forked from mia/Aegisub
71894fd769
There was some magic bit operations to calculate the cache block offsets. This only works when both bytes_per_sample and channels are power of 2. Originally the format is assumed to be int16 mono, which satisfies this requirement. However in case we use original audio data, the channels can be something not a power of 2 (e.g. for 5.1 channel audio the number of channels is 6). This will break the calculation. We rewrite the calculation, without using those bit operations.
98 lines
3.2 KiB
C++
98 lines
3.2 KiB
C++
// Copyright (c) 2014, Thomas Goyne <plorkyeran@aegisub.org>
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//
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// Permission to use, copy, modify, and distribute this software for any
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// purpose with or without fee is hereby granted, provided that the above
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// copyright notice and this permission notice appear in all copies.
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//
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// THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
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// WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
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// MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
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// ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
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// WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
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// ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
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// OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
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//
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// Aegisub Project http://www.aegisub.org/
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#include "libaegisub/audio/provider.h"
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#include "libaegisub/make_unique.h"
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#include <array>
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#include <boost/container/stable_vector.hpp>
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#include <thread>
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namespace {
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using namespace agi;
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#define CacheBits 22
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#define CacheBlockSize (1 << CacheBits)
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class RAMAudioProvider final : public AudioProviderWrapper {
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#ifdef _MSC_VER
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boost::container::stable_vector<char[CacheBlockSize]> blockcache;
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#else
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boost::container::stable_vector<std::array<char, CacheBlockSize>> blockcache;
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#endif
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std::atomic<bool> cancelled = {false};
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std::thread decoder;
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void FillBuffer(void *buf, int64_t start, int64_t count) const override;
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public:
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RAMAudioProvider(std::unique_ptr<AudioProvider> src)
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: AudioProviderWrapper(std::move(src))
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{
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decoded_samples = 0;
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try {
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blockcache.resize((num_samples * bytes_per_sample * channels + CacheBlockSize - 1) >> CacheBits);
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}
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catch (std::bad_alloc const&) {
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throw AudioProviderError("Not enough memory available to cache in RAM");
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}
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decoder = std::thread([&] {
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int64_t readsize = CacheBlockSize / bytes_per_sample / channels;
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for (size_t i = 0; i < blockcache.size(); i++) {
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if (cancelled) break;
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auto actual_read = std::min<int64_t>(readsize, num_samples - i * readsize);
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source->GetAudio(&blockcache[i][0], i * readsize, actual_read);
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decoded_samples += actual_read;
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}
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});
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}
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~RAMAudioProvider() {
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cancelled = true;
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decoder.join();
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}
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};
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void RAMAudioProvider::FillBuffer(void *buf, int64_t start, int64_t count) const {
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auto charbuf = static_cast<char *>(buf);
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for (int64_t bytes_remaining = count * bytes_per_sample * channels; bytes_remaining; ) {
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if (start >= decoded_samples) {
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memset(charbuf, 0, bytes_remaining);
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break;
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}
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const int64_t samples_per_block = CacheBlockSize / bytes_per_sample / channels;
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const size_t i = start / samples_per_block;
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const int start_offset = (start % samples_per_block) * bytes_per_sample * channels;
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const int read_size = std::min<int>(bytes_remaining, samples_per_block * bytes_per_sample * channels - start_offset);
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memcpy(charbuf, &blockcache[i][start_offset], read_size);
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charbuf += read_size;
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bytes_remaining -= read_size;
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start += read_size / bytes_per_sample / channels;
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}
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}
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}
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namespace agi {
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std::unique_ptr<AudioProvider> CreateRAMAudioProvider(std::unique_ptr<AudioProvider> src) {
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return agi::make_unique<RAMAudioProvider>(std::move(src));
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}
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}
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