summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/modules/audio_processing/rms_level.h
diff options
context:
space:
mode:
authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/rms_level.h
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/rms_level.h')
-rw-r--r--third_party/libwebrtc/modules/audio_processing/rms_level.h77
1 files changed, 77 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/rms_level.h b/third_party/libwebrtc/modules/audio_processing/rms_level.h
new file mode 100644
index 0000000000..fbece19ecd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/rms_level.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2014 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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+#define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Computes the root mean square (RMS) level in dBFs (decibels from digital
+// full-scale) of audio data. The computation follows RFC 6465:
+// https://tools.ietf.org/html/rfc6465
+// with the intent that it can provide the RTP audio level indication.
+//
+// The expected approach is to provide constant-sized chunks of audio to
+// Analyze(). When enough chunks have been accumulated to form a packet, call
+// Average() to get the audio level indicator for the RTP header.
+class RmsLevel {
+ public:
+ struct Levels {
+ int average;
+ int peak;
+ };
+
+ enum : int { kMinLevelDb = 127, kInaudibleButNotMuted = 126 };
+
+ RmsLevel();
+ ~RmsLevel();
+
+ // Can be called to reset internal states, but is not required during normal
+ // operation.
+ void Reset();
+
+ // Pass each chunk of audio to Analyze() to accumulate the level.
+ void Analyze(rtc::ArrayView<const int16_t> data);
+ void Analyze(rtc::ArrayView<const float> data);
+
+ // If all samples with the given `length` have a magnitude of zero, this is
+ // a shortcut to avoid some computation.
+ void AnalyzeMuted(size_t length);
+
+ // Computes the RMS level over all data passed to Analyze() since the last
+ // call to Average(). The returned value is positive but should be interpreted
+ // as negative as per the RFC. It is constrained to [0, 127]. Resets the
+ // internal state to start a new measurement period.
+ int Average();
+
+ // Like Average() above, but also returns the RMS peak value. Resets the
+ // internal state to start a new measurement period.
+ Levels AverageAndPeak();
+
+ private:
+ // Compares `block_size` with `block_size_`. If they are different, calls
+ // Reset() and stores the new size.
+ void CheckBlockSize(size_t block_size);
+
+ float sum_square_;
+ size_t sample_count_;
+ float max_sum_square_;
+ absl::optional<size_t> block_size_;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_