/* -*- 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 */
