mirror of
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276 lines
9.8 KiB
C++
276 lines
9.8 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim: set ts=8 sts=2 et sw=2 tw=80: */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#if !defined(AudioConverter_h)
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# define AudioConverter_h
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# include "MediaInfo.h"
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// Forward declaration
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typedef struct SpeexResamplerState_ SpeexResamplerState;
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namespace mozilla {
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template <AudioConfig::SampleFormat T>
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struct AudioDataBufferTypeChooser;
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template <>
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struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_U8> {
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typedef uint8_t Type;
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};
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template <>
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struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S16> {
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typedef int16_t Type;
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};
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template <>
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struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S24LSB> {
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typedef int32_t Type;
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};
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template <>
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struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S24> {
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typedef int32_t Type;
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};
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template <>
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struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S32> {
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typedef int32_t Type;
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};
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template <>
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struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_FLT> {
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typedef float Type;
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};
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// 'Value' is the type used externally to deal with stored value.
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// AudioDataBuffer can perform conversion between different SampleFormat
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// content.
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template <AudioConfig::SampleFormat Format,
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typename Value = typename AudioDataBufferTypeChooser<Format>::Type>
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class AudioDataBuffer {
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public:
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AudioDataBuffer() = default;
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AudioDataBuffer(Value* aBuffer, size_t aLength) : mBuffer(aBuffer, aLength) {}
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explicit AudioDataBuffer(const AudioDataBuffer& aOther)
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: mBuffer(aOther.mBuffer) {}
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AudioDataBuffer(AudioDataBuffer&& aOther)
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: mBuffer(std::move(aOther.mBuffer)) {}
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template <AudioConfig::SampleFormat OtherFormat, typename OtherValue>
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explicit AudioDataBuffer(
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const AudioDataBuffer<OtherFormat, OtherValue>& other) {
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// TODO: Convert from different type, may use asm routines.
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MOZ_CRASH("Conversion not implemented yet");
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}
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// A u8, s16 and float aligned buffer can only be treated as
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// FORMAT_U8, FORMAT_S16 and FORMAT_FLT respectively.
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// So allow them as copy and move constructors.
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explicit AudioDataBuffer(const AlignedByteBuffer& aBuffer)
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: mBuffer(aBuffer) {
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static_assert(Format == AudioConfig::FORMAT_U8,
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"Conversion not implemented yet");
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}
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explicit AudioDataBuffer(const AlignedShortBuffer& aBuffer)
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: mBuffer(aBuffer) {
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static_assert(Format == AudioConfig::FORMAT_S16,
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"Conversion not implemented yet");
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}
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explicit AudioDataBuffer(const AlignedFloatBuffer& aBuffer)
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: mBuffer(aBuffer) {
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static_assert(Format == AudioConfig::FORMAT_FLT,
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"Conversion not implemented yet");
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}
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explicit AudioDataBuffer(AlignedByteBuffer&& aBuffer)
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: mBuffer(std::move(aBuffer)) {
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static_assert(Format == AudioConfig::FORMAT_U8,
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"Conversion not implemented yet");
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}
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explicit AudioDataBuffer(AlignedShortBuffer&& aBuffer)
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: mBuffer(std::move(aBuffer)) {
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static_assert(Format == AudioConfig::FORMAT_S16,
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"Conversion not implemented yet");
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}
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explicit AudioDataBuffer(AlignedFloatBuffer&& aBuffer)
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: mBuffer(std::move(aBuffer)) {
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static_assert(Format == AudioConfig::FORMAT_FLT,
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"Conversion not implemented yet");
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}
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AudioDataBuffer& operator=(AudioDataBuffer&& aOther) {
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mBuffer = std::move(aOther.mBuffer);
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return *this;
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}
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AudioDataBuffer& operator=(const AudioDataBuffer& aOther) {
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mBuffer = aOther.mBuffer;
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return *this;
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}
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Value* Data() const { return mBuffer.Data(); }
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size_t Length() const { return mBuffer.Length(); }
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size_t Size() const { return mBuffer.Size(); }
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AlignedBuffer<Value> Forget() {
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// Correct type -> Just give values as-is.
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return std::move(mBuffer);
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}
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private:
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AlignedBuffer<Value> mBuffer;
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};
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typedef AudioDataBuffer<AudioConfig::FORMAT_DEFAULT> AudioSampleBuffer;
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class AudioConverter {
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public:
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AudioConverter(const AudioConfig& aIn, const AudioConfig& aOut);
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~AudioConverter();
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// Convert the AudioDataBuffer.
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// Conversion will be done in place if possible. Otherwise a new buffer will
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// be returned.
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// Providing an empty buffer and resampling is expected, the resampler
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// will be drained.
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template <AudioConfig::SampleFormat Format, typename Value>
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AudioDataBuffer<Format, Value> Process(
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AudioDataBuffer<Format, Value>&& aBuffer) {
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MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format() &&
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mIn.Format() == Format);
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AudioDataBuffer<Format, Value> buffer = std::move(aBuffer);
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if (CanWorkInPlace()) {
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AlignedBuffer<Value> temp = buffer.Forget();
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Process(temp, temp.Data(), SamplesInToFrames(temp.Length()));
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return AudioDataBuffer<Format, Value>(std::move(temp));
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}
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return Process(buffer);
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}
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template <AudioConfig::SampleFormat Format, typename Value>
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AudioDataBuffer<Format, Value> Process(
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const AudioDataBuffer<Format, Value>& aBuffer) {
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MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format() &&
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mIn.Format() == Format);
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// Perform the downmixing / reordering in temporary buffer.
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size_t frames = SamplesInToFrames(aBuffer.Length());
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AlignedBuffer<Value> temp1;
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if (!temp1.SetLength(FramesOutToSamples(frames))) {
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return AudioDataBuffer<Format, Value>(std::move(temp1));
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}
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frames = ProcessInternal(temp1.Data(), aBuffer.Data(), frames);
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if (mIn.Rate() == mOut.Rate()) {
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MOZ_ALWAYS_TRUE(temp1.SetLength(FramesOutToSamples(frames)));
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return AudioDataBuffer<Format, Value>(std::move(temp1));
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}
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// At this point, temp1 contains the buffer reordered and downmixed.
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// If we are downsampling we can re-use it.
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AlignedBuffer<Value>* outputBuffer = &temp1;
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AlignedBuffer<Value> temp2;
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if (!frames || mOut.Rate() > mIn.Rate()) {
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// We are upsampling or about to drain, we can't work in place.
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// Allocate another temporary buffer where the upsampling will occur.
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if (!temp2.SetLength(
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FramesOutToSamples(ResampleRecipientFrames(frames)))) {
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return AudioDataBuffer<Format, Value>(std::move(temp2));
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}
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outputBuffer = &temp2;
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}
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if (!frames) {
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frames = DrainResampler(outputBuffer->Data());
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} else {
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frames = ResampleAudio(outputBuffer->Data(), temp1.Data(), frames);
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}
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MOZ_ALWAYS_TRUE(outputBuffer->SetLength(FramesOutToSamples(frames)));
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return AudioDataBuffer<Format, Value>(std::move(*outputBuffer));
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}
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// Attempt to convert the AudioDataBuffer in place.
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// Will return 0 if the conversion wasn't possible.
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template <typename Value>
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size_t Process(Value* aBuffer, size_t aFrames) {
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MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format());
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if (!CanWorkInPlace()) {
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return 0;
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}
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size_t frames = ProcessInternal(aBuffer, aBuffer, aFrames);
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if (frames && mIn.Rate() != mOut.Rate()) {
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frames = ResampleAudio(aBuffer, aBuffer, aFrames);
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}
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return frames;
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}
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template <typename Value>
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size_t Process(AlignedBuffer<Value>& aOutBuffer, const Value* aInBuffer,
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size_t aFrames) {
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MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format());
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MOZ_ASSERT((aFrames && aInBuffer) || !aFrames);
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// Up/down mixing first
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if (!aOutBuffer.SetLength(FramesOutToSamples(aFrames))) {
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MOZ_ALWAYS_TRUE(aOutBuffer.SetLength(0));
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return 0;
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}
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size_t frames = ProcessInternal(aOutBuffer.Data(), aInBuffer, aFrames);
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MOZ_ASSERT(frames == aFrames);
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// Check if resampling is needed
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if (mIn.Rate() == mOut.Rate()) {
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return frames;
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}
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// Prepare output in cases of drain or up-sampling
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if ((!frames || mOut.Rate() > mIn.Rate()) &&
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!aOutBuffer.SetLength(
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FramesOutToSamples(ResampleRecipientFrames(frames)))) {
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MOZ_ALWAYS_TRUE(aOutBuffer.SetLength(0));
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return 0;
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}
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if (!frames) {
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frames = DrainResampler(aOutBuffer.Data());
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} else {
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frames = ResampleAudio(aOutBuffer.Data(), aInBuffer, frames);
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}
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// Update with the actual buffer length
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MOZ_ALWAYS_TRUE(aOutBuffer.SetLength(FramesOutToSamples(frames)));
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return frames;
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}
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bool CanWorkInPlace() const;
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bool CanReorderAudio() const {
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return mIn.Layout().MappingTable(mOut.Layout());
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}
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static bool CanConvert(const AudioConfig& aIn, const AudioConfig& aOut);
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const AudioConfig& InputConfig() const { return mIn; }
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const AudioConfig& OutputConfig() const { return mOut; }
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private:
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const AudioConfig mIn;
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const AudioConfig mOut;
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// mChannelOrderMap will be empty if we do not know how to proceed with this
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// channel layout.
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AutoTArray<uint8_t, AudioConfig::ChannelLayout::MAX_CHANNELS>
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mChannelOrderMap;
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/**
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* ProcessInternal
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* Parameters:
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* aOut : destination buffer where converted samples will be copied
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* aIn : source buffer
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* aSamples: number of frames in source buffer
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*
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* Return Value: number of frames converted or 0 if error
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*/
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size_t ProcessInternal(void* aOut, const void* aIn, size_t aFrames);
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void ReOrderInterleavedChannels(void* aOut, const void* aIn,
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size_t aFrames) const;
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size_t DownmixAudio(void* aOut, const void* aIn, size_t aFrames) const;
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size_t UpmixAudio(void* aOut, const void* aIn, size_t aFrames) const;
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size_t FramesOutToSamples(size_t aFrames) const;
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size_t SamplesInToFrames(size_t aSamples) const;
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size_t FramesOutToBytes(size_t aFrames) const;
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// Resampler context.
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SpeexResamplerState* mResampler;
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size_t ResampleAudio(void* aOut, const void* aIn, size_t aFrames);
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size_t ResampleRecipientFrames(size_t aFrames) const;
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void RecreateResampler();
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size_t DrainResampler(void* aOut);
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};
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} // namespace mozilla
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#endif /* AudioConverter_h */
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