react-native-tiny-wavpack-decoder 0.1.0

package/src/tiny-wavpack/lib/wputils.c

@@ -0,0 +1,351 @@
+////////////////////////////////////////////////////////////////////////////
+//                           **** WAVPACK ****                            //
+//                  Hybrid Lossless Wavefile Compressor                   //
+//              Copyright (c) 1998 - 2006 Conifer Software.               //
+//                          All Rights Reserved.                          //
+//      Distributed under the BSD Software License (see license.txt)      //
+////////////////////////////////////////////////////////////////////////////
+
+// wputils.c
+
+// This module provides a high-level interface for decoding WavPack 4.0 audio
+// streams and files. WavPack data is read with a stream reading callback. No
+// direct seeking is provided for, but it is possible to start decoding
+// anywhere in a WavPack stream. In this case, WavPack will be able to provide
+// the sample-accurate position when it synchs with the data and begins
+// decoding.
+
+#include "wavpack.h"
+
+#include <string.h>
+
+///////////////////////////// local table storage ////////////////////////////
+
+const uint32_t sample_rates [] = { 6000, 8000, 9600, 11025, 12000, 16000, 22050,
+    24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000 };
+
+///////////////////////////// executable code ////////////////////////////////
+
+static uint32_t read_next_header (read_stream infile, WavpackHeader *wphdr);
+        
+// This function reads data from the specified stream in search of a valid
+// WavPack 4.0 audio block. If this fails in 1 megabyte (or an invalid or
+// unsupported WavPack block is encountered) then an appropriate message is
+// copied to "error" and NULL is returned, otherwise a pointer to a
+// WavpackContext structure is returned (which is used to call all other
+// functions in this module). This can be initiated at the beginning of a
+// WavPack file, or anywhere inside a WavPack file. To determine the exact
+// position within the file use WavpackGetSampleIndex(). For demonstration
+// purposes this uses a single static copy of the WavpackContext structure,
+// so obviously it cannot be used for more than one file at a time. Also,
+// this function will not handle "correction" files, plays only the first
+// two channels of multi-channel files, and is limited in resolution in some
+// large integer or floating point files (but always provides at least 24 bits
+// of resolution).
+
+static WavpackContext wpc;
+
+WavpackContext *WavpackOpenFileInput (read_stream infile, char *error)
+{
+    WavpackStream *wps = &wpc.stream;
+    uint32_t bcount;
+
+    CLEAR (wpc);
+    wpc.infile = infile;
+    wpc.total_samples = (uint32_t) -1;
+    wpc.norm_offset = 0;
+    wpc.open_flags = 0;
+
+    // open the source file for reading and store the size
+
+    while (!wps->wphdr.block_samples) {
+
+        bcount = read_next_header (wpc.infile, &wps->wphdr);
+
+        if (bcount == (uint32_t) -1) {
+            strcpy (error, "not compatible with this version of WavPack file!");
+            return NULL;
+        }
+
+        if (wps->wphdr.block_samples && wps->wphdr.total_samples != (uint32_t) -1)
+            wpc.total_samples = wps->wphdr.total_samples;
+
+        if (!unpack_init (&wpc)) {
+            strcpy (error, wpc.error_message [0] ? wpc.error_message :
+                "not compatible with this version of WavPack file!");
+
+            return NULL;
+        }
+    }
+
+    wpc.config.flags &= ~0xff;
+    wpc.config.flags |= wps->wphdr.flags & 0xff;
+    wpc.config.bytes_per_sample = (wps->wphdr.flags & BYTES_STORED) + 1;
+    wpc.config.float_norm_exp = wps->float_norm_exp;
+
+    wpc.config.bits_per_sample = (wpc.config.bytes_per_sample * 8) - 
+        ((wps->wphdr.flags & SHIFT_MASK) >> SHIFT_LSB);
+
+    if (wpc.config.flags & FLOAT_DATA) {
+        wpc.config.bytes_per_sample = 3;
+        wpc.config.bits_per_sample = 24;
+    }
+
+    if (!wpc.config.sample_rate) {
+        if (!wps || !wps->wphdr.block_samples || (wps->wphdr.flags & SRATE_MASK) == SRATE_MASK)
+            wpc.config.sample_rate = 44100;
+        else
+            wpc.config.sample_rate = sample_rates [(wps->wphdr.flags & SRATE_MASK) >> SRATE_LSB];
+    }
+
+    if (!wpc.config.num_channels) {
+        wpc.config.num_channels = (wps->wphdr.flags & MONO_FLAG) ? 1 : 2;
+        wpc.config.channel_mask = 0x5 - wpc.config.num_channels;
+    }
+
+    if (!(wps->wphdr.flags & FINAL_BLOCK))
+        wpc.reduced_channels = (wps->wphdr.flags & MONO_FLAG) ? 1 : 2;
+
+    return &wpc;
+}
+
+// This function obtains general information about an open file and returns
+// a mask with the following bit values:
+
+// MODE_LOSSLESS:  file is lossless (pure lossless only)
+// MODE_HYBRID:  file is hybrid mode (lossy part only)
+// MODE_FLOAT:  audio data is 32-bit ieee floating point (but will provided
+//               in 24-bit integers for convenience)
+// MODE_HIGH:  file was created in "high" mode (information only)
+// MODE_FAST:  file was created in "fast" mode (information only)
+
+int WavpackGetMode (WavpackContext *wpc)
+{
+    int mode = 0;
+
+    if (wpc) {
+        if (wpc->config.flags & CONFIG_HYBRID_FLAG)
+            mode |= MODE_HYBRID;
+        else if (!(wpc->config.flags & CONFIG_LOSSY_MODE))
+            mode |= MODE_LOSSLESS;
+
+        if (wpc->lossy_blocks)
+            mode &= ~MODE_LOSSLESS;
+
+        if (wpc->config.flags & CONFIG_FLOAT_DATA)
+            mode |= MODE_FLOAT;
+
+        if (wpc->config.flags & CONFIG_HIGH_FLAG)
+            mode |= MODE_HIGH;
+
+        if (wpc->config.flags & CONFIG_FAST_FLAG)
+            mode |= MODE_FAST;
+    }
+
+    return mode;
+}
+
+// Unpack the specified number of samples from the current file position.
+// Note that "samples" here refers to "complete" samples, which would be
+// 2 longs for stereo files. The audio data is returned right-justified in
+// 32-bit longs in the endian mode native to the executing processor. So,
+// if the original data was 16-bit, then the values returned would be
+// +/-32k. Floating point data will be returned as 24-bit integers (and may
+// also be clipped). The actual number of samples unpacked is returned,
+// which should be equal to the number requested unless the end of fle is
+// encountered or an error occurs.
+
+uint32_t WavpackUnpackSamples (WavpackContext *wpc, int32_t *buffer, uint32_t samples)
+{
+    WavpackStream *wps = &wpc->stream;
+    uint32_t bcount, samples_unpacked = 0, samples_to_unpack;
+    int num_channels = wpc->config.num_channels;
+
+    while (samples) {
+        if (!wps->wphdr.block_samples || !(wps->wphdr.flags & INITIAL_BLOCK) ||
+            wps->sample_index >= wps->wphdr.block_index + wps->wphdr.block_samples) {
+                bcount = read_next_header (wpc->infile, &wps->wphdr);
+
+                if (bcount == (uint32_t) -1)
+                    break;
+
+                if (!wps->wphdr.block_samples || wps->sample_index == wps->wphdr.block_index)
+                    if (!unpack_init (wpc))
+                        break;
+        }
+
+        if (!wps->wphdr.block_samples || !(wps->wphdr.flags & INITIAL_BLOCK) ||
+            wps->sample_index >= wps->wphdr.block_index + wps->wphdr.block_samples)
+                continue;
+
+        if (wps->sample_index < wps->wphdr.block_index) {
+            samples_to_unpack = wps->wphdr.block_index - wps->sample_index;
+
+            if (samples_to_unpack > samples)
+                samples_to_unpack = samples;
+
+            wps->sample_index += samples_to_unpack;
+            samples_unpacked += samples_to_unpack;
+            samples -= samples_to_unpack;
+
+            if (wpc->reduced_channels)
+                samples_to_unpack *= wpc->reduced_channels;
+            else
+                samples_to_unpack *= num_channels;
+
+            while (samples_to_unpack--)
+                *buffer++ = 0;
+
+            continue;
+        }
+
+        samples_to_unpack = wps->wphdr.block_index + wps->wphdr.block_samples - wps->sample_index;
+
+        if (samples_to_unpack > samples)
+            samples_to_unpack = samples;
+
+        unpack_samples (wpc, buffer, samples_to_unpack);
+
+        if (wpc->reduced_channels)
+            buffer += samples_to_unpack * wpc->reduced_channels;
+        else
+            buffer += samples_to_unpack * num_channels;
+
+        samples_unpacked += samples_to_unpack;
+        samples -= samples_to_unpack;
+
+        if (wps->sample_index == wps->wphdr.block_index + wps->wphdr.block_samples) {
+            if (check_crc_error (wpc))
+                wpc->crc_errors++;
+        }
+
+        if (wps->sample_index == wpc->total_samples)
+            break;
+    }
+
+    return samples_unpacked;
+}
+
+// Get total number of samples contained in the WavPack file, or -1 if unknown
+
+uint32_t WavpackGetNumSamples (WavpackContext *wpc)
+{
+    return wpc ? wpc->total_samples : (uint32_t) -1;
+}
+
+// Get the current sample index position, or -1 if unknown
+
+uint32_t WavpackGetSampleIndex (WavpackContext *wpc)
+{
+    if (wpc)
+        return wpc->stream.sample_index;
+
+    return (uint32_t) -1;
+}
+
+// Get the number of errors encountered so far
+
+int WavpackGetNumErrors (WavpackContext *wpc)
+{
+    return wpc ? wpc->crc_errors : 0;
+}
+
+// return TRUE if any uncorrected lossy blocks were actually written or read
+
+int WavpackLossyBlocks (WavpackContext *wpc)
+{
+    return wpc ? wpc->lossy_blocks : 0;
+}
+
+// Returns the sample rate of the specified WavPack file
+
+uint32_t WavpackGetSampleRate (WavpackContext *wpc)
+{
+    return wpc ? wpc->config.sample_rate : 44100;
+}
+
+// Returns the number of channels of the specified WavPack file. Note that
+// this is the actual number of channels contained in the file, but this
+// version can only decode the first two.
+
+int WavpackGetNumChannels (WavpackContext *wpc)
+{
+    return wpc ? wpc->config.num_channels : 2;
+}
+
+// Returns the actual number of valid bits per sample contained in the
+// original file, which may or may not be a multiple of 8. Floating data
+// always has 32 bits, integers may be from 1 to 32 bits each. When this
+// value is not a multiple of 8, then the "extra" bits are located in the
+// LSBs of the results. That is, values are right justified when unpacked
+// into longs, but are left justified in the number of bytes used by the
+// original data.
+
+int WavpackGetBitsPerSample (WavpackContext *wpc)
+{
+    return wpc ? wpc->config.bits_per_sample : 16;
+}
+
+// Returns the number of bytes used for each sample (1 to 4) in the original
+// file. This is required information for the user of this module because the
+// audio data is returned in the LOWER bytes of the long buffer and must be
+// left-shifted 8, 16, or 24 bits if normalized longs are required.
+
+int WavpackGetBytesPerSample (WavpackContext *wpc)
+{
+    return wpc ? wpc->config.bytes_per_sample : 2;
+}
+
+// This function will return the actual number of channels decoded from the
+// file (which may or may not be less than the actual number of channels, but
+// will always be 1 or 2). Normally, this will be the front left and right
+// channels of a multi-channel file.
+
+int WavpackGetReducedChannels (WavpackContext *wpc)
+{
+    if (wpc)
+        return wpc->reduced_channels ? wpc->reduced_channels : wpc->config.num_channels;
+    else
+        return 2;
+}
+
+// Read from current file position until a valid 32-byte WavPack 4.0 header is
+// found and read into the specified pointer. The number of bytes skipped is
+// returned. If no WavPack header is found within 1 meg, then a -1 is returned
+// to indicate the error. No additional bytes are read past the header and it
+// is returned in the processor's native endian mode. Seeking is not required.
+
+static uint32_t read_next_header (read_stream infile, WavpackHeader *wphdr)
+{
+    char buffer [sizeof (*wphdr)], *sp = buffer + sizeof (*wphdr), *ep = sp;
+    uint32_t bytes_skipped = 0;
+    int bleft;
+
+    while (1) {
+        if (sp < ep) {
+            bleft = ep - sp;
+            memcpy (buffer, sp, bleft);
+        }
+        else
+            bleft = 0;
+
+        if (infile (buffer + bleft, sizeof (*wphdr) - bleft) != (int32_t) sizeof (*wphdr) - bleft)
+            return -1;
+
+        sp = buffer;
+
+        if (*sp++ == 'w' && *sp == 'v' && *++sp == 'p' && *++sp == 'k' &&
+            !(*++sp & 1) && sp [2] < 16 && !sp [3] && sp [5] == 4 &&
+            sp [4] >= (MIN_STREAM_VERS & 0xff) && sp [4] <= (MAX_STREAM_VERS & 0xff)) {
+                memcpy (wphdr, buffer, sizeof (*wphdr));
+                little_endian_to_native (wphdr, WavpackHeaderFormat);
+                return bytes_skipped;
+            }
+
+        while (sp < ep && *sp != 'w')
+            sp++;
+
+        if ((bytes_skipped += sp - buffer) > 1048576L)
+            return -1;
+    }
+}