/* * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with this program; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include "config.h" #include "decoder_api.h" #include "audio_check.h" #include "path.h" #include "utils.h" #include <wavpack/wavpack.h> #include <glib.h> #include <assert.h> #include <unistd.h> #include <stdio.h> #include <stdlib.h> #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "wavpack" /* pick 1020 since its devisible for 8,16,24, and 32-bit audio */ #define CHUNK_SIZE 1020 #define ERRORLEN 80 static struct { const char *name; enum tag_type type; } tagtypes[] = { { "artist", TAG_ARTIST }, { "album", TAG_ALBUM }, { "title", TAG_TITLE }, { "track", TAG_TRACK }, { "name", TAG_NAME }, { "genre", TAG_GENRE }, { "date", TAG_DATE }, { "composer", TAG_COMPOSER }, { "performer", TAG_PERFORMER }, { "comment", TAG_COMMENT }, { "disc", TAG_DISC }, }; /** A pointer type for format converter function. */ typedef void (*format_samples_t)( int bytes_per_sample, void *buffer, uint32_t count ); /* * This function has been borrowed from the tiny player found on * wavpack.com. Modifications were required because mpd only handles * max 24-bit samples. */ static void format_samples_int(int bytes_per_sample, void *buffer, uint32_t count) { int32_t *src = buffer; switch (bytes_per_sample) { case 1: { int8_t *dst = buffer; /* * The asserts like the following one are because we do the * formatting of samples within a single buffer. The size * of the output samples never can be greater than the size * of the input ones. Otherwise we would have an overflow. */ assert_static(sizeof(*dst) <= sizeof(*src)); /* pass through and align 8-bit samples */ while (count--) { *dst++ = *src++; } break; } case 2: { uint16_t *dst = buffer; assert_static(sizeof(*dst) <= sizeof(*src)); /* pass through and align 16-bit samples */ while (count--) { *dst++ = *src++; } break; } case 3: case 4: /* do nothing */ break; } } /* * This function converts floating point sample data to 24-bit integer. */ static void format_samples_float(G_GNUC_UNUSED int bytes_per_sample, void *buffer, uint32_t count) { int32_t *dst = buffer; float *src = buffer; assert_static(sizeof(*dst) <= sizeof(*src)); while (count--) { *dst++ = (int32_t)(*src++ + 0.5f); } } /** * Choose a MPD sample format from libwavpacks' number of bits. */ static enum sample_format wavpack_bits_to_sample_format(bool is_float, int bytes_per_sample) { if (is_float) return SAMPLE_FORMAT_S24_P32; switch (bytes_per_sample) { case 1: return SAMPLE_FORMAT_S8; case 2: return SAMPLE_FORMAT_S16; case 3: return SAMPLE_FORMAT_S24_P32; case 4: return SAMPLE_FORMAT_S32; default: return SAMPLE_FORMAT_UNDEFINED; } } /* * This does the main decoding thing. * Requires an already opened WavpackContext. */ static void wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek) { GError *error = NULL; bool is_float; enum sample_format sample_format; struct audio_format audio_format; format_samples_t format_samples; char chunk[CHUNK_SIZE]; int samples_requested, samples_got; float total_time; int bytes_per_sample, output_sample_size; is_float = (WavpackGetMode(wpc) & MODE_FLOAT) != 0; sample_format = wavpack_bits_to_sample_format(is_float, WavpackGetBytesPerSample(wpc)); if (!audio_format_init_checked(&audio_format, WavpackGetSampleRate(wpc), sample_format, WavpackGetNumChannels(wpc), &error)) { g_warning("%s", error->message); g_error_free(error); return; } if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT) { format_samples = format_samples_float; } else { format_samples = format_samples_int; } total_time = WavpackGetNumSamples(wpc); total_time /= audio_format.sample_rate; bytes_per_sample = WavpackGetBytesPerSample(wpc); output_sample_size = audio_format_frame_size(&audio_format); /* wavpack gives us all kind of samples in a 32-bit space */ samples_requested = sizeof(chunk) / (4 * audio_format.channels); decoder_initialized(decoder, &audio_format, can_seek, total_time); do { if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) { if (can_seek) { unsigned where = decoder_seek_where(decoder) * audio_format.sample_rate; if (WavpackSeekSample(wpc, where)) { decoder_command_finished(decoder); } else { decoder_seek_error(decoder); } } else { decoder_seek_error(decoder); } } if (decoder_get_command(decoder) == DECODE_COMMAND_STOP) { break; } samples_got = WavpackUnpackSamples( wpc, (int32_t *)chunk, samples_requested ); if (samples_got > 0) { int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 + 0.5); format_samples( bytes_per_sample, chunk, samples_got * audio_format.channels ); decoder_data( decoder, NULL, chunk, samples_got * output_sample_size, bitrate ); } } while (samples_got > 0); } /** * Locate and parse a floating point tag. Returns true if it was * found. */ static bool wavpack_tag_float(WavpackContext *wpc, const char *key, float *value_r) { char buffer[64]; int ret; ret = WavpackGetTagItem(wpc, key, buffer, sizeof(buffer)); if (ret <= 0) return false; *value_r = atof(buffer); return true; } static bool wavpack_replaygain(struct replay_gain_info *replay_gain_info, WavpackContext *wpc) { bool found = false; replay_gain_info_init(replay_gain_info); found |= wavpack_tag_float( wpc, "replaygain_track_gain", &replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain ); found |= wavpack_tag_float( wpc, "replaygain_track_peak", &replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak ); found |= wavpack_tag_float( wpc, "replaygain_album_gain", &replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain ); found |= wavpack_tag_float( wpc, "replaygain_album_peak", &replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak ); return found; } /* * Reads metainfo from the specified file. */ static struct tag * wavpack_tagdup(const char *fname) { WavpackContext *wpc; struct tag *tag; char error[ERRORLEN]; char *s; int size, allocated_size; wpc = WavpackOpenFileInput(fname, error, OPEN_TAGS, 0); if (wpc == NULL) { g_warning( "failed to open WavPack file \"%s\": %s\n", fname, error ); return NULL; } tag = tag_new(); tag->time = WavpackGetNumSamples(wpc); tag->time /= WavpackGetSampleRate(wpc); allocated_size = 0; s = NULL; for (unsigned i = 0; i < G_N_ELEMENTS(tagtypes); ++i) { size = WavpackGetTagItem(wpc, tagtypes[i].name, NULL, 0); if (size > 0) { ++size; /* EOS */ if (s == NULL) { s = g_malloc(size); allocated_size = size; } else if (size > allocated_size) { char *t = (char *)g_realloc(s, size); allocated_size = size; s = t; } WavpackGetTagItem(wpc, tagtypes[i].name, s, size); tag_add_item(tag, tagtypes[i].type, s); } } g_free(s); WavpackCloseFile(wpc); return tag; } /* * mpd input_stream <=> WavpackStreamReader wrapper callbacks */ /* This struct is needed for per-stream last_byte storage. */ struct wavpack_input { struct decoder *decoder; struct input_stream *is; /* Needed for push_back_byte() */ int last_byte; }; /** * Little wrapper for struct wavpack_input to cast from void *. */ static struct wavpack_input * wpin(void *id) { assert(id); return id; } static int32_t wavpack_input_read_bytes(void *id, void *data, int32_t bcount) { uint8_t *buf = (uint8_t *)data; int32_t i = 0; if (wpin(id)->last_byte != EOF) { *buf++ = wpin(id)->last_byte; wpin(id)->last_byte = EOF; --bcount; ++i; } /* wavpack fails if we return a partial read, so we just wait until the buffer is full */ while (bcount > 0) { size_t nbytes = decoder_read( wpin(id)->decoder, wpin(id)->is, buf, bcount ); if (nbytes == 0) { /* EOF, error or a decoder command */ break; } i += nbytes; bcount -= nbytes; buf += nbytes; } return i; } static uint32_t wavpack_input_get_pos(void *id) { return wpin(id)->is->offset; } static int wavpack_input_set_pos_abs(void *id, uint32_t pos) { return input_stream_seek(wpin(id)->is, pos, SEEK_SET, NULL) ? 0 : -1; } static int wavpack_input_set_pos_rel(void *id, int32_t delta, int mode) { return input_stream_seek(wpin(id)->is, delta, mode, NULL) ? 0 : -1; } static int wavpack_input_push_back_byte(void *id, int c) { if (wpin(id)->last_byte == EOF) { wpin(id)->last_byte = c; return c; } else { return EOF; } } static uint32_t wavpack_input_get_length(void *id) { if (wpin(id)->is->size < 0) return 0; return wpin(id)->is->size; } static int wavpack_input_can_seek(void *id) { return wpin(id)->is->seekable; } static WavpackStreamReader mpd_is_reader = { .read_bytes = wavpack_input_read_bytes, .get_pos = wavpack_input_get_pos, .set_pos_abs = wavpack_input_set_pos_abs, .set_pos_rel = wavpack_input_set_pos_rel, .push_back_byte = wavpack_input_push_back_byte, .get_length = wavpack_input_get_length, .can_seek = wavpack_input_can_seek, .write_bytes = NULL /* no need to write edited tags */ }; static void wavpack_input_init(struct wavpack_input *isp, struct decoder *decoder, struct input_stream *is) { isp->decoder = decoder; isp->is = is; isp->last_byte = EOF; } static struct input_stream * wavpack_open_wvc(struct decoder *decoder, const char *uri, struct wavpack_input *wpi) { struct input_stream *is_wvc; char *wvc_url = NULL; char first_byte; size_t nbytes; /* * As we use dc->utf8url, this function will be bad for * single files. utf8url is not absolute file path :/ */ if (uri == NULL) return false; wvc_url = g_strconcat(uri, "c", NULL); is_wvc = input_stream_open(wvc_url, NULL); g_free(wvc_url); if (is_wvc == NULL) return NULL; /* * And we try to buffer in order to get know * about a possible 404 error. */ nbytes = decoder_read( decoder, is_wvc, &first_byte, sizeof(first_byte) ); if (nbytes == 0) { input_stream_close(is_wvc); return NULL; } /* push it back */ wavpack_input_init(wpi, decoder, is_wvc); wpi->last_byte = first_byte; return is_wvc; } /* * Decodes a stream. */ static void wavpack_streamdecode(struct decoder * decoder, struct input_stream *is) { char error[ERRORLEN]; WavpackContext *wpc; struct input_stream *is_wvc; int open_flags = OPEN_NORMALIZE; struct wavpack_input isp, isp_wvc; bool can_seek = is->seekable; is_wvc = wavpack_open_wvc(decoder, is->uri, &isp_wvc); if (is_wvc != NULL) { open_flags |= OPEN_WVC; can_seek &= is_wvc->seekable; } if (!can_seek) { open_flags |= OPEN_STREAMING; } wavpack_input_init(&isp, decoder, is); wpc = WavpackOpenFileInputEx( &mpd_is_reader, &isp, open_flags & OPEN_WVC ? &isp_wvc : NULL, error, open_flags, 23 ); if (wpc == NULL) { g_warning("failed to open WavPack stream: %s\n", error); return; } wavpack_decode(decoder, wpc, can_seek); WavpackCloseFile(wpc); if (open_flags & OPEN_WVC) { input_stream_close(is_wvc); } } /* * Decodes a file. */ static void wavpack_filedecode(struct decoder *decoder, const char *fname) { char error[ERRORLEN]; WavpackContext *wpc; wpc = WavpackOpenFileInput( fname, error, OPEN_TAGS | OPEN_WVC | OPEN_NORMALIZE, 23 ); if (wpc == NULL) { g_warning( "failed to open WavPack file \"%s\": %s\n", fname, error ); return; } struct replay_gain_info replay_gain_info; if (wavpack_replaygain(&replay_gain_info, wpc)) decoder_replay_gain(decoder, &replay_gain_info); wavpack_decode(decoder, wpc, true); WavpackCloseFile(wpc); } static char const *const wavpack_suffixes[] = { "wv", NULL }; static char const *const wavpack_mime_types[] = { "audio/x-wavpack", NULL }; const struct decoder_plugin wavpack_decoder_plugin = { .name = "wavpack", .stream_decode = wavpack_streamdecode, .file_decode = wavpack_filedecode, .tag_dup = wavpack_tagdup, .suffixes = wavpack_suffixes, .mime_types = wavpack_mime_types };