Replace AudioSilenceDetector with an AudioPowerMonitor.

media/audio/audio_power_monitor.cc

Index: media/audio/audio_power_monitor.cc
diff --git a/media/audio/audio_power_monitor.cc b/media/audio/audio_power_monitor.cc
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index 0000000000000000000000000000000000000000..f2fac987e84804bf226f2a5ddf357afc37425f89
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+++ b/media/audio/audio_power_monitor.cc
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+// Copyright (c) 2013 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "media/audio/audio_power_monitor.h"
+
+#include <cmath>
+#include <limits>
+
+#include "base/bind.h"
+#include "base/float_util.h"
+#include "base/message_loop.h"
+#include "base/time.h"
+#include "media/base/audio_bus.h"
+
+namespace media {
+
+// dBFS reference level is 1.0f (the maximum possible value of a sample).
+static const float kReferenceLevel = 1.0f;
+
+const float AudioPowerMonitor::kZeroPowerDBFS =
+    -std::numeric_limits<float>::infinity();
+
+const float AudioPowerMonitor::kMaxPowerDBFS = 0.0f;
+
+AudioPowerMonitor::AudioPowerMonitor(
+    int sample_rate,
+    const base::TimeDelta& measurement_period,
+    base::MessageLoop* message_loop,
+    const PowerMeasurementCallback& callback)
+    : num_frames_per_callback_(sample_rate * measurement_period.InSecondsF()),
+      message_loop_(message_loop),
+      notify_power_level_(callback) {
+  DCHECK(message_loop_);
+  DCHECK(!notify_power_level_.is_null());
+  ResetScanAccumulations();
+}
+
+AudioPowerMonitor::~AudioPowerMonitor() {
+}
+
+void AudioPowerMonitor::Scan(const AudioBus& buffer, int num_frames) {
+  DCHECK_LE(num_frames, buffer.frames());
+  const int num_channels = buffer.channels();
+  if (num_frames <= 0 || num_channels <= 0)
+    return;
+
+  // The overall goal here is to compute the RMS of the amplitudes over all
+  // channels.  An accumulation of sum-of-squares is maintained over one or more
+  // calls to this method.  Then, once a sufficient number of frames have been
+  // scanned, the accumulated values are converted into dBFS units and a
+  // callback is attempted.
+
+  // Compute the sum of squared amplitudes for the audio signal in |buffer|.
+  //
+  // TODO(miu): Implement optimized SSE/NEON to efficiently calculate
+  // sum-of-squares (or dot product, for more general-purpose use?) in
+  // media/base/vector_math.
+  float sum_of_squares = 0.0f;
+  for (int i = 0; i < num_channels; ++i) {
+    const float* p = buffer.channel(i);
+    const float* const end_of_samples = p + num_frames;
+    for (; p < end_of_samples; ++p)
+      sum_of_squares += (*p) * (*p);
+  }

[Chris Rogers, 2013/05/16 22:39:02]

Instead of computing RMS power here in this inner loop, it's a lot better to
average over a long time period.

You can do this as follows by using a member state variable ave_power_
(initialized to zero), then:

for (; p < end_of_samples; ++p) {
  float sample = *p;  // avoid computing *p twice which is inefficient
  ave_power_ += (ave_power_ - sample*sample) * k;
}

// k is a time-constant value which is relatively small, for example 0.001

See the discreteTimeConstantForSampleRate() method in:
https://svn.webkit.org/repository/webkit/trunk/Source/WebCore/platform/audio/...

to see how to calculate this value for any "time-constant" at any sample-rate

This implements a 1st order low-pass filter which will smooth the sample values
and will retain the smoothing state across calls to Scan().

[miu, 2013/07/02 06:10:16]

Done.

+  if (base::IsNaN(sum_of_squares)) {
+    sum_of_squares = 0.0f;
+  } else {
+    const float max_before_clipping = (1.0f * 1.0f) * num_frames * num_channels;
+    if (sum_of_squares > max_before_clipping)
+      sum_of_squares = max_before_clipping;
+  }

[Chris Rogers, 2013/05/16 22:39:02]

For clipping, you need to check every single sample to make sure it does not
exceed the full-scale (range from -1 -> +1).  If even one sample exceeds that
range then you'll want to implement a peak-and-hold (register the clip and hold
the UI red for a second, then release)

Checking each sample is easy, since you just need to loop through all of them
and compare fabs(sample) > 1.0

[miu, 2013/07/02 06:10:16]

Good catch.  Done.

+
+  // Accumulate with existing values.
+  sum_of_squares_so_far_ += sum_of_squares / num_channels;
+  frames_so_far_ += num_frames;
+
+  // Once enough frames have been scanned, try to post a task to run the
+  // callback with the dBFS result.  The posting of the task is guaranteed to be
+  // non-blocking, and therefore could fail.  If that happens, the accumulated
+  // values will be retained.  We expect this to be a good policy since
+  // TryPostTask() should rarely fail.  Therefore, only rarely will the
+  // measurement period be stretched, and only rarely will it be stretched by
+  // more than one extra AudioBus' worth of frames.
+  if (frames_so_far_ >= num_frames_per_callback_) {
+    const float average_power = sum_of_squares_so_far_ / frames_so_far_;
+    const float measurement_in_dbfs =
+        10.0f * log10(average_power / (kReferenceLevel * kReferenceLevel));
+    if (message_loop_->TryPostTask(
+            FROM_HERE,
+            base::Bind(notify_power_level_, measurement_in_dbfs))) {
+      ResetScanAccumulations();
+    }
+  }
+}
+
+void AudioPowerMonitor::ResetScanAccumulations() {
+  sum_of_squares_so_far_ = 0.0f;
+  frames_so_far_ = 0;
+}
+
+}  // namespace media