states.simulated_clock->TimeInMilliseconds());
uint32_t new_estimate = states.aimd_rate_control->LatestEstimate();
EXPECT_NEAR(new_estimate, static_cast<uint32_t>(1.5 * kAckedBitrate + 10000),
2000);
EXPECT_EQ(new_estimate, prev_estimate);
}
} // namespace webrtc