/usr/include/thunderbird/AudioSegment.h is in thunderbird-dev 1:24.4.0+build1-0ubuntu1.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 | /* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* 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_AUDIOSEGMENT_H_
#define MOZILLA_AUDIOSEGMENT_H_
#include "MediaSegment.h"
#include "nsISupportsImpl.h"
#include "AudioSampleFormat.h"
#include "SharedBuffer.h"
namespace mozilla {
class AudioStream;
/**
* For auto-arrays etc, guess this as the common number of channels.
*/
const int GUESS_AUDIO_CHANNELS = 2;
// We ensure that the graph advances in steps that are multiples of the Web
// Audio block size
const uint32_t WEBAUDIO_BLOCK_SIZE_BITS = 7;
const uint32_t WEBAUDIO_BLOCK_SIZE = 1 << WEBAUDIO_BLOCK_SIZE_BITS;
void InterleaveAndConvertBuffer(const void** aSourceChannels,
AudioSampleFormat aSourceFormat,
int32_t aLength, float aVolume,
int32_t aChannels,
AudioDataValue* aOutput);
/**
* Given an array of input channels (aChannelData), downmix to aOutputChannels,
* interleave the channel data. A total of aOutputChannels*aDuration
* interleaved samples will be copied to a channel buffer in aOutput.
*/
void DownmixAndInterleave(const nsTArray<const void*>& aChannelData,
AudioSampleFormat aSourceFormat, int32_t aDuration,
float aVolume, uint32_t aOutputChannels,
AudioDataValue* aOutput);
/**
* An AudioChunk represents a multi-channel buffer of audio samples.
* It references an underlying ThreadSharedObject which manages the lifetime
* of the buffer. An AudioChunk maintains its own duration and channel data
* pointers so it can represent a subinterval of a buffer without copying.
* An AudioChunk can store its individual channels anywhere; it maintains
* separate pointers to each channel's buffer.
*/
struct AudioChunk {
typedef mozilla::AudioSampleFormat SampleFormat;
// Generic methods
void SliceTo(TrackTicks aStart, TrackTicks aEnd)
{
NS_ASSERTION(aStart >= 0 && aStart < aEnd && aEnd <= mDuration,
"Slice out of bounds");
if (mBuffer) {
MOZ_ASSERT(aStart < INT32_MAX, "Can't slice beyond 32-bit sample lengths");
for (uint32_t channel = 0; channel < mChannelData.Length(); ++channel) {
mChannelData[channel] = AddAudioSampleOffset(mChannelData[channel],
mBufferFormat, int32_t(aStart));
}
}
mDuration = aEnd - aStart;
}
TrackTicks GetDuration() const { return mDuration; }
bool CanCombineWithFollowing(const AudioChunk& aOther) const
{
if (aOther.mBuffer != mBuffer) {
return false;
}
if (mBuffer) {
NS_ASSERTION(aOther.mBufferFormat == mBufferFormat,
"Wrong metadata about buffer");
NS_ASSERTION(aOther.mChannelData.Length() == mChannelData.Length(),
"Mismatched channel count");
if (mDuration > INT32_MAX) {
return false;
}
for (uint32_t channel = 0; channel < mChannelData.Length(); ++channel) {
if (aOther.mChannelData[channel] != AddAudioSampleOffset(mChannelData[channel],
mBufferFormat, int32_t(mDuration))) {
return false;
}
}
}
return true;
}
bool IsNull() const { return mBuffer == nullptr; }
void SetNull(TrackTicks aDuration)
{
mBuffer = nullptr;
mChannelData.Clear();
mDuration = aDuration;
mVolume = 1.0f;
}
TrackTicks mDuration; // in frames within the buffer
nsRefPtr<ThreadSharedObject> mBuffer; // the buffer object whose lifetime is managed; null means data is all zeroes
nsTArray<const void*> mChannelData; // one pointer per channel; empty if and only if mBuffer is null
float mVolume; // volume multiplier to apply (1.0f if mBuffer is nonnull)
SampleFormat mBufferFormat; // format of frames in mBuffer (only meaningful if mBuffer is nonnull)
};
/**
* A list of audio samples consisting of a sequence of slices of SharedBuffers.
* The audio rate is determined by the track, not stored in this class.
*/
class AudioSegment : public MediaSegmentBase<AudioSegment, AudioChunk> {
public:
typedef mozilla::AudioSampleFormat SampleFormat;
AudioSegment() : MediaSegmentBase<AudioSegment, AudioChunk>(AUDIO) {}
void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
const nsTArray<const float*>& aChannelData,
int32_t aDuration)
{
AudioChunk* chunk = AppendChunk(aDuration);
chunk->mBuffer = aBuffer;
for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
chunk->mChannelData.AppendElement(aChannelData[channel]);
}
chunk->mVolume = 1.0f;
chunk->mBufferFormat = AUDIO_FORMAT_FLOAT32;
}
void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
const nsTArray<const int16_t*>& aChannelData,
int32_t aDuration)
{
AudioChunk* chunk = AppendChunk(aDuration);
chunk->mBuffer = aBuffer;
for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
chunk->mChannelData.AppendElement(aChannelData[channel]);
}
chunk->mVolume = 1.0f;
chunk->mBufferFormat = AUDIO_FORMAT_S16;
}
// Consumes aChunk, and returns a pointer to the persistent copy of aChunk
// in the segment.
AudioChunk* AppendAndConsumeChunk(AudioChunk* aChunk)
{
AudioChunk* chunk = AppendChunk(aChunk->mDuration);
chunk->mBuffer = aChunk->mBuffer.forget();
chunk->mChannelData.SwapElements(aChunk->mChannelData);
chunk->mVolume = aChunk->mVolume;
chunk->mBufferFormat = aChunk->mBufferFormat;
return chunk;
}
void ApplyVolume(float aVolume);
void WriteTo(AudioStream* aOutput);
static Type StaticType() { return AUDIO; }
};
}
#endif /* MOZILLA_AUDIOSEGMENT_H_ */
|