Bitrate controller for VideoToolbox encoder.

webrtc/modules/video_coding/bitrate_adjuster.cc

Index: webrtc/modules/video_coding/bitrate_adjuster.cc
diff --git a/webrtc/modules/video_coding/bitrate_adjuster.cc b/webrtc/modules/video_coding/bitrate_adjuster.cc
new file mode 100644
index 0000000000000000000000000000000000000000..9f3d939288fb1af13013e0274706733d8b6727a3
--- /dev/null
+++ b/webrtc/modules/video_coding/bitrate_adjuster.cc
@@ -0,0 +1,124 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/include/bitrate_adjuster.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
+
+namespace webrtc {
+
+// 2 seconds.
+const uint32_t BitrateAdjuster::kBitrateUpdateIntervalMs = 2000;
+
+// 5 percent of original.
+const uint32_t BitrateAdjuster::kBitrateTolerancePct = 5;
+
+BitrateAdjuster::BitrateAdjuster(BitrateAdjusterObserver* observer)
+    : observer_(observer), bitrate_tracker_(30 * 20) {
+  RTC_DCHECK(observer_);
+  Reset();
+}
+
+void BitrateAdjuster::SetTargetBitrate(uint32_t bitrate_kbit) {
+  rtc::CritScope cs(&crit_);
+  target_bitrate_kbit_ = bitrate_kbit;
+}
+
+uint32_t BitrateAdjuster::GetTargetBitrate() const {
+  rtc::CritScope cs(&crit_);
+  return target_bitrate_kbit_;
+}
+
+uint32_t BitrateAdjuster::GetAdjustedBitrate() const {
+  rtc::CritScope cs(&crit_);
+  return adjusted_bitrate_kbit_;
+}
+
+void BitrateAdjuster::Update(size_t frame_size) {
+  rtc::CritScope cs(&crit_);
+  uint32_t current_time_ms = rtc::Time();
+  UpdateBitrate(current_time_ms);
+  UpdateBitrateTracker(frame_size, current_time_ms);
+}
+
+// Only safe to call this after Update calls have stopped
+void BitrateAdjuster::Reset() {
+  rtc::CritScope cs(&crit_);
+  target_bitrate_kbit_ = 0;
+  adjusted_bitrate_kbit_ = 0;
+  last_bitrate_update_time_ms_ = 0;
+  last_update_time_ms_ = 0;
+  bitrate_tracker_.Reset();
+}
+
+void BitrateAdjuster::UpdateBitrate(uint32_t current_time_ms) {
+  rtc::CritScope cs(&crit_);
+  if (current_time_ms - last_bitrate_update_time_ms_ <
+      kBitrateUpdateIntervalMs) {
+    return;
+  }
+  if (bitrate_tracker_.count() < 60) {

[sprang, 2016/02/09 10:46:26]

Please used a named constant for this, or derive from the existing ones

[tkchin_webrtc, 2016/02/11 00:03:20]

Done.

+    // Wait for some samples before doing anything. Roughly 2s of samples
+    // at 30fps.
+    return;
+  }
+  float target_bitrate = target_bitrate_kbit_;
+
+  // Compute exponentially weighted average of bitrate samples.
+  // 0.02 means last 60 samples represent about 70% of weight.
+  float bitrate = bitrate_tracker_.ComputeWeightedMean(0.02);
+  float error = target_bitrate - bitrate;

[sprang, 2016/02/09 10:46:26]

Can we perhaps use webrtc::RateStatistics instead of rtc::RollingAccumulator
here? Then we can get an explicit time based window instead of constants
assuming 30fps.

[tkchin_webrtc, 2016/02/11 00:03:20]

Sure, that seems like a good idea.
Weird to introduce dependency on remote_bitrate_estimator though. I'll move
ratestatistics to webrtc/base.

[sprang_webrtc, 2016/02/15 08:51:32]

That sounds great, thanks!

+
+  // Adjust if we've overshot by any amount or if we've undershot too much.
+  if (bitrate > target_bitrate ||
+      error * 100 > kBitrateTolerancePct * target_bitrate) {
+    // Adjust the bitrate by a fraction of the error.
+    float adjustment = .5 * error;
+    float adjusted_bitrate = target_bitrate + adjustment;
+
+    // Clamp the adjustment to [0.5 * target, 0.95 * target].
+    float min_bitrate = .5 * target_bitrate;
+    float max_bitrate =
+        static_cast<float>((100 - kBitrateTolerancePct) * target_bitrate) /
+        100.;

[sprang, 2016/02/09 10:46:26]

Why do we need to clamp it to below target_bitrate?

[tkchin_webrtc, 2016/02/11 00:03:20]

It seemed to reduce oscillations. We know that the encoder consistently
overshoots at 100%, so clamping it a bit under after it actually does so seems
to make sense.

[sprang_webrtc, 2016/02/15 08:51:32]

Fair enough, perhaps a comment to that effect would be good for posterity tough?
:)

+    adjusted_bitrate = std::max(adjusted_bitrate, min_bitrate);
+    adjusted_bitrate = std::min(adjusted_bitrate, max_bitrate);
+
+    // Set the adjustment if it's not already set.
+    float last_adjusted_bitrate = adjusted_bitrate_kbit_;
+    if (adjusted_bitrate != last_adjusted_bitrate) {
+      LOG(LS_INFO) << "Adjusting encoder bitrate:"
+                   << "\n  target_bitrate:" << target_bitrate
+                   << "\n  actual_bitrate:" << bitrate
+                   << "\n  previous_bitrate:" << (uint32_t)last_adjusted_bitrate
+                   << "\n  adjusted_bitrate:" << (uint32_t)adjusted_bitrate;
+      adjusted_bitrate_kbit_ = adjusted_bitrate;
+      observer_->OnBitrateAdjusted(adjusted_bitrate);

[sprang, 2016/02/09 10:46:26]

Perhaps we should have a limit on how often this will be called, so that we
don't accidentally screw with the encoder by calling it too often. Max every N
frames?

[tkchin_webrtc, 2016/02/11 00:03:20]

The update only happens once every t seconds, so it should be pretty safe.
Counting frames in addition to that seems fine to me as well.

[sprang_webrtc, 2016/02/15 08:51:32]

Ah, right. I'm fine with either then.

+    }
+  }
+  last_bitrate_update_time_ms_ = current_time_ms;
+}
+
+void BitrateAdjuster::UpdateBitrateTracker(size_t frame_size,
+                                           uint32_t current_time_ms) {
+  rtc::CritScope cs(&crit_);
+  if (current_time_ms > last_update_time_ms_) {
+    // Convert to kbit/s.
+    float bitrate_sample =
+        static_cast<float>(frame_size * 8) /
+        static_cast<float>(current_time_ms - last_update_time_ms_);
+    bitrate_tracker_.AddSample(bitrate_sample);
+  }
+  last_update_time_ms_ = current_time_ms;
+}
+
+}  // namespace webrtc