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LibAudio+LibDSP: Switch samples to 32-bit float instead of 64-bit float
This has been overkill from the start, and it has been bugging me for a long time. With this change, we're probably a bit slower on most platforms but save huge amounts of space with all in-memory sample datastructures.
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16 changed files with 329 additions and 329 deletions
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@ -134,7 +134,7 @@ MaybeLoaderError FlacLoaderPlugin::parse_header()
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++total_meta_blocks;
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}
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dbgln_if(AFLACLOADER_DEBUG, "Parsed FLAC header: blocksize {}-{}{}, framesize {}-{}, {}Hz, {}bit, {} channels, {} samples total ({:.2f}s), MD5 {}, data start at {:x} bytes, {} headers total (skipped {})", m_min_block_size, m_max_block_size, is_fixed_blocksize_stream() ? " (constant)" : "", m_min_frame_size, m_max_frame_size, m_sample_rate, pcm_bits_per_sample(m_sample_format), m_num_channels, m_total_samples, static_cast<double>(m_total_samples) / static_cast<double>(m_sample_rate), md5_checksum, m_data_start_location, total_meta_blocks, total_meta_blocks - meta_blocks_parsed);
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dbgln_if(AFLACLOADER_DEBUG, "Parsed FLAC header: blocksize {}-{}{}, framesize {}-{}, {}Hz, {}bit, {} channels, {} samples total ({:.2f}s), MD5 {}, data start at {:x} bytes, {} headers total (skipped {})", m_min_block_size, m_max_block_size, is_fixed_blocksize_stream() ? " (constant)" : "", m_min_frame_size, m_max_frame_size, m_sample_rate, pcm_bits_per_sample(m_sample_format), m_num_channels, m_total_samples, static_cast<float>(m_total_samples) / static_cast<float>(m_sample_rate), md5_checksum, m_data_start_location, total_meta_blocks, total_meta_blocks - meta_blocks_parsed);
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return {};
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}
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@ -402,7 +402,7 @@ MaybeLoaderError FlacLoaderPlugin::next_frame(Span<Sample> target_vector)
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VERIFY(left.size() == right.size() && left.size() == m_current_frame->sample_count);
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double sample_rescale = static_cast<double>(1 << (pcm_bits_per_sample(m_current_frame->bit_depth) - 1));
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float sample_rescale = static_cast<float>(1 << (pcm_bits_per_sample(m_current_frame->bit_depth) - 1));
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dbgln_if(AFLACLOADER_DEBUG, "Sample rescaled from {} bits: factor {:.1f}", pcm_bits_per_sample(m_current_frame->bit_depth), sample_rescale);
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// zip together channels
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