Fri, 16 Jan 2015 04:50:19 +0100
Replace accessor implementation with direct member state manipulation, by
request https://trac.torproject.org/projects/tor/ticket/9701#comment:32
michael@0 | 1 | /* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ |
michael@0 | 2 | /* vim: set ts=8 sts=2 et sw=2 tw=80: */ |
michael@0 | 3 | /* This Source Code Form is subject to the terms of the Mozilla Public |
michael@0 | 4 | * License, v. 2.0. If a copy of the MPL was not distributed with this |
michael@0 | 5 | * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ |
michael@0 | 6 | #if !defined(AudioCompactor_h) |
michael@0 | 7 | #define AudioCompactor_h |
michael@0 | 8 | |
michael@0 | 9 | #include "MediaQueue.h" |
michael@0 | 10 | #include "MediaData.h" |
michael@0 | 11 | #include "VideoUtils.h" |
michael@0 | 12 | |
michael@0 | 13 | namespace mozilla { |
michael@0 | 14 | |
michael@0 | 15 | class AudioCompactor |
michael@0 | 16 | { |
michael@0 | 17 | public: |
michael@0 | 18 | AudioCompactor(MediaQueue<AudioData>& aQueue) |
michael@0 | 19 | : mQueue(aQueue) |
michael@0 | 20 | { } |
michael@0 | 21 | |
michael@0 | 22 | // Push audio data into the underlying queue with minimal heap allocation |
michael@0 | 23 | // slop. This method is responsible for allocating AudioDataValue[] buffers. |
michael@0 | 24 | // The caller must provide a functor to copy the data into the buffers. The |
michael@0 | 25 | // functor must provide the following signature: |
michael@0 | 26 | // |
michael@0 | 27 | // uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples); |
michael@0 | 28 | // |
michael@0 | 29 | // The functor must copy as many complete frames as possible to the provided |
michael@0 | 30 | // buffer given its length (in AudioDataValue elements). The number of frames |
michael@0 | 31 | // copied must be returned. This copy functor must support being called |
michael@0 | 32 | // multiple times in order to copy the audio data fully. The copy functor |
michael@0 | 33 | // must copy full frames as partial frames will be ignored. |
michael@0 | 34 | template<typename CopyFunc> |
michael@0 | 35 | bool Push(int64_t aOffset, int64_t aTime, int32_t aSampleRate, |
michael@0 | 36 | uint32_t aFrames, uint32_t aChannels, CopyFunc aCopyFunc) |
michael@0 | 37 | { |
michael@0 | 38 | // If we are losing more than a reasonable amount to padding, try to chunk |
michael@0 | 39 | // the data. |
michael@0 | 40 | size_t maxSlop = AudioDataSize(aFrames, aChannels) / MAX_SLOP_DIVISOR; |
michael@0 | 41 | |
michael@0 | 42 | while (aFrames > 0) { |
michael@0 | 43 | uint32_t samples = GetChunkSamples(aFrames, aChannels, maxSlop); |
michael@0 | 44 | nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[samples]); |
michael@0 | 45 | |
michael@0 | 46 | // Copy audio data to buffer using caller-provided functor. |
michael@0 | 47 | uint32_t framesCopied = aCopyFunc(buffer, samples); |
michael@0 | 48 | |
michael@0 | 49 | NS_ASSERTION(framesCopied <= aFrames, "functor copied too many frames"); |
michael@0 | 50 | |
michael@0 | 51 | CheckedInt64 duration = FramesToUsecs(framesCopied, aSampleRate); |
michael@0 | 52 | if (!duration.isValid()) { |
michael@0 | 53 | return false; |
michael@0 | 54 | } |
michael@0 | 55 | |
michael@0 | 56 | mQueue.Push(new AudioData(aOffset, |
michael@0 | 57 | aTime, |
michael@0 | 58 | duration.value(), |
michael@0 | 59 | framesCopied, |
michael@0 | 60 | buffer.forget(), |
michael@0 | 61 | aChannels)); |
michael@0 | 62 | |
michael@0 | 63 | // Remove the frames we just pushed into the queue and loop if there is |
michael@0 | 64 | // more to be done. |
michael@0 | 65 | aTime += duration.value(); |
michael@0 | 66 | aFrames -= framesCopied; |
michael@0 | 67 | |
michael@0 | 68 | // NOTE: No need to update aOffset as its only an approximation anyway. |
michael@0 | 69 | } |
michael@0 | 70 | |
michael@0 | 71 | return true; |
michael@0 | 72 | } |
michael@0 | 73 | |
michael@0 | 74 | // Copy functor suitable for copying audio samples already in the |
michael@0 | 75 | // AudioDataValue format/layout expected by AudioStream on this platform. |
michael@0 | 76 | class NativeCopy |
michael@0 | 77 | { |
michael@0 | 78 | public: |
michael@0 | 79 | NativeCopy(const uint8_t* aSource, size_t aSourceBytes, |
michael@0 | 80 | uint32_t aChannels) |
michael@0 | 81 | : mSource(aSource) |
michael@0 | 82 | , mSourceBytes(aSourceBytes) |
michael@0 | 83 | , mChannels(aChannels) |
michael@0 | 84 | , mNextByte(0) |
michael@0 | 85 | { } |
michael@0 | 86 | |
michael@0 | 87 | uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples); |
michael@0 | 88 | |
michael@0 | 89 | private: |
michael@0 | 90 | const uint8_t* const mSource; |
michael@0 | 91 | const size_t mSourceBytes; |
michael@0 | 92 | const uint32_t mChannels; |
michael@0 | 93 | size_t mNextByte; |
michael@0 | 94 | }; |
michael@0 | 95 | |
michael@0 | 96 | // Allow 12.5% slop before chunking kicks in. Public so that the gtest can |
michael@0 | 97 | // access it. |
michael@0 | 98 | static const size_t MAX_SLOP_DIVISOR = 8; |
michael@0 | 99 | |
michael@0 | 100 | private: |
michael@0 | 101 | // Compute the number of AudioDataValue samples that will be fit the most |
michael@0 | 102 | // frames while keeping heap allocation slop less than the given threshold. |
michael@0 | 103 | static uint32_t |
michael@0 | 104 | GetChunkSamples(uint32_t aFrames, uint32_t aChannels, size_t aMaxSlop); |
michael@0 | 105 | |
michael@0 | 106 | static size_t BytesPerFrame(uint32_t aChannels) |
michael@0 | 107 | { |
michael@0 | 108 | return sizeof(AudioDataValue) * aChannels; |
michael@0 | 109 | } |
michael@0 | 110 | |
michael@0 | 111 | static size_t AudioDataSize(uint32_t aFrames, uint32_t aChannels) |
michael@0 | 112 | { |
michael@0 | 113 | return aFrames * BytesPerFrame(aChannels); |
michael@0 | 114 | } |
michael@0 | 115 | |
michael@0 | 116 | MediaQueue<AudioData> &mQueue; |
michael@0 | 117 | }; |
michael@0 | 118 | |
michael@0 | 119 | } // namespace mozilla |
michael@0 | 120 | |
michael@0 | 121 | #endif // AudioCompactor_h |