mirror of
https://github.com/RGBCube/serenity
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9a2c80c791
This commit addresses two issues: 1. If you play a 96 KHz Wave file, the slider position is incorrect, because it is assumed all files are 44.1 KHz. 2. For high-bitrate files, there are audio dropouts due to not buffering enough audio data. Issue 1 is addressed by scaling the number of played samples by the ratio between the source and destination sample rates. Issue 2 is addressed by buffering a certain number of milliseconds worth of audio data (instead of a fixed number of bytes). This makes the the buffer size independent of the source sample rate. Some of the code is redesigned to be simpler. The code that did the book-keeping of which buffers need to be loaded and which have been already played has been removed. Instead, we enqueue a new buffer based on a low watermark of samples remaining in the audio server queue. Other small fixes include: 1. Disable the stop button when playback is finished. 2. Remove hard-coded instances of 44100. 3. Update the GUI every 50 ms (was 100), which improves visualizations.
87 lines
2.8 KiB
C++
87 lines
2.8 KiB
C++
/*
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* Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
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* Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>
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*
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* SPDX-License-Identifier: BSD-2-Clause
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*/
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#pragma once
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#include <AK/ByteBuffer.h>
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#include <AK/MemoryStream.h>
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#include <AK/OwnPtr.h>
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#include <AK/RefPtr.h>
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#include <AK/Stream.h>
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#include <AK/String.h>
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#include <AK/StringView.h>
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#include <AK/WeakPtr.h>
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#include <LibAudio/Buffer.h>
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#include <LibAudio/Loader.h>
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#include <LibCore/File.h>
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#include <LibCore/FileStream.h>
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namespace Audio {
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class Buffer;
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// defines for handling the WAV header data
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#define WAVE_FORMAT_PCM 0x0001 // PCM
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#define WAVE_FORMAT_IEEE_FLOAT 0x0003 // IEEE float
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#define WAVE_FORMAT_ALAW 0x0006 // 8-bit ITU-T G.711 A-law
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#define WAVE_FORMAT_MULAW 0x0007 // 8-bit ITU-T G.711 µ-law
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#define WAVE_FORMAT_EXTENSIBLE 0xFFFE // Determined by SubFormat
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// Parses a WAV file and produces an Audio::Buffer.
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class WavLoaderPlugin : public LoaderPlugin {
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public:
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WavLoaderPlugin(const StringView& path);
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WavLoaderPlugin(const ByteBuffer& buffer);
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virtual bool sniff() override { return valid; }
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virtual bool has_error() override { return !m_error_string.is_null(); }
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virtual const char* error_string() override { return m_error_string.characters(); }
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// The Buffer returned contains input data resampled at the
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// destination audio device sample rate.
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virtual RefPtr<Buffer> get_more_samples(size_t max_bytes_to_read_from_input = 128 * KiB) override;
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virtual void reset() override { return seek(0); }
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// sample_index 0 is the start of the raw audio sample data
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// within the file/stream.
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virtual void seek(const int sample_index) override;
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virtual int loaded_samples() override { return m_loaded_samples; }
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virtual int total_samples() override { return m_total_samples; }
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virtual u32 sample_rate() override { return m_sample_rate; }
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virtual u16 num_channels() override { return m_num_channels; }
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virtual PcmSampleFormat pcm_format() override { return m_sample_format; }
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virtual RefPtr<Core::File> file() override { return m_file; }
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private:
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bool parse_header();
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bool valid { false };
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RefPtr<Core::File> m_file;
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OwnPtr<AK::InputStream> m_stream;
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AK::InputMemoryStream* m_memory_stream;
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String m_error_string;
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// TODO: We should probably move resampling into the audio server.
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//
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// It would avoid duplicate resampling code and would allow clients
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// to be agnostic of the destination audio device's sample rate.
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OwnPtr<ResampleHelper> m_resampler;
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u32 m_sample_rate { 0 };
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u16 m_num_channels { 0 };
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PcmSampleFormat m_sample_format;
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size_t m_byte_offset_of_data_samples { 0 };
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// FIXME: Get this value from the audio server
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int m_device_sample_rate { 44100 };
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int m_loaded_samples { 0 };
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int m_total_samples { 0 };
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};
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}
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