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/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2017 - ROLI Ltd.
JUCE is an open source library subject to commercial or open-source
licensing.
By using JUCE, you agree to the terms of both the JUCE 5 End-User License
Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
27th April 2017).
End User License Agreement: www.juce.com/juce-5-licence
Privacy Policy: www.juce.com/juce-5-privacy-policy
Or: You may also use this code under the terms of the GPL v3 (see
www.gnu.org/licenses).
JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
DISCLAIMED.
==============================================================================
*/
namespace juce
{
class AudioFormat;
//==============================================================================
/**
Reads samples from an audio file stream.
A subclass that reads a specific type of audio format will be created by
an AudioFormat object.
@see AudioFormat, AudioFormatWriter
@tags{Audio}
*/
class JUCE_API AudioFormatReader
{
protected:
//==============================================================================
/** Creates an AudioFormatReader object.
@param sourceStream the stream to read from - this will be deleted
by this object when it is no longer needed. (Some
specialised readers might not use this parameter and
can leave it as nullptr).
@param formatName the description that will be returned by the getFormatName()
method
*/
AudioFormatReader (InputStream* sourceStream,
const String& formatName);
public:
/** Destructor. */
virtual ~AudioFormatReader();
//==============================================================================
/** Returns a description of what type of format this is.
E.g. "AIFF"
*/
const String& getFormatName() const noexcept { return formatName; }
//==============================================================================
/** Reads samples from the stream.
@param destChannels an array of float buffers into which the sample data for each
channel will be written. Channels that aren't needed can be null
@param numDestChannels the number of array elements in the destChannels array
@param startSampleInSource the position in the audio file or stream at which the samples
should be read, as a number of samples from the start of the
stream. It's ok for this to be beyond the start or end of the
available data - any samples that are out-of-range will be returned
as zeros.
@param numSamplesToRead the number of samples to read. If this is greater than the number
of samples that the file or stream contains. the result will be padded
with zeros
@returns true if the operation succeeded, false if there was an error. Note
that reading sections of data beyond the extent of the stream isn't an
error - the reader should just return zeros for these regions
@see readMaxLevels
*/
bool read (float* const* destChannels, int numDestChannels,
int64 startSampleInSource, int numSamplesToRead);
/** Reads samples from the stream.
@param destChannels an array of buffers into which the sample data for each
channel will be written.
If the format is fixed-point, each channel will be written
as an array of 32-bit signed integers using the full
range -0x80000000 to 0x7fffffff, regardless of the source's
bit-depth. If it is a floating-point format, you should cast
the resulting array to a (float**) to get the values (in the
range -1.0 to 1.0 or beyond)
If the format is stereo, then destChannels[0] is the left channel
data, and destChannels[1] is the right channel.
The numDestChannels parameter indicates how many pointers this array
contains, but some of these pointers can be null if you don't want to
read data for some of the channels
@param numDestChannels the number of array elements in the destChannels array
@param startSampleInSource the position in the audio file or stream at which the samples
should be read, as a number of samples from the start of the
stream. It's ok for this to be beyond the start or end of the
available data - any samples that are out-of-range will be returned
as zeros.
@param numSamplesToRead the number of samples to read. If this is greater than the number
of samples that the file or stream contains. the result will be padded
with zeros
@param fillLeftoverChannelsWithCopies if true, this indicates that if there's no source data available
for some of the channels that you pass in, then they should be filled with
copies of valid source channels.
E.g. if you're reading a mono file and you pass 2 channels to this method, then
if fillLeftoverChannelsWithCopies is true, both destination channels will be filled
with the same data from the file's single channel. If fillLeftoverChannelsWithCopies
was false, then only the first channel would be filled with the file's contents, and
the second would be cleared. If there are many channels, e.g. you try to read 4 channels
from a stereo file, then the last 3 would all end up with copies of the same data.
@returns true if the operation succeeded, false if there was an error. Note
that reading sections of data beyond the extent of the stream isn't an
error - the reader should just return zeros for these regions
@see readMaxLevels
*/
bool read (int* const* destChannels,
int numDestChannels,
int64 startSampleInSource,
int numSamplesToRead,
bool fillLeftoverChannelsWithCopies);
/** Fills a section of an AudioBuffer from this reader.
This will convert the reader's fixed- or floating-point data to
the buffer's floating-point format, and will try to intelligently
cope with mismatches between the number of channels in the reader
and the buffer.
*/
void read (AudioBuffer<float>* buffer,
int startSampleInDestBuffer,
int numSamples,
int64 readerStartSample,
bool useReaderLeftChan,
bool useReaderRightChan);
/** Finds the highest and lowest sample levels from a section of the audio stream.
This will read a block of samples from the stream, and measure the
highest and lowest sample levels from the channels in that section, returning
these as normalised floating-point levels.
@param startSample the offset into the audio stream to start reading from. It's
ok for this to be beyond the start or end of the stream.
@param numSamples how many samples to read
@param results this array will be filled with Range values for each channel.
The array must contain numChannels elements.
@param numChannelsToRead the number of channels of data to scan. This must be
more than zero, but not more than the total number of channels
that the reader contains
@see read
*/
virtual void readMaxLevels (int64 startSample, int64 numSamples,
Range<float>* results, int numChannelsToRead);
/** Finds the highest and lowest sample levels from a section of the audio stream.
This will read a block of samples from the stream, and measure the
highest and lowest sample levels from the channels in that section, returning
these as normalised floating-point levels.
@param startSample the offset into the audio stream to start reading from. It's
ok for this to be beyond the start or end of the stream.
@param numSamples how many samples to read
@param lowestLeft on return, this is the lowest absolute sample from the left channel
@param highestLeft on return, this is the highest absolute sample from the left channel
@param lowestRight on return, this is the lowest absolute sample from the right
channel (if there is one)
@param highestRight on return, this is the highest absolute sample from the right
channel (if there is one)
@see read
*/
virtual void readMaxLevels (int64 startSample, int64 numSamples,
float& lowestLeft, float& highestLeft,
float& lowestRight, float& highestRight);
/** Scans the source looking for a sample whose magnitude is in a specified range.
This will read from the source, either forwards or backwards between two sample
positions, until it finds a sample whose magnitude lies between two specified levels.
If it finds a suitable sample, it returns its position; if not, it will return -1.
There's also a minimumConsecutiveSamples setting to help avoid spikes or zero-crossing
points when you're searching for a continuous range of samples
@param startSample the first sample to look at
@param numSamplesToSearch the number of samples to scan. If this value is negative,
the search will go backwards
@param magnitudeRangeMinimum the lowest magnitude (inclusive) that is considered a hit, from 0 to 1.0
@param magnitudeRangeMaximum the highest magnitude (inclusive) that is considered a hit, from 0 to 1.0
@param minimumConsecutiveSamples if this is > 0, the method will only look for a sequence
of this many consecutive samples, all of which lie
within the target range. When it finds such a sequence,
it returns the position of the first in-range sample
it found (i.e. the earliest one if scanning forwards, the
latest one if scanning backwards)
*/
int64 searchForLevel (int64 startSample,
int64 numSamplesToSearch,
double magnitudeRangeMinimum,
double magnitudeRangeMaximum,
int minimumConsecutiveSamples);
//==============================================================================
/** The sample-rate of the stream. */
double sampleRate = 0;
/** The number of bits per sample, e.g. 16, 24, 32. */
unsigned int bitsPerSample = 0;
/** The total number of samples in the audio stream. */
int64 lengthInSamples = 0;
/** The total number of channels in the audio stream. */
unsigned int numChannels = 0;
/** Indicates whether the data is floating-point or fixed. */
bool usesFloatingPointData = false;
/** A set of metadata values that the reader has pulled out of the stream.
Exactly what these values are depends on the format, so you can
check out the format implementation code to see what kind of stuff
they understand.
*/
StringPairArray metadataValues;
/** The input stream, for use by subclasses. */
InputStream* input;
//==============================================================================
/** Get the channel layout of the audio stream. */
virtual AudioChannelSet getChannelLayout();
//==============================================================================
/** Subclasses must implement this method to perform the low-level read operation.
Callers should use read() instead of calling this directly.
@param destChannels the array of destination buffers to fill. Some of these
pointers may be null
@param numDestChannels the number of items in the destChannels array. This
value is guaranteed not to be greater than the number of
channels that this reader object contains
@param startOffsetInDestBuffer the number of samples from the start of the
dest data at which to begin writing
@param startSampleInFile the number of samples into the source data at which
to begin reading. This value is guaranteed to be >= 0.
@param numSamples the number of samples to read
*/
virtual bool readSamples (int** destChannels,
int numDestChannels,
int startOffsetInDestBuffer,
int64 startSampleInFile,
int numSamples) = 0;
protected:
//==============================================================================
/** Used by AudioFormatReader subclasses to copy data to different formats. */
template <class DestSampleType, class SourceSampleType, class SourceEndianness>
struct ReadHelper
{
using DestType = AudioData::Pointer<DestSampleType, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>;
using SourceType = AudioData::Pointer<SourceSampleType, SourceEndianness, AudioData::Interleaved, AudioData::Const>;
template <typename TargetType>
static void read (TargetType* const* destData, int destOffset, int numDestChannels,
const void* sourceData, int numSourceChannels, int numSamples) noexcept
{
for (int i = 0; i < numDestChannels; ++i)
{
if (void* targetChan = destData[i])
{
DestType dest (targetChan);
dest += destOffset;
if (i < numSourceChannels)
dest.convertSamples (SourceType (addBytesToPointer (sourceData, i * SourceType::getBytesPerSample()), numSourceChannels), numSamples);
else
dest.clearSamples (numSamples);
}
}
}
};
/** Used by AudioFormatReader subclasses to clear any parts of the data blocks that lie
beyond the end of their available length.
*/
static void clearSamplesBeyondAvailableLength (int** destChannels, int numDestChannels,
int startOffsetInDestBuffer, int64 startSampleInFile,
int& numSamples, int64 fileLengthInSamples)
{
jassert (destChannels != nullptr);
const int64 samplesAvailable = fileLengthInSamples - startSampleInFile;
if (samplesAvailable < numSamples)
{
for (int i = numDestChannels; --i >= 0;)
if (destChannels[i] != nullptr)
zeromem (destChannels[i] + startOffsetInDestBuffer, sizeof (int) * (size_t) numSamples);
numSamples = (int) samplesAvailable;
}
}
private:
String formatName;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioFormatReader)
};
} // namespace juce
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