//////////////////////////////////////////////////////////////////////////// // **** WAVPACK **** // // Hybrid Lossless Wavefile Compressor // // Copyright (c) 1998 - 2004 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 static void strcpy_loc (char *dst, char *src) { while ((*dst++ = *src++) != 0); } ///////////////////////////// local table storage //////////////////////////// static 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 IBSS_ATTR; 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_loc (error, "invalid WavPack file!"); return NULL; } wps->block_bytes_left = wps->wphdr.ckSize - 24; if ((wps->wphdr.flags & UNKNOWN_FLAGS) || wps->wphdr.version < MIN_STREAM_VERS || wps->wphdr.version > MAX_STREAM_VERS) { strcpy_loc (error, "invalid 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_loc (error, wpc.error_message [0] ? wpc.error_message : "invalid 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.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 // 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 int32_t's for stereo files. The audio data is returned right-justified in // 32-bit int32_t's 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 can also be returned if the source was // floating point data (and this is normalized to +/-1.0). 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; wps->block_bytes_left = wps->wphdr.ckSize - 24; if (wps->wphdr.version < MIN_STREAM_VERS || wps->wphdr.version > MAX_STREAM_VERS) { strcpy_loc (wpc->error_message, "invalid WavPack file!"); 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 int32_t's, 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 int32_t buffer and must be // left-shifted 8, 16, or 24 bits if normalized int32_t's 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) > 1024 * 1024) return -1; } } // Open context for writing WavPack files. The returned context pointer is used // in all following calls to the library. A return value of NULL indicates // that memory could not be allocated for the context. WavpackContext *WavpackOpenFileOutput (void) { CLEAR (wpc); return &wpc; } // Set configuration for writing WavPack files. This must be done before // sending any actual samples, however it is okay to send wrapper or other // metadata before calling this. The "config" structure contains the following // required information: // config->bytes_per_sample see WavpackGetBytesPerSample() for info // config->bits_per_sample see WavpackGetBitsPerSample() for info // config->num_channels self evident // config->sample_rate self evident // In addition, the following fields and flags may be set: // config->flags: // -------------- // o CONFIG_HYBRID_FLAG select hybrid mode (must set bitrate) // o CONFIG_JOINT_STEREO select joint stereo (must set override also) // o CONFIG_JOINT_OVERRIDE override default joint stereo selection // o CONFIG_HYBRID_SHAPE select hybrid noise shaping (set override & // shaping_weight != 0.0) // o CONFIG_SHAPE_OVERRIDE override default hybrid noise shaping // (set CONFIG_HYBRID_SHAPE and shaping_weight) // o CONFIG_FAST_FLAG "fast" compression mode // o CONFIG_HIGH_FLAG "high" compression mode // o CONFIG_BITRATE_KBPS hybrid bitrate is kbps, not bits / sample // config->bitrate hybrid bitrate in either bits/sample or kbps // config->shaping_weight hybrid noise shaping coefficient override // config->float_norm_exp select floating-point data (127 for +/-1.0) // If the number of samples to be written is known then it should be passed // here. If the duration is not known then pass -1. In the case that the size // is not known (or the writing is terminated early) then it is suggested that // the application retrieve the first block written and let the library update // the total samples indication. A function is provided to do this update and // it should be done to the "correction" file also. If this cannot be done // (because a pipe is being used, for instance) then a valid WavPack will still // be created, but when applications want to access that file they will have // to seek all the way to the end to determine the actual duration. Also, if // a RIFF header has been included then it should be updated as well or the // WavPack file will not be directly unpackable to a valid wav file (although // it will still be usable by itself). A return of FALSE indicates an error. int WavpackSetConfiguration (WavpackContext *wpc, WavpackConfig *config, uint32_t total_samples) { WavpackStream *wps = &wpc->stream; uint32_t flags = (config->bytes_per_sample - 1), shift = 0; int num_chans = config->num_channels; int i; if ((wpc->config.flags & CONFIG_HYBRID_FLAG) || wpc->config.float_norm_exp || num_chans < 1 || num_chans > 2) return FALSE; wpc->total_samples = total_samples; wpc->config.sample_rate = config->sample_rate; wpc->config.num_channels = config->num_channels; wpc->config.bits_per_sample = config->bits_per_sample; wpc->config.bytes_per_sample = config->bytes_per_sample; wpc->config.flags = config->flags; shift = (config->bytes_per_sample * 8) - config->bits_per_sample; for (i = 0; i < 15; ++i) if (wpc->config.sample_rate == sample_rates [i]) break; flags |= i << SRATE_LSB; flags |= shift << SHIFT_LSB; flags |= CROSS_DECORR; if (!(config->flags & CONFIG_JOINT_OVERRIDE) || (config->flags & CONFIG_JOINT_STEREO)) flags |= JOINT_STEREO; flags |= INITIAL_BLOCK | FINAL_BLOCK; if (num_chans == 1) { flags &= ~(JOINT_STEREO | CROSS_DECORR | HYBRID_BALANCE); flags |= MONO_FLAG; } flags &= ~MAG_MASK; flags += (1 << MAG_LSB) * ((flags & BYTES_STORED) * 8 + 7); memcpy (wps->wphdr.ckID, "wvpk", 4); wps->wphdr.ckSize = sizeof (WavpackHeader) - 8; wps->wphdr.total_samples = wpc->total_samples; wps->wphdr.version = CUR_STREAM_VERS; wps->wphdr.flags = flags; pack_init (wpc); return TRUE; } // Add wrapper (currently RIFF only) to WavPack blocks. This should be called // before sending any audio samples. If the exact contents of the RIFF header // are not known because, for example, the file duration is uncertain or // trailing chunks are possible, simply write a "dummy" header of the correct // 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, uint32_t 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 int32_ts 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, int32_t *sample_buffer, uint32_t 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. uint32_t WavpackFinishBlock (WavpackContext *wpc) { WavpackStream *wps = &wpc->stream; uint32_t 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; }