85e03767f7
improved efficiency somewhat by looping through data in tighter passes. Code is basically ready for an attempt at direct recording. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7088 a1c6a512-1295-4272-9138-f99709370657
549 lines
20 KiB
C
549 lines
20 KiB
C
////////////////////////////////////////////////////////////////////////////
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// **** WAVPACK **** //
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// Hybrid Lossless Wavefile Compressor //
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// Copyright (c) 1998 - 2004 Conifer Software. //
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// All Rights Reserved. //
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// Distributed under the BSD Software License (see license.txt) //
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////////////////////////////////////////////////////////////////////////////
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// wputils.c
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// This module provides a high-level interface for decoding WavPack 4.0 audio
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// streams and files. WavPack data is read with a stream reading callback. No
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// direct seeking is provided for, but it is possible to start decoding
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// anywhere in a WavPack stream. In this case, WavPack will be able to provide
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// the sample-accurate position when it synchs with the data and begins
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// decoding.
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#include "wavpack.h"
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#include <string.h>
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static void strcpy_loc (char *dst, char *src) { while ((*dst++ = *src++) != 0); }
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///////////////////////////// local table storage ////////////////////////////
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const ulong sample_rates [] = { 6000, 8000, 9600, 11025, 12000, 16000, 22050,
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24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000 };
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///////////////////////////// executable code ////////////////////////////////
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static ulong read_next_header (read_stream infile, WavpackHeader *wphdr);
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// This function reads data from the specified stream in search of a valid
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// WavPack 4.0 audio block. If this fails in 1 megabyte (or an invalid or
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// unsupported WavPack block is encountered) then an appropriate message is
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// copied to "error" and NULL is returned, otherwise a pointer to a
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// WavpackContext structure is returned (which is used to call all other
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// functions in this module). This can be initiated at the beginning of a
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// WavPack file, or anywhere inside a WavPack file. To determine the exact
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// position within the file use WavpackGetSampleIndex(). For demonstration
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// purposes this uses a single static copy of the WavpackContext structure,
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// so obviously it cannot be used for more than one file at a time. Also,
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// this function will not handle "correction" files, plays only the first
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// two channels of multi-channel files, and is limited in resolution in some
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// large integer or floating point files (but always provides at least 24 bits
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// of resolution).
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static WavpackContext wpc IDATA_ATTR;
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WavpackContext *WavpackOpenFileInput (read_stream infile, char *error)
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{
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WavpackStream *wps = &wpc.stream;
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ulong bcount;
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CLEAR (wpc);
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wpc.infile = infile;
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wpc.total_samples = (ulong) -1;
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wpc.norm_offset = 0;
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wpc.open_flags = 0;
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// open the source file for reading and store the size
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while (!wps->wphdr.block_samples) {
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bcount = read_next_header (wpc.infile, &wps->wphdr);
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if (bcount == (ulong) -1) {
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strcpy_loc (error, "invalid WavPack file!");
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return NULL;
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}
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if ((wps->wphdr.flags & UNKNOWN_FLAGS) || wps->wphdr.version < 0x402 || wps->wphdr.version > 0x40f) {
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strcpy_loc (error, "invalid WavPack file!");
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return NULL;
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}
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if (wps->wphdr.block_samples && wps->wphdr.total_samples != (ulong) -1)
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wpc.total_samples = wps->wphdr.total_samples;
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if (!unpack_init (&wpc)) {
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strcpy_loc (error, wpc.error_message [0] ? wpc.error_message :
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"invalid WavPack file!");
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return NULL;
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}
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}
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wpc.config.flags &= ~0xff;
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wpc.config.flags |= wps->wphdr.flags & 0xff;
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wpc.config.bytes_per_sample = (wps->wphdr.flags & BYTES_STORED) + 1;
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wpc.config.float_norm_exp = wps->float_norm_exp;
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wpc.config.bits_per_sample = (wpc.config.bytes_per_sample * 8) -
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((wps->wphdr.flags & SHIFT_MASK) >> SHIFT_LSB);
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if (!wpc.config.sample_rate) {
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if (!wps || !wps->wphdr.block_samples || (wps->wphdr.flags & SRATE_MASK) == SRATE_MASK)
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wpc.config.sample_rate = 44100;
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else
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wpc.config.sample_rate = sample_rates [(wps->wphdr.flags & SRATE_MASK) >> SRATE_LSB];
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}
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if (!wpc.config.num_channels) {
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wpc.config.num_channels = (wps->wphdr.flags & MONO_FLAG) ? 1 : 2;
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wpc.config.channel_mask = 0x5 - wpc.config.num_channels;
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}
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if (!(wps->wphdr.flags & FINAL_BLOCK))
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wpc.reduced_channels = (wps->wphdr.flags & MONO_FLAG) ? 1 : 2;
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return &wpc;
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}
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// This function obtains general information about an open file and returns
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// a mask with the following bit values:
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// MODE_LOSSLESS: file is lossless (pure lossless only)
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// MODE_HYBRID: file is hybrid mode (lossy part only)
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// MODE_FLOAT: audio data is 32-bit ieee floating point
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// MODE_HIGH: file was created in "high" mode (information only)
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// MODE_FAST: file was created in "fast" mode (information only)
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int WavpackGetMode (WavpackContext *wpc)
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{
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int mode = 0;
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if (wpc) {
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if (wpc->config.flags & CONFIG_HYBRID_FLAG)
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mode |= MODE_HYBRID;
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else if (!(wpc->config.flags & CONFIG_LOSSY_MODE))
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mode |= MODE_LOSSLESS;
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if (wpc->lossy_blocks)
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mode &= ~MODE_LOSSLESS;
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if (wpc->config.flags & CONFIG_FLOAT_DATA)
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mode |= MODE_FLOAT;
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if (wpc->config.flags & CONFIG_HIGH_FLAG)
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mode |= MODE_HIGH;
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if (wpc->config.flags & CONFIG_FAST_FLAG)
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mode |= MODE_FAST;
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}
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return mode;
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}
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// Unpack the specified number of samples from the current file position.
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// Note that "samples" here refers to "complete" samples, which would be
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// 2 longs for stereo files. The audio data is returned right-justified in
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// 32-bit longs in the endian mode native to the executing processor. So,
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// if the original data was 16-bit, then the values returned would be
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// +/-32k. Floating point data can also be returned if the source was
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// floating point data (and this is normalized to +/-1.0). The actual number
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// of samples unpacked is returned, which should be equal to the number
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// requested unless the end of fle is encountered or an error occurs.
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ulong WavpackUnpackSamples (WavpackContext *wpc, long *buffer, ulong samples)
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{
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WavpackStream *wps = &wpc->stream;
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ulong bcount, samples_unpacked = 0, samples_to_unpack;
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int num_channels = wpc->config.num_channels;
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while (samples) {
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if (!wps->wphdr.block_samples || !(wps->wphdr.flags & INITIAL_BLOCK) ||
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wps->sample_index >= wps->wphdr.block_index + wps->wphdr.block_samples) {
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bcount = read_next_header (wpc->infile, &wps->wphdr);
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if (bcount == (ulong) -1)
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break;
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if (wps->wphdr.version < 0x402 || wps->wphdr.version > 0x40f) {
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strcpy_loc (wpc->error_message, "invalid WavPack file!");
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break;
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}
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if (!wps->wphdr.block_samples || wps->sample_index == wps->wphdr.block_index)
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if (!unpack_init (wpc))
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break;
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}
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if (!wps->wphdr.block_samples || !(wps->wphdr.flags & INITIAL_BLOCK) ||
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wps->sample_index >= wps->wphdr.block_index + wps->wphdr.block_samples)
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continue;
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if (wps->sample_index < wps->wphdr.block_index) {
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samples_to_unpack = wps->wphdr.block_index - wps->sample_index;
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if (samples_to_unpack > samples)
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samples_to_unpack = samples;
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wps->sample_index += samples_to_unpack;
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samples_unpacked += samples_to_unpack;
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samples -= samples_to_unpack;
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if (wpc->reduced_channels)
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samples_to_unpack *= wpc->reduced_channels;
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else
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samples_to_unpack *= num_channels;
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while (samples_to_unpack--)
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*buffer++ = 0;
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continue;
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}
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samples_to_unpack = wps->wphdr.block_index + wps->wphdr.block_samples - wps->sample_index;
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if (samples_to_unpack > samples)
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samples_to_unpack = samples;
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unpack_samples (wpc, buffer, samples_to_unpack);
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if (wpc->reduced_channels)
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buffer += samples_to_unpack * wpc->reduced_channels;
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else
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buffer += samples_to_unpack * num_channels;
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samples_unpacked += samples_to_unpack;
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samples -= samples_to_unpack;
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if (wps->sample_index == wps->wphdr.block_index + wps->wphdr.block_samples) {
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if (check_crc_error (wpc))
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wpc->crc_errors++;
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}
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if (wps->sample_index == wpc->total_samples)
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break;
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}
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return samples_unpacked;
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}
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// Get total number of samples contained in the WavPack file, or -1 if unknown
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ulong WavpackGetNumSamples (WavpackContext *wpc)
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{
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return wpc ? wpc->total_samples : (ulong) -1;
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}
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// Get the current sample index position, or -1 if unknown
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ulong WavpackGetSampleIndex (WavpackContext *wpc)
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{
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if (wpc)
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return wpc->stream.sample_index;
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return (ulong) -1;
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}
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// Get the number of errors encountered so far
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int WavpackGetNumErrors (WavpackContext *wpc)
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{
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return wpc ? wpc->crc_errors : 0;
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}
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// return TRUE if any uncorrected lossy blocks were actually written or read
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int WavpackLossyBlocks (WavpackContext *wpc)
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{
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return wpc ? wpc->lossy_blocks : 0;
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}
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// Returns the sample rate of the specified WavPack file
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ulong WavpackGetSampleRate (WavpackContext *wpc)
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{
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return wpc ? wpc->config.sample_rate : 44100;
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}
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// Returns the number of channels of the specified WavPack file. Note that
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// this is the actual number of channels contained in the file, but this
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// version can only decode the first two.
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int WavpackGetNumChannels (WavpackContext *wpc)
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{
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return wpc ? wpc->config.num_channels : 2;
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}
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// Returns the actual number of valid bits per sample contained in the
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// original file, which may or may not be a multiple of 8. Floating data
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// always has 32 bits, integers may be from 1 to 32 bits each. When this
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// value is not a multiple of 8, then the "extra" bits are located in the
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// LSBs of the results. That is, values are right justified when unpacked
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// into longs, but are left justified in the number of bytes used by the
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// original data.
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int WavpackGetBitsPerSample (WavpackContext *wpc)
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{
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return wpc ? wpc->config.bits_per_sample : 16;
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}
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// Returns the number of bytes used for each sample (1 to 4) in the original
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// file. This is required information for the user of this module because the
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// audio data is returned in the LOWER bytes of the long buffer and must be
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// left-shifted 8, 16, or 24 bits if normalized longs are required.
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int WavpackGetBytesPerSample (WavpackContext *wpc)
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{
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return wpc ? wpc->config.bytes_per_sample : 2;
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}
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// This function will return the actual number of channels decoded from the
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// file (which may or may not be less than the actual number of channels, but
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// will always be 1 or 2). Normally, this will be the front left and right
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// channels of a multi-channel file.
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int WavpackGetReducedChannels (WavpackContext *wpc)
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{
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if (wpc)
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return wpc->reduced_channels ? wpc->reduced_channels : wpc->config.num_channels;
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else
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return 2;
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}
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// Read from current file position until a valid 32-byte WavPack 4.0 header is
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// found and read into the specified pointer. The number of bytes skipped is
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// returned. If no WavPack header is found within 1 meg, then a -1 is returned
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// to indicate the error. No additional bytes are read past the header and it
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// is returned in the processor's native endian mode. Seeking is not required.
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static ulong read_next_header (read_stream infile, WavpackHeader *wphdr)
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{
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char buffer [sizeof (*wphdr)], *sp = buffer + sizeof (*wphdr), *ep = sp;
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ulong bytes_skipped = 0;
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int bleft;
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while (1) {
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if (sp < ep) {
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bleft = ep - sp;
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memcpy (buffer, sp, bleft);
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}
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else
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bleft = 0;
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if (infile (buffer + bleft, sizeof (*wphdr) - bleft) != (long) sizeof (*wphdr) - bleft)
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return -1;
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sp = buffer;
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if (*sp++ == 'w' && *sp == 'v' && *++sp == 'p' && *++sp == 'k' &&
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!(*++sp & 1) && sp [2] < 16 && !sp [3] && sp [5] == 4 && sp [4] >= 2 && sp [4] <= 0xf) {
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memcpy (wphdr, buffer, sizeof (*wphdr));
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little_endian_to_native (wphdr, WavpackHeaderFormat);
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return bytes_skipped;
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}
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while (sp < ep && *sp != 'w')
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sp++;
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if ((bytes_skipped += sp - buffer) > 1024 * 1024)
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return -1;
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}
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}
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// Open context for writing WavPack files. The returned context pointer is used
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// in all following calls to the library. A return value of NULL indicates
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// that memory could not be allocated for the context.
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WavpackContext *WavpackOpenFileOutput (void)
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{
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CLEAR (wpc);
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return &wpc;
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}
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// Set configuration for writing WavPack files. This must be done before
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// sending any actual samples, however it is okay to send wrapper or other
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// metadata before calling this. The "config" structure contains the following
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// required information:
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// config->bytes_per_sample see WavpackGetBytesPerSample() for info
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// config->bits_per_sample see WavpackGetBitsPerSample() for info
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// config->num_channels self evident
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// config->sample_rate self evident
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// In addition, the following fields and flags may be set:
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// config->flags:
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// --------------
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// o CONFIG_HYBRID_FLAG select hybrid mode (must set bitrate)
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// o CONFIG_JOINT_STEREO select joint stereo (must set override also)
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// o CONFIG_JOINT_OVERRIDE override default joint stereo selection
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// o CONFIG_HYBRID_SHAPE select hybrid noise shaping (set override &
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// shaping_weight != 0.0)
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// o CONFIG_SHAPE_OVERRIDE override default hybrid noise shaping
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// (set CONFIG_HYBRID_SHAPE and shaping_weight)
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// o CONFIG_FAST_FLAG "fast" compression mode
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// o CONFIG_HIGH_FLAG "high" compression mode
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// o CONFIG_BITRATE_KBPS hybrid bitrate is kbps, not bits / sample
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// config->bitrate hybrid bitrate in either bits/sample or kbps
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// config->shaping_weight hybrid noise shaping coefficient override
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// config->float_norm_exp select floating-point data (127 for +/-1.0)
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// If the number of samples to be written is known then it should be passed
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// here. If the duration is not known then pass -1. In the case that the size
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// is not known (or the writing is terminated early) then it is suggested that
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// the application retrieve the first block written and let the library update
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// the total samples indication. A function is provided to do this update and
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// it should be done to the "correction" file also. If this cannot be done
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// (because a pipe is being used, for instance) then a valid WavPack will still
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// be created, but when applications want to access that file they will have
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// to seek all the way to the end to determine the actual duration. Also, if
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// a RIFF header has been included then it should be updated as well or the
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// WavPack file will not be directly unpackable to a valid wav file (although
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// it will still be usable by itself). A return of FALSE indicates an error.
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int WavpackSetConfiguration (WavpackContext *wpc, WavpackConfig *config, ulong total_samples)
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{
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WavpackStream *wps = &wpc->stream;
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ulong flags = (config->bytes_per_sample - 1), shift = 0;
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int num_chans = config->num_channels;
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int i;
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if ((wpc->config.flags & CONFIG_HYBRID_FLAG) ||
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wpc->config.float_norm_exp ||
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num_chans < 1 || num_chans > 2)
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return FALSE;
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wpc->total_samples = total_samples;
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wpc->config.sample_rate = config->sample_rate;
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wpc->config.num_channels = config->num_channels;
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wpc->config.bits_per_sample = config->bits_per_sample;
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wpc->config.bytes_per_sample = config->bytes_per_sample;
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wpc->config.flags = config->flags;
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shift = (config->bytes_per_sample * 8) - config->bits_per_sample;
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for (i = 0; i < 15; ++i)
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if (wpc->config.sample_rate == sample_rates [i])
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break;
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flags |= i << SRATE_LSB;
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flags |= shift << SHIFT_LSB;
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flags |= CROSS_DECORR;
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if (!(config->flags & CONFIG_JOINT_OVERRIDE) || (config->flags & CONFIG_JOINT_STEREO))
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flags |= JOINT_STEREO;
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flags |= INITIAL_BLOCK | FINAL_BLOCK;
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if (num_chans == 1) {
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flags &= ~(JOINT_STEREO | CROSS_DECORR | HYBRID_BALANCE);
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flags |= MONO_FLAG;
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}
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flags &= ~MAG_MASK;
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flags += (1 << MAG_LSB) * ((flags & BYTES_STORED) * 8 + 7);
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memcpy (wps->wphdr.ckID, "wvpk", 4);
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wps->wphdr.ckSize = sizeof (WavpackHeader) - 8;
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wps->wphdr.total_samples = wpc->total_samples;
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wps->wphdr.version = 0x403;
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wps->wphdr.flags = flags;
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pack_init (wpc);
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return TRUE;
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}
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// Add wrapper (currently RIFF only) to WavPack blocks. This should be called
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// before sending any audio samples. If the exact contents of the RIFF header
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// are not known because, for example, the file duration is uncertain or
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// trailing chunks are possible, simply write a "dummy" header of the correct
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// length. When all data has been written it will be possible to read the
|
|
// first block written and update the header directly. An example of this can
|
|
// be found in the Audition filter.
|
|
|
|
void WavpackAddWrapper (WavpackContext *wpc, void *data, ulong bcount)
|
|
{
|
|
wpc->wrapper_data = data;
|
|
wpc->wrapper_bytes = bcount;
|
|
}
|
|
|
|
// Start a WavPack block to be stored in the specified buffer. This must be
|
|
// called before calling WavpackPackSamples(). Note that writing CANNOT wrap
|
|
// in the buffer; the entire output block must fit in the buffer.
|
|
|
|
int WavpackStartBlock (WavpackContext *wpc, uchar *begin, uchar *end)
|
|
{
|
|
wpc->stream.blockbuff = begin;
|
|
wpc->stream.blockend = end;
|
|
return pack_start_block (wpc);
|
|
}
|
|
|
|
// Pack the specified samples. Samples must be stored in longs in the native
|
|
// endian format of the executing processor. The number of samples specified
|
|
// indicates composite samples (sometimes called "frames"). So, the actual
|
|
// number of data points would be this "sample_count" times the number of
|
|
// channels. The caller must decide how many samples to place in each
|
|
// WavPack block (1/2 second is common), but this function may be called as
|
|
// many times as desired to build the final block (and performs the actual
|
|
// compression during the call). A return of FALSE indicates an error.
|
|
|
|
int WavpackPackSamples (WavpackContext *wpc, long *sample_buffer, ulong sample_count)
|
|
{
|
|
if (!sample_count || pack_samples (wpc, sample_buffer, sample_count))
|
|
return TRUE;
|
|
|
|
strcpy_loc (wpc->error_message, "output buffer overflowed!");
|
|
return FALSE;
|
|
}
|
|
|
|
// Finish the WavPack block being built, returning the total size of the
|
|
// block in bytes. Note that the possible conversion of the WavPack header to
|
|
// little-endian takes place here.
|
|
|
|
ulong WavpackFinishBlock (WavpackContext *wpc)
|
|
{
|
|
WavpackStream *wps = &wpc->stream;
|
|
ulong bcount;
|
|
|
|
pack_finish_block (wpc);
|
|
bcount = ((WavpackHeader *) wps->blockbuff)->ckSize + 8;
|
|
native_to_little_endian ((WavpackHeader *) wps->blockbuff, WavpackHeaderFormat);
|
|
|
|
return bcount;
|
|
}
|
|
|
|
// Given the pointer to the first block written (to either a .wv or .wvc file),
|
|
// update the block with the actual number of samples written. This should
|
|
// be done if WavpackSetConfiguration() was called with an incorrect number
|
|
// of samples (or -1). It is the responsibility of the application to read and
|
|
// rewrite the block. An example of this can be found in the Audition filter.
|
|
|
|
void WavpackUpdateNumSamples (WavpackContext *wpc, void *first_block)
|
|
{
|
|
little_endian_to_native (wpc, WavpackHeaderFormat);
|
|
((WavpackHeader *) first_block)->total_samples = WavpackGetSampleIndex (wpc);
|
|
native_to_little_endian (wpc, WavpackHeaderFormat);
|
|
}
|
|
|
|
// Given the pointer to the first block written to a WavPack file, this
|
|
// function returns the location of the stored RIFF header that was originally
|
|
// written with WavpackAddWrapper(). This would normally be used to update
|
|
// the wav header to indicate that a different number of samples was actually
|
|
// written or if additional RIFF chunks are written at the end of the file.
|
|
// It is the responsibility of the application to read and rewrite the block.
|
|
// An example of this can be found in the Audition filter.
|
|
|
|
void *WavpackGetWrapperLocation (void *first_block)
|
|
{
|
|
if (((uchar *) first_block) [32] == ID_RIFF_HEADER)
|
|
return ((uchar *) first_block) + 34;
|
|
else
|
|
return NULL;
|
|
}
|
|
|