Add m-esr52 at 52.6.0

1 files changed, 259 insertions, 0 deletions

diff --git a/dom/media/AudioSampleFormat.h b/dom/media/AudioSampleFormat.h
new file mode 100644
index 000000000..997180c65
--- /dev/null
+++ b/dom/media/AudioSampleFormat.h
@@ -0,0 +1,259 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef MOZILLA_AUDIOSAMPLEFORMAT_H_
+#define MOZILLA_AUDIOSAMPLEFORMAT_H_
+
+#include "nsAlgorithm.h"
+#include <algorithm>
+
+namespace mozilla {
+
+/**
+ * Audio formats supported in MediaStreams and media elements.
+ *
+ * Only one of these is supported by AudioStream, and that is determined
+ * at compile time (roughly, FLOAT32 on desktops, S16 on mobile). Media decoders
+ * produce that format only; queued AudioData always uses that format.
+ */
+enum AudioSampleFormat
+{
+  // Native-endian signed 16-bit audio samples
+  AUDIO_FORMAT_S16,
+  // Signed 32-bit float samples
+  AUDIO_FORMAT_FLOAT32,
+  // Silence: format will be chosen later
+  AUDIO_FORMAT_SILENCE,
+  // The format used for output by AudioStream.
+#ifdef MOZ_SAMPLE_TYPE_S16
+  AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_S16
+#else
+  AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_FLOAT32
+#endif
+};
+
+enum {
+  MAX_AUDIO_SAMPLE_SIZE = sizeof(float)
+};
+
+template <AudioSampleFormat Format> class AudioSampleTraits;
+
+template <> class AudioSampleTraits<AUDIO_FORMAT_FLOAT32> {
+public:
+  typedef float Type;
+};
+template <> class AudioSampleTraits<AUDIO_FORMAT_S16> {
+public:
+  typedef int16_t Type;
+};
+
+typedef AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type AudioDataValue;
+
+template<typename T> class AudioSampleTypeToFormat;
+
+template <> class AudioSampleTypeToFormat<float> {
+public:
+  static const AudioSampleFormat Format = AUDIO_FORMAT_FLOAT32;
+};
+
+template <> class AudioSampleTypeToFormat<short> {
+public:
+  static const AudioSampleFormat Format = AUDIO_FORMAT_S16;
+};
+
+// Single-sample conversion
+/*
+ * Use "2^N" conversion since it's simple, fast, "bit transparent", used by
+ * many other libraries and apparently behaves reasonably.
+ * http://blog.bjornroche.com/2009/12/int-float-int-its-jungle-out-there.html
+ * http://blog.bjornroche.com/2009/12/linearity-and-dynamic-range-in-int.html
+ */
+inline float
+AudioSampleToFloat(float aValue)
+{
+  return aValue;
+}
+inline float
+AudioSampleToFloat(int16_t aValue)
+{
+  return aValue/32768.0f;
+}
+inline float
+AudioSampleToFloat(int32_t aValue)
+{
+  return aValue/(float)(1U<<31);
+}
+
+template <typename T> T FloatToAudioSample(float aValue);
+
+template <> inline float
+FloatToAudioSample<float>(float aValue)
+{
+  return aValue;
+}
+template <> inline int16_t
+FloatToAudioSample<int16_t>(float aValue)
+{
+  float v = aValue*32768.0f;
+  float clamped = std::max(-32768.0f, std::min(32767.0f, v));
+  return int16_t(clamped);
+}
+
+template <typename T> T UInt8bitToAudioSample(uint8_t aValue);
+
+template <> inline float
+UInt8bitToAudioSample<float>(uint8_t aValue)
+{
+  return aValue * (static_cast<float>(2) / UINT8_MAX) - static_cast<float>(1);
+}
+template <> inline int16_t
+UInt8bitToAudioSample<int16_t>(uint8_t aValue)
+{
+  return (int16_t(aValue) << 8) + aValue + INT16_MIN;
+}
+
+template <typename T> T IntegerToAudioSample(int16_t aValue);
+
+template <> inline float
+IntegerToAudioSample<float>(int16_t aValue)
+{
+  return aValue / 32768.0f;
+}
+template <> inline int16_t
+IntegerToAudioSample<int16_t>(int16_t aValue)
+{
+  return aValue;
+}
+
+template <typename T> T Int24bitToAudioSample(int32_t aValue);
+
+template <> inline float
+Int24bitToAudioSample<float>(int32_t aValue)
+{
+  return aValue / static_cast<float>(1 << 23);
+}
+template <> inline int16_t
+Int24bitToAudioSample<int16_t>(int32_t aValue)
+{
+  return aValue / 256;
+}
+
+template<typename SrcT, typename DstT>
+inline void
+ConvertAudioSample(SrcT aIn, DstT& aOut);
+
+template<>
+inline void
+ConvertAudioSample(int16_t aIn, int16_t & aOut)
+{
+  aOut = aIn;
+}
+
+template<>
+inline void
+ConvertAudioSample(int16_t aIn, float& aOut)
+{
+  aOut = AudioSampleToFloat(aIn);
+}
+
+template<>
+inline void
+ConvertAudioSample(float aIn, float& aOut)
+{
+  aOut = aIn;
+}
+
+template<>
+inline void
+ConvertAudioSample(float aIn, int16_t& aOut)
+{
+  aOut = FloatToAudioSample<int16_t>(aIn);
+}
+
+// Sample buffer conversion
+
+template <typename From, typename To> inline void
+ConvertAudioSamples(const From* aFrom, To* aTo, int aCount)
+{
+  for (int i = 0; i < aCount; ++i) {
+    aTo[i] = FloatToAudioSample<To>(AudioSampleToFloat(aFrom[i]));
+  }
+}
+inline void
+ConvertAudioSamples(const int16_t* aFrom, int16_t* aTo, int aCount)
+{
+  memcpy(aTo, aFrom, sizeof(*aTo)*aCount);
+}
+inline void
+ConvertAudioSamples(const float* aFrom, float* aTo, int aCount)
+{
+  memcpy(aTo, aFrom, sizeof(*aTo)*aCount);
+}
+
+// Sample buffer conversion with scale
+
+template <typename From, typename To> inline void
+ConvertAudioSamplesWithScale(const From* aFrom, To* aTo, int aCount, float aScale)
+{
+  if (aScale == 1.0f) {
+    ConvertAudioSamples(aFrom, aTo, aCount);
+    return;
+  }
+  for (int i = 0; i < aCount; ++i) {
+    aTo[i] = FloatToAudioSample<To>(AudioSampleToFloat(aFrom[i])*aScale);
+  }
+}
+inline void
+ConvertAudioSamplesWithScale(const int16_t* aFrom, int16_t* aTo, int aCount, float aScale)
+{
+  if (aScale == 1.0f) {
+    ConvertAudioSamples(aFrom, aTo, aCount);
+    return;
+  }
+  if (0.0f <= aScale && aScale < 1.0f) {
+    int32_t scale = int32_t((1 << 16) * aScale);
+    for (int i = 0; i < aCount; ++i) {
+      aTo[i] = int16_t((int32_t(aFrom[i]) * scale) >> 16);
+    }
+    return;
+  }
+  for (int i = 0; i < aCount; ++i) {
+    aTo[i] = FloatToAudioSample<int16_t>(AudioSampleToFloat(aFrom[i])*aScale);
+  }
+}
+
+// In place audio sample scaling.
+inline void
+ScaleAudioSamples(float* aBuffer, int aCount, float aScale)
+{
+  for (int32_t i = 0; i < aCount; ++i) {
+    aBuffer[i] *= aScale;
+  }
+}
+
+inline void
+ScaleAudioSamples(short* aBuffer, int aCount, float aScale)
+{
+  int32_t volume = int32_t((1 << 16) * aScale);
+  for (int32_t i = 0; i < aCount; ++i) {
+    aBuffer[i] = short((int32_t(aBuffer[i]) * volume) >> 16);
+  }
+}
+
+inline const void*
+AddAudioSampleOffset(const void* aBase, AudioSampleFormat aFormat,
+                     int32_t aOffset)
+{
+  static_assert(AUDIO_FORMAT_S16 == 0, "Bad constant");
+  static_assert(AUDIO_FORMAT_FLOAT32 == 1, "Bad constant");
+  NS_ASSERTION(aFormat == AUDIO_FORMAT_S16 || aFormat == AUDIO_FORMAT_FLOAT32,
+               "Unknown format");
+
+  return static_cast<const uint8_t*>(aBase) + (aFormat + 1)*2*aOffset;
+}
+
+} // namespace mozilla
+
+#endif /* MOZILLA_AUDIOSAMPLEFORMAT_H_ */