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bf129486d9
Additionally, indicate in clearer fashion where setting length can't fail. MozReview-Commit-ID: LSZtCclqhK1 --HG-- extra : rebase_source : 2dd254649809bc534c1dfc78e33ce3f46fae323c
241 lines
8.4 KiB
C++
241 lines
8.4 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> struct AudioDataBufferTypeChooser;
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template <> struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_U8>
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{ typedef uint8_t Type; };
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template <> struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S16>
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{ typedef int16_t Type; };
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template <> struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S24LSB>
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{ typedef int32_t Type; };
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template <> struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S24>
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{ typedef int32_t Type; };
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template <> struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_S32>
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{ typedef int32_t Type; };
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template <> struct AudioDataBufferTypeChooser<AudioConfig::FORMAT_FLT>
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{ typedef float Type; };
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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 content.
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template <AudioConfig::SampleFormat Format, typename Value = typename AudioDataBufferTypeChooser<Format>::Type>
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class AudioDataBuffer
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{
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public:
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AudioDataBuffer() {}
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AudioDataBuffer(Value* aBuffer, size_t aLength)
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: mBuffer(aBuffer, aLength)
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{}
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explicit AudioDataBuffer(const AudioDataBuffer& aOther)
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: mBuffer(aOther.mBuffer)
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{}
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AudioDataBuffer(AudioDataBuffer&& aOther)
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: mBuffer(Move(aOther.mBuffer))
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{}
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template <AudioConfig::SampleFormat OtherFormat, typename OtherValue>
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explicit AudioDataBuffer(const AudioDataBuffer<OtherFormat, OtherValue>& other)
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{
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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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{
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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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{
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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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{
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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(Move(aBuffer))
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{
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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(Move(aBuffer))
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{
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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(Move(aBuffer))
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{
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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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{
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mBuffer = 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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{
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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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{
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// Correct type -> Just give values as-is.
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return 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(AudioDataBuffer<Format, Value>&& aBuffer)
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{
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MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format() && mIn.Format() == Format);
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AudioDataBuffer<Format, Value> buffer = Move(aBuffer);
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if (CanWorkInPlace()) {
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size_t frames = SamplesInToFrames(buffer.Length());
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frames = ProcessInternal(buffer.Data(), buffer.Data(), frames);
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if (frames && mIn.Rate() != mOut.Rate()) {
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frames = ResampleAudio(buffer.Data(), buffer.Data(), frames);
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}
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AlignedBuffer<Value> temp = buffer.Forget();
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temp.SetLength(FramesOutToSamples(frames));
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return AudioDataBuffer<Format, Value>(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(const AudioDataBuffer<Format, Value>& aBuffer)
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{
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MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == mOut.Format() && 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>(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>(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(FramesOutToSamples(ResampleRecipientFrames(frames)))) {
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return AudioDataBuffer<Format, Value>(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>(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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{
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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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bool CanWorkInPlace() const;
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bool CanReorderAudio() const
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{
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return mIn.Layout().MappingTable(mOut.Layout());
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}
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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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uint8_t mChannelOrderMap[MAX_AUDIO_CHANNELS];
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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, 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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