| /* |
| * Copyright (C) 2010 Google Inc. All rights reserved. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions |
| * are met: |
| * |
| * 1. Redistributions of source code must retain the above copyright |
| * notice, this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright |
| * notice, this list of conditions and the following disclaimer in the |
| * documentation and/or other materials provided with the distribution. |
| * 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of |
| * its contributors may be used to endorse or promote products derived |
| * from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY |
| * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
| * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
| * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
| * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
| * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
| * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND |
| * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
| * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF |
| * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #ifndef AudioBus_h |
| #define AudioBus_h |
| |
| #include "platform/audio/AudioChannel.h" |
| #include "wtf/Noncopyable.h" |
| #include "wtf/PassOwnPtr.h" |
| #include "wtf/ThreadSafeRefCounted.h" |
| #include "wtf/Vector.h" |
| |
| namespace WebCore { |
| |
| // An AudioBus represents a collection of one or more AudioChannels. |
| // The data layout is "planar" as opposed to "interleaved". |
| // An AudioBus with one channel is mono, an AudioBus with two channels is stereo, etc. |
| class PLATFORM_EXPORT AudioBus : public ThreadSafeRefCounted<AudioBus> { |
| WTF_MAKE_NONCOPYABLE(AudioBus); |
| public: |
| enum { |
| ChannelLeft = 0, |
| ChannelRight = 1, |
| ChannelCenter = 2, // center and mono are the same |
| ChannelMono = 2, |
| ChannelLFE = 3, |
| ChannelSurroundLeft = 4, |
| ChannelSurroundRight = 5, |
| }; |
| |
| enum { |
| LayoutCanonical = 0 |
| // Can define non-standard layouts here |
| }; |
| |
| enum ChannelInterpretation { |
| Speakers, |
| Discrete, |
| }; |
| |
| // allocate indicates whether or not to initially have the AudioChannels created with managed storage. |
| // Normal usage is to pass true here, in which case the AudioChannels will memory-manage their own storage. |
| // If allocate is false then setChannelMemory() has to be called later on for each channel before the AudioBus is useable... |
| static PassRefPtr<AudioBus> create(unsigned numberOfChannels, size_t length, bool allocate = true); |
| |
| // Tells the given channel to use an externally allocated buffer. |
| void setChannelMemory(unsigned channelIndex, float* storage, size_t length); |
| |
| // Channels |
| unsigned numberOfChannels() const { return m_channels.size(); } |
| |
| AudioChannel* channel(unsigned channel) { return m_channels[channel].get(); } |
| const AudioChannel* channel(unsigned channel) const { return const_cast<AudioBus*>(this)->m_channels[channel].get(); } |
| AudioChannel* channelByType(unsigned type); |
| const AudioChannel* channelByType(unsigned type) const; |
| |
| // Number of sample-frames |
| size_t length() const { return m_length; } |
| |
| // resizeSmaller() can only be called with a new length <= the current length. |
| // The data stored in the bus will remain undisturbed. |
| void resizeSmaller(size_t newLength); |
| |
| // Sample-rate : 0.0 if unknown or "don't care" |
| float sampleRate() const { return m_sampleRate; } |
| void setSampleRate(float sampleRate) { m_sampleRate = sampleRate; } |
| |
| // Zeroes all channels. |
| void zero(); |
| |
| // Clears the silent flag on all channels. |
| void clearSilentFlag(); |
| |
| // Returns true if the silent bit is set on all channels. |
| bool isSilent() const; |
| |
| // Returns true if the channel count and frame-size match. |
| bool topologyMatches(const AudioBus &sourceBus) const; |
| |
| // Creates a new buffer from a range in the source buffer. |
| // 0 may be returned if the range does not fit in the sourceBuffer |
| static PassRefPtr<AudioBus> createBufferFromRange(const AudioBus* sourceBuffer, unsigned startFrame, unsigned endFrame); |
| |
| |
| // Creates a new AudioBus by sample-rate converting sourceBus to the newSampleRate. |
| // setSampleRate() must have been previously called on sourceBus. |
| // Note: sample-rate conversion is already handled in the file-reading code for the mac port, so we don't need this. |
| static PassRefPtr<AudioBus> createBySampleRateConverting(const AudioBus* sourceBus, bool mixToMono, double newSampleRate); |
| |
| // Creates a new AudioBus by mixing all the channels down to mono. |
| // If sourceBus is already mono, then the returned AudioBus will simply be a copy. |
| static PassRefPtr<AudioBus> createByMixingToMono(const AudioBus* sourceBus); |
| |
| // Scales all samples by the same amount. |
| void scale(float scale); |
| |
| void reset() { m_isFirstTime = true; } // for de-zippering |
| |
| // Copies the samples from the source bus to this one. |
| // This is just a simple per-channel copy if the number of channels match, otherwise an up-mix or down-mix is done. |
| void copyFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers); |
| |
| // Sums the samples from the source bus to this one. |
| // This is just a simple per-channel summing if the number of channels match, otherwise an up-mix or down-mix is done. |
| void sumFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers); |
| |
| // Copy each channel from sourceBus into our corresponding channel. |
| // We scale by targetGain (and our own internal gain m_busGain), performing "de-zippering" to smoothly change from *lastMixGain to (targetGain*m_busGain). |
| // The caller is responsible for setting up lastMixGain to point to storage which is unique for every "stream" which will be applied to this bus. |
| // This represents the dezippering memory. |
| void copyWithGainFrom(const AudioBus &sourceBus, float* lastMixGain, float targetGain); |
| |
| // Copies the sourceBus by scaling with sample-accurate gain values. |
| void copyWithSampleAccurateGainValuesFrom(const AudioBus &sourceBus, float* gainValues, unsigned numberOfGainValues); |
| |
| // Returns maximum absolute value across all channels (useful for normalization). |
| float maxAbsValue() const; |
| |
| // Makes maximum absolute value == 1.0 (if possible). |
| void normalize(); |
| |
| static PassRefPtr<AudioBus> loadPlatformResource(const char* name, float sampleRate); |
| |
| protected: |
| AudioBus() { }; |
| |
| AudioBus(unsigned numberOfChannels, size_t length, bool allocate); |
| |
| void speakersCopyFrom(const AudioBus&); |
| void discreteCopyFrom(const AudioBus&); |
| void speakersSumFrom(const AudioBus&); |
| void discreteSumFrom(const AudioBus&); |
| void speakersSumFrom5_1_ToMono(const AudioBus&); |
| |
| size_t m_length; |
| Vector<OwnPtr<AudioChannel> > m_channels; |
| int m_layout; |
| float m_busGain; |
| OwnPtr<AudioFloatArray> m_dezipperGainValues; |
| bool m_isFirstTime; |
| float m_sampleRate; // 0.0 if unknown or N/A |
| }; |
| |
| } // WebCore |
| |
| #endif // AudioBus_h |