X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/f8f8039fc44104b43b94467d17a8d71fc40f219b..e7eb3a2744aa45179daea235800753d3d1955338:/server/decode.c

diff --git a/server/decode.c b/server/decode.c
index 434ad00..ef999b3 100644
--- a/server/decode.c
+++ b/server/decode.c
@@ -1,43 +1,41 @@
 /*
  * This file is part of DisOrder
- * Copyright (C) 2007 Richard Kettlewell
+ * Copyright (C) 2007, 2008 Richard Kettlewell
  *
- * This program is free software; you can redistribute it and/or modify
+ * 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
+ * the Free Software Foundation, either version 3 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.
- *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307
- * USA
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
  */
 /** @file server/decode.c
  * @brief General-purpose decoder for use by speaker process
  */
 
-#include <config.h>
-#include "types.h"
-
-#include <getopt.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <errno.h>
-#include <stdlib.h>
-#include <locale.h>
-#include <assert.h>
-#include <fnmatch.h>
+#include "disorder-server.h"
+
 #include <mad.h>
-#include <ao/ao.h>
+#include <vorbis/vorbisfile.h>
+
+/* libFLAC has had an API change and stupidly taken away the old API */
+#if HAVE_FLAC_FILE_DECODER_H
+# include <FLAC/file_decoder.h>
+#else
+# include <FLAC/stream_decoder.h>
+#define FLAC__FileDecoder FLAC__StreamDecoder
+#define FLAC__FileDecoderState FLAC__StreamDecoderState
+#endif
+
+#include "wav.h"
+#include "speaker-protocol.h"
 
-#include "log.h"
-#include "syscalls.h"
-#include "defs.h"
 
 /** @brief Encoding lookup table type */
 struct decoder {
@@ -57,51 +55,71 @@ static FILE *outputfp;
 static const char *path;
 
 /** @brief Input buffer */
-static unsigned char buffer[1048576];
-
-/** @brief Open the input file */
-static void open_input(void) {
-  if((inputfd = open(path, O_RDONLY)) < 0)
-    fatal(errno, "opening %s", path);
-}
+static char input_buffer[1048576];
 
-/** @brief Fill the buffer
- * @return Number of bytes read
- */
-static size_t fill(void) {
-  int n = read(inputfd, buffer, sizeof buffer);
+/** @brief Number of bytes read into buffer */
+static int input_count;
 
-  if(n < 0)
-    fatal(errno, "reading from %s", path);
-  return n;
+/** @brief Write an 8-bit word */
+static inline void output_8(int n) {
+  if(putc(n, outputfp) < 0)
+    fatal(errno, "decoding %s: output error", path);
 }
 
 /** @brief Write a 16-bit word in bigendian format */
 static inline void output_16(uint16_t n) {
   if(putc(n >> 8, outputfp) < 0
-     || putc(n & 0xFF, outputfp) < 0)
+     || putc(n, outputfp) < 0)
+    fatal(errno, "decoding %s: output error", path);
+}
+
+/** @brief Write a 24-bit word in bigendian format */
+static inline void output_24(uint32_t n) {
+  if(putc(n >> 16, outputfp) < 0
+     || putc(n >> 8, outputfp) < 0
+     || putc(n, outputfp) < 0)
+    fatal(errno, "decoding %s: output error", path);
+}
+
+/** @brief Write a 32-bit word in bigendian format */
+static inline void output_32(uint32_t n) {
+  if(putc(n >> 24, outputfp) < 0
+     || putc(n >> 16, outputfp) < 0
+     || putc(n >> 8, outputfp) < 0
+     || putc(n, outputfp) < 0)
     fatal(errno, "decoding %s: output error", path);
 }
 
-/** @brief Write the header
- * If called more than once, either does nothing (if you kept the same
- * output encoding) or fails (if you changed it).
+/** @brief Write a block header
+ * @param rate Sample rate in Hz
+ * @param channels Channel count (currently only 1 or 2 supported)
+ * @param bits Bits per sample (must be a multiple of 8, no more than 64)
+ * @param nbytes Total number of data bytes
+ * @param endian @ref ENDIAN_BIG or @ref ENDIAN_LITTLE
+ *
+ * Checks that the sample format is a supported one (so other calls do not have
+ * to) and calls fatal() on error.
  */
 static void output_header(int rate,
 			  int channels,
-			  int bits) {
-  static int already_written_header;
-  struct ao_sample_format format;
-
-  if(!already_written_header) {
-     format.rate = rate;
-     format.bits = bits;
-     format.channels = channels;
-     format.byte_format = AO_FMT_BIG;
-     if(fwrite(&format, sizeof format, 1, outputfp) < 1)
-       fatal(errno, "decoding %s: writing format header", path);
-     already_written_header = 1;
-   }
+			  int bits,
+                          int nbytes,
+                          int endian) {
+  struct stream_header header;
+
+  if(bits <= 0 || bits % 8 || bits > 64)
+    fatal(0, "decoding %s: unsupported sample size %d bits", path, bits);
+  if(channels <= 0 || channels > 2)
+    fatal(0, "decoding %s: unsupported channel count %d", path, channels);
+  if(rate <= 0)
+    fatal(0, "decoding %s: nonsensical sample rate %dHz", path, rate);
+  header.rate = rate;
+  header.bits = bits;
+  header.channels = channels;
+  header.endian = endian;
+  header.nbytes = nbytes;
+  if(fwrite(&header, sizeof header, 1, outputfp) < 1)
+    fatal(errno, "decoding %s: writing format header", path);
 }
 
 /** @brief Dithering state
@@ -120,11 +138,11 @@ static inline unsigned long prng(unsigned long state)
 
 /** @brief Generic linear sample quantize and dither routine
  * Filched from mpg321, which credits it to Robert Leslie */
-#define bits 16
 static long audio_linear_dither(mad_fixed_t sample,
 				struct audio_dither *dither) {
   unsigned int scalebits;
   mad_fixed_t output, mask, rnd;
+  const int bits = 16;
 
   enum {
     MIN = -MAD_F_ONE,
@@ -172,7 +190,6 @@ static long audio_linear_dither(mad_fixed_t sample,
   /* scale */
   return output >> scalebits;
 }
-#undef bits
 
 /** @brief MP3 output callback */
 static enum mad_flow mp3_output(void attribute((unused)) *data,
@@ -184,7 +201,9 @@ static enum mad_flow mp3_output(void attribute((unused)) *data,
 
   output_header(header->samplerate,
 		pcm->channels,
-		16);
+		16,
+                2 * pcm->channels * pcm->length,
+                ENDIAN_BIG);
   switch(pcm->channels) {
   case 1:
     while(n--)
@@ -196,8 +215,6 @@ static enum mad_flow mp3_output(void attribute((unused)) *data,
       output_16(audio_linear_dither(*r++, rd));
     }
     break;
-  default:
-    fatal(0, "decoding %s: unsupported channel count %d", path, pcm->channels);
   }
   return MAD_FLOW_CONTINUE;
 }
@@ -205,30 +222,42 @@ static enum mad_flow mp3_output(void attribute((unused)) *data,
 /** @brief MP3 input callback */
 static enum mad_flow mp3_input(void attribute((unused)) *data,
 			       struct mad_stream *stream) {
-  const size_t n = fill();
-  fprintf(stderr, "n=%zu\n", n);
-  if(!n)
+  int used, remain, n;
+
+  /* libmad requires its caller to do ALL the buffering work, including coping
+   * with partial frames.  Given that it appears to be completely undocumented
+   * you could perhaps be forgiven for not discovering this...  */
+  if(input_count) {
+    /* Compute total number of bytes consumed */
+    used = (char *)stream->next_frame - input_buffer;
+    /* Compute number of bytes left to consume */
+    remain = input_count - used;
+    memmove(input_buffer, input_buffer + used, remain);
+  } else {
+    remain = 0;
+  }
+  /* Read new data */
+  n = read(inputfd, input_buffer + remain, (sizeof input_buffer) - remain);
+  if(n < 0)
+    fatal(errno, "reading from %s", path);
+  /* Compute total number of bytes available */
+  input_count = remain + n;
+  if(input_count)
+    mad_stream_buffer(stream, (unsigned char *)input_buffer, input_count);
+  if(n)
+    return MAD_FLOW_CONTINUE;
+  else
     return MAD_FLOW_STOP;
-  mad_stream_buffer(stream, buffer, n);
-  return MAD_FLOW_CONTINUE;
 }
 
-
 /** @brief MP3 error callback */
 static enum mad_flow mp3_error(void attribute((unused)) *data,
 			       struct mad_stream *stream,
 			       struct mad_frame attribute((unused)) *frame) {
-  error(0, "decoding %s: %s (%#04x)",
-	path, mad_stream_errorstr(stream), stream->error);
-  return MAD_FLOW_CONTINUE;
-}
-
-/** @brief MP3 header callback */
-static enum mad_flow mp3_header(void attribute((unused)) *data,
-				struct mad_header const *header) {
-  output_header(header->samplerate,
-		MAD_NCHANNELS(header),
-		16);
+  if(0)
+    /* Just generates pointless verbosity l-( */
+    error(0, "decoding %s: %s (%#04x)",
+          path, mad_stream_errorstr(stream), stream->error);
   return MAD_FLOW_CONTINUE;
 }
 
@@ -236,18 +265,148 @@ static enum mad_flow mp3_header(void attribute((unused)) *data,
 static void decode_mp3(void) {
   struct mad_decoder mad[1];
 
-  open_input();
-  mad_decoder_init(mad, 0/*data*/, mp3_input, mp3_header, 0/*filter*/,
+  if((inputfd = open(path, O_RDONLY)) < 0)
+    fatal(errno, "opening %s", path);
+  mad_decoder_init(mad, 0/*data*/, mp3_input, 0/*header*/, 0/*filter*/,
 		   mp3_output, mp3_error, 0/*message*/);
   if(mad_decoder_run(mad, MAD_DECODER_MODE_SYNC))
     exit(1);
   mad_decoder_finish(mad);
 }
 
+/** @brief OGG decoder */
+static void decode_ogg(void) {
+  FILE *fp;
+  OggVorbis_File vf[1];
+  int err;
+  long n;
+  int bitstream;
+  vorbis_info *vi;
+
+  if(!(fp = fopen(path, "rb")))
+    fatal(errno, "cannot open %s", path);
+  /* There doesn't seem to be any standard function for mapping the error codes
+   * to strings l-( */
+  if((err = ov_open(fp, vf, 0/*initial*/, 0/*ibytes*/)))
+    fatal(0, "ov_fopen %s: %d", path, err);
+  if(!(vi = ov_info(vf, 0/*link*/)))
+    fatal(0, "ov_info %s: failed", path);
+  while((n = ov_read(vf, input_buffer, sizeof input_buffer, 1/*bigendianp*/,
+                     2/*bytes/word*/, 1/*signed*/, &bitstream))) {
+    if(n < 0)
+      fatal(0, "ov_read %s: %ld", path, n);
+    if(bitstream > 0)
+      fatal(0, "only single-bitstream ogg files are supported");
+    output_header(vi->rate, vi->channels, 16/*bits*/, n, ENDIAN_BIG);
+    if(fwrite(input_buffer, 1, n, outputfp) < (size_t)n)
+      fatal(errno, "decoding %s: writing sample data", path);
+  }
+}
+
+/** @brief Sample data callback used by decode_wav() */
+static int wav_write(struct wavfile attribute((unused)) *f,
+                     const char *data,
+                     size_t nbytes,
+                     void attribute((unused)) *u) {
+  if(fwrite(data, 1, nbytes, outputfp) < nbytes)
+    fatal(errno, "decoding %s: writing sample data", path);
+  return 0;
+}
+
+/** @brief WAV file decoder */
+static void decode_wav(void) {
+  struct wavfile f[1];
+  int err;
+
+  if((err = wav_init(f, path)))
+    fatal(err, "opening %s", path);
+  output_header(f->rate, f->channels, f->bits, f->datasize, ENDIAN_LITTLE);
+  if((err = wav_data(f, wav_write, 0)))
+    fatal(err, "error decoding %s", path);
+}
+
+/** @brief Metadata callback for FLAC decoder
+ *
+ * This is a no-op here.
+ */
+static void flac_metadata(const FLAC__FileDecoder attribute((unused)) *decoder,
+			  const FLAC__StreamMetadata attribute((unused)) *metadata,
+			  void attribute((unused)) *client_data) {
+}
+
+/** @brief Error callback for FLAC decoder */
+static void flac_error(const FLAC__FileDecoder attribute((unused)) *decoder,
+		       FLAC__StreamDecoderErrorStatus status,
+		       void attribute((unused)) *client_data) {
+  fatal(0, "error decoding %s: %s", path,
+        FLAC__StreamDecoderErrorStatusString[status]);
+}
+
+/** @brief Write callback for FLAC decoder */
+static FLAC__StreamDecoderWriteStatus flac_write
+    (const FLAC__FileDecoder attribute((unused)) *decoder,
+     const FLAC__Frame *frame,
+     const FLAC__int32 *const buffer[],
+     void attribute((unused)) *client_data) {
+  size_t n, c;
+
+  output_header(frame->header.sample_rate,
+                frame->header.channels,
+                frame->header.bits_per_sample,
+                (frame->header.channels * frame->header.blocksize
+                 * frame->header.bits_per_sample) / 8,
+                ENDIAN_BIG);
+  for(n = 0; n < frame->header.blocksize; ++n) {
+    for(c = 0; c < frame->header.channels; ++c) {
+      switch(frame->header.bits_per_sample) {
+      case 8: output_8(buffer[c][n]); break;
+      case 16: output_16(buffer[c][n]); break;
+      case 24: output_24(buffer[c][n]); break;
+      case 32: output_32(buffer[c][n]); break;
+      }
+    }
+  }
+  return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+}
+
+
+/** @brief FLAC file decoder */
+static void decode_flac(void) {
+#if HAVE_FLAC_FILE_DECODER_H
+  FLAC__FileDecoder *fd = 0;
+  FLAC__FileDecoderState fs;
+
+  if(!(fd = FLAC__file_decoder_new()))
+    fatal(0, "FLAC__file_decoder_new failed");
+  if(!(FLAC__file_decoder_set_filename(fd, path)))
+    fatal(0, "FLAC__file_set_filename failed");
+  FLAC__file_decoder_set_metadata_callback(fd, flac_metadata);
+  FLAC__file_decoder_set_error_callback(fd, flac_error);
+  FLAC__file_decoder_set_write_callback(fd, flac_write);
+  if((fs = FLAC__file_decoder_init(fd)))
+    fatal(0, "FLAC__file_decoder_init: %s", FLAC__FileDecoderStateString[fs]);
+  FLAC__file_decoder_process_until_end_of_file(fd);
+#else
+  FLAC__StreamDecoder *sd = 0;
+  FLAC__StreamDecoderInitStatus is;
+
+  if((is = FLAC__stream_decoder_init_file(sd, path, flac_write, flac_metadata,
+                                          flac_error, 0)))
+    fatal(0, "FLAC__stream_decoder_init_file %s: %s",
+          path, FLAC__StreamDecoderInitStatusString[is]);
+#endif
+}
+
 /** @brief Lookup table of decoders */
 static const struct decoder decoders[] = {
   { "*.mp3", decode_mp3 },
   { "*.MP3", decode_mp3 },
+  { "*.ogg", decode_ogg },
+  { "*.OGG", decode_ogg },
+  { "*.flac", decode_flac },
+  { "*.FLAC", decode_flac },
+  { "*.wav", decode_wav },
+  { "*.WAV", decode_wav },
   { 0, 0 }
 };
 
@@ -271,13 +430,6 @@ static void help(void) {
   exit(0);
 }
 
-/* Display version number and terminate. */
-static void version(void) {
-  xprintf("disorder-decode version %s\n", disorder_version_string);
-  xfclose(stdout);
-  exit(0);
-}
-
 int main(int argc, char **argv) {
   int n;
   const char *e;
@@ -287,7 +439,7 @@ int main(int argc, char **argv) {
   while((n = getopt_long(argc, argv, "hV", options, 0)) >= 0) {
     switch(n) {
     case 'h': help();
-    case 'V': version();
+    case 'V': version("disorder-decode");
     default: fatal(0, "invalid option");
     }
   }