mirror of
https://github.com/mozilla/gecko-dev.git
synced 2024-12-01 00:32:11 +00:00
278 lines
9.8 KiB
C++
278 lines
9.8 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
|
|
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
|
|
/* 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/. */
|
|
|
|
#if !defined(AudioConverter_h)
|
|
# define AudioConverter_h
|
|
|
|
# include "MediaInfo.h"
|
|
|
|
// Forward declaration
|
|
typedef struct SpeexResamplerState_ SpeexResamplerState;
|
|
|
|
namespace mozilla {
|
|
|
|
template <AudioConfig::SampleFormat T>
|
|
struct AudioDataBufferTypeChooser;
|
|
template <>
|
|
struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_U8> {
|
|
typedef uint8_t Type;
|
|
};
|
|
template <>
|
|
struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S16> {
|
|
typedef int16_t Type;
|
|
};
|
|
template <>
|
|
struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S24LSB> {
|
|
typedef int32_t Type;
|
|
};
|
|
template <>
|
|
struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S24> {
|
|
typedef int32_t Type;
|
|
};
|
|
template <>
|
|
struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S32> {
|
|
typedef int32_t Type;
|
|
};
|
|
template <>
|
|
struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_FLT> {
|
|
typedef float Type;
|
|
};
|
|
|
|
// 'Value' is the type used externally to deal with stored value.
|
|
// AudioDataBuffer can perform conversion between different SampleFormat
|
|
// content.
|
|
template <AudioConfig::SampleFormat Format,
|
|
typename Value = typename AudioDataBufferTypeChooser<Format>::Type>
|
|
class AudioDataBuffer {
|
|
public:
|
|
AudioDataBuffer() = default;
|
|
AudioDataBuffer(Value* aBuffer, size_t aLength) : mBuffer(aBuffer, aLength) {}
|
|
explicit AudioDataBuffer(const AudioDataBuffer& aOther)
|
|
: mBuffer(aOther.mBuffer) {}
|
|
AudioDataBuffer(AudioDataBuffer&& aOther)
|
|
: mBuffer(std::move(aOther.mBuffer)) {}
|
|
template <AudioConfig::SampleFormat OtherFormat, typename OtherValue>
|
|
explicit AudioDataBuffer(
|
|
const AudioDataBuffer<OtherFormat, OtherValue>& other) {
|
|
// TODO: Convert from different type, may use asm routines.
|
|
MOZ_CRASH("Conversion not implemented yet");
|
|
}
|
|
|
|
// A u8, s16 and float aligned buffer can only be treated as
|
|
// FORMAT_U8, FORMAT_S16 and FORMAT_FLT respectively.
|
|
// So allow them as copy and move constructors.
|
|
explicit AudioDataBuffer(const AlignedByteBuffer& aBuffer)
|
|
: mBuffer(aBuffer) {
|
|
static_assert(Format == AudioConfig::FORMAT_U8,
|
|
"Conversion not implemented yet");
|
|
}
|
|
explicit AudioDataBuffer(const AlignedShortBuffer& aBuffer)
|
|
: mBuffer(aBuffer) {
|
|
static_assert(Format == AudioConfig::FORMAT_S16,
|
|
"Conversion not implemented yet");
|
|
}
|
|
explicit AudioDataBuffer(const AlignedFloatBuffer& aBuffer)
|
|
: mBuffer(aBuffer) {
|
|
static_assert(Format == AudioConfig::FORMAT_FLT,
|
|
"Conversion not implemented yet");
|
|
}
|
|
explicit AudioDataBuffer(AlignedByteBuffer&& aBuffer)
|
|
: mBuffer(std::move(aBuffer)) {
|
|
static_assert(Format == AudioConfig::FORMAT_U8,
|
|
"Conversion not implemented yet");
|
|
}
|
|
explicit AudioDataBuffer(AlignedShortBuffer&& aBuffer)
|
|
: mBuffer(std::move(aBuffer)) {
|
|
static_assert(Format == AudioConfig::FORMAT_S16,
|
|
"Conversion not implemented yet");
|
|
}
|
|
explicit AudioDataBuffer(AlignedFloatBuffer&& aBuffer)
|
|
: mBuffer(std::move(aBuffer)) {
|
|
static_assert(Format == AudioConfig::FORMAT_FLT,
|
|
"Conversion not implemented yet");
|
|
}
|
|
AudioDataBuffer& operator=(AudioDataBuffer&& aOther) {
|
|
mBuffer = std::move(aOther.mBuffer);
|
|
return *this;
|
|
}
|
|
AudioDataBuffer& operator=(const AudioDataBuffer& aOther) {
|
|
mBuffer = aOther.mBuffer;
|
|
return *this;
|
|
}
|
|
|
|
Value* Data() const { return mBuffer.Data(); }
|
|
size_t Length() const { return mBuffer.Length(); }
|
|
size_t Size() const { return mBuffer.Size(); }
|
|
AlignedBuffer<Value> Forget() {
|
|
// Correct type -> Just give values as-is.
|
|
return std::move(mBuffer);
|
|
}
|
|
|
|
private:
|
|
AlignedBuffer<Value> mBuffer;
|
|
};
|
|
|
|
typedef AudioDataBuffer<AudioConfig::FORMAT_DEFAULT> AudioSampleBuffer;
|
|
|
|
class AudioConverter {
|
|
public:
|
|
AudioConverter(const AudioConfig& aIn, const AudioConfig& aOut);
|
|
~AudioConverter();
|
|
|
|
// Convert the AudioDataBuffer.
|
|
// Conversion will be done in place if possible. Otherwise a new buffer will
|
|
// be returned.
|
|
// Providing an empty buffer and resampling is expected, the resampler
|
|
// will be drained.
|
|
template <AudioConfig::SampleFormat Format, typename Value>
|
|
AudioDataBuffer<Format, Value> Process(
|
|
AudioDataBuffer<Format, Value>&& aBuffer) {
|
|
MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format() &&
|
|
mIn.Format() == Format);
|
|
AudioDataBuffer<Format, Value> buffer = std::move(aBuffer);
|
|
if (CanWorkInPlace()) {
|
|
AlignedBuffer<Value> temp = buffer.Forget();
|
|
Process(temp, temp.Data(), SamplesInToFrames(temp.Length()));
|
|
return AudioDataBuffer<Format, Value>(std::move(temp));
|
|
;
|
|
}
|
|
return Process(buffer);
|
|
}
|
|
|
|
template <AudioConfig::SampleFormat Format, typename Value>
|
|
AudioDataBuffer<Format, Value> Process(
|
|
const AudioDataBuffer<Format, Value>& aBuffer) {
|
|
MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format() &&
|
|
mIn.Format() == Format);
|
|
// Perform the downmixing / reordering in temporary buffer.
|
|
size_t frames = SamplesInToFrames(aBuffer.Length());
|
|
AlignedBuffer<Value> temp1;
|
|
if (!temp1.SetLength(FramesOutToSamples(frames))) {
|
|
return AudioDataBuffer<Format, Value>(std::move(temp1));
|
|
}
|
|
frames = ProcessInternal(temp1.Data(), aBuffer.Data(), frames);
|
|
if (mIn.Rate() == mOut.Rate()) {
|
|
MOZ_ALWAYS_TRUE(temp1.SetLength(FramesOutToSamples(frames)));
|
|
return AudioDataBuffer<Format, Value>(std::move(temp1));
|
|
}
|
|
|
|
// At this point, temp1 contains the buffer reordered and downmixed.
|
|
// If we are downsampling we can re-use it.
|
|
AlignedBuffer<Value>* outputBuffer = &temp1;
|
|
AlignedBuffer<Value> temp2;
|
|
if (!frames || mOut.Rate() > mIn.Rate()) {
|
|
// We are upsampling or about to drain, we can't work in place.
|
|
// Allocate another temporary buffer where the upsampling will occur.
|
|
if (!temp2.SetLength(
|
|
FramesOutToSamples(ResampleRecipientFrames(frames)))) {
|
|
return AudioDataBuffer<Format, Value>(std::move(temp2));
|
|
}
|
|
outputBuffer = &temp2;
|
|
}
|
|
if (!frames) {
|
|
frames = DrainResampler(outputBuffer->Data());
|
|
} else {
|
|
frames = ResampleAudio(outputBuffer->Data(), temp1.Data(), frames);
|
|
}
|
|
MOZ_ALWAYS_TRUE(outputBuffer->SetLength(FramesOutToSamples(frames)));
|
|
return AudioDataBuffer<Format, Value>(std::move(*outputBuffer));
|
|
}
|
|
|
|
// Attempt to convert the AudioDataBuffer in place.
|
|
// Will return 0 if the conversion wasn't possible.
|
|
template <typename Value>
|
|
size_t Process(Value* aBuffer, size_t aFrames) {
|
|
MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format());
|
|
if (!CanWorkInPlace()) {
|
|
return 0;
|
|
}
|
|
size_t frames = ProcessInternal(aBuffer, aBuffer, aFrames);
|
|
if (frames && mIn.Rate() != mOut.Rate()) {
|
|
frames = ResampleAudio(aBuffer, aBuffer, aFrames);
|
|
}
|
|
return frames;
|
|
}
|
|
|
|
template <typename Value>
|
|
size_t Process(AlignedBuffer<Value>& aOutBuffer, const Value* aInBuffer,
|
|
size_t aFrames) {
|
|
MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format());
|
|
MOZ_ASSERT((aFrames && aInBuffer) || !aFrames);
|
|
// Up/down mixing first
|
|
if (!aOutBuffer.SetLength(FramesOutToSamples(aFrames))) {
|
|
MOZ_ALWAYS_TRUE(aOutBuffer.SetLength(0));
|
|
return 0;
|
|
}
|
|
size_t frames = ProcessInternal(aOutBuffer.Data(), aInBuffer, aFrames);
|
|
MOZ_ASSERT(frames == aFrames);
|
|
// Check if resampling is needed
|
|
if (mIn.Rate() == mOut.Rate()) {
|
|
return frames;
|
|
}
|
|
// Prepare output in cases of drain or up-sampling
|
|
if ((!frames || mOut.Rate() > mIn.Rate()) &&
|
|
!aOutBuffer.SetLength(
|
|
FramesOutToSamples(ResampleRecipientFrames(frames)))) {
|
|
MOZ_ALWAYS_TRUE(aOutBuffer.SetLength(0));
|
|
return 0;
|
|
}
|
|
if (!frames) {
|
|
frames = DrainResampler(aOutBuffer.Data());
|
|
} else {
|
|
frames = ResampleAudio(aOutBuffer.Data(), aInBuffer, frames);
|
|
}
|
|
// Update with the actual buffer length
|
|
MOZ_ALWAYS_TRUE(aOutBuffer.SetLength(FramesOutToSamples(frames)));
|
|
return frames;
|
|
}
|
|
|
|
bool CanWorkInPlace() const;
|
|
bool CanReorderAudio() const {
|
|
return mIn.Layout().MappingTable(mOut.Layout());
|
|
}
|
|
static bool CanConvert(const AudioConfig& aIn, const AudioConfig& aOut);
|
|
|
|
const AudioConfig& InputConfig() const { return mIn; }
|
|
const AudioConfig& OutputConfig() const { return mOut; }
|
|
|
|
private:
|
|
const AudioConfig mIn;
|
|
const AudioConfig mOut;
|
|
// mChannelOrderMap will be empty if we do not know how to proceed with this
|
|
// channel layout.
|
|
AutoTArray<uint8_t, AudioConfig::ChannelLayout::MAX_CHANNELS>
|
|
mChannelOrderMap;
|
|
/**
|
|
* ProcessInternal
|
|
* Parameters:
|
|
* aOut : destination buffer where converted samples will be copied
|
|
* aIn : source buffer
|
|
* aSamples: number of frames in source buffer
|
|
*
|
|
* Return Value: number of frames converted or 0 if error
|
|
*/
|
|
size_t ProcessInternal(void* aOut, const void* aIn, size_t aFrames);
|
|
void ReOrderInterleavedChannels(void* aOut, const void* aIn,
|
|
size_t aFrames) const;
|
|
size_t DownmixAudio(void* aOut, const void* aIn, size_t aFrames) const;
|
|
size_t UpmixAudio(void* aOut, const void* aIn, size_t aFrames) const;
|
|
|
|
size_t FramesOutToSamples(size_t aFrames) const;
|
|
size_t SamplesInToFrames(size_t aSamples) const;
|
|
size_t FramesOutToBytes(size_t aFrames) const;
|
|
|
|
// Resampler context.
|
|
SpeexResamplerState* mResampler;
|
|
size_t ResampleAudio(void* aOut, const void* aIn, size_t aFrames);
|
|
size_t ResampleRecipientFrames(size_t aFrames) const;
|
|
void RecreateResampler();
|
|
size_t DrainResampler(void* aOut);
|
|
};
|
|
|
|
} // namespace mozilla
|
|
|
|
#endif /* AudioConverter_h */
|