X-Git-Url: https://git.distorted.org.uk/~mdw/disorder/blobdiff_plain/39492555cd1d64e2f8a3474038f61d28c5c56de5..ce6c36be6c2df99afd01a7a602debb321322e113:/server/decode.c?ds=sidebyside

diff --git a/server/decode.c b/server/decode.c
index c63ca20..d1be348 100644
--- a/server/decode.c
+++ b/server/decode.c
@@ -33,10 +33,12 @@
 #include <assert.h>
 #include <fnmatch.h>
 #include <mad.h>
+#include <vorbis/vorbisfile.h>
 
 #include "log.h"
 #include "syscalls.h"
 #include "defs.h"
+#include "wav.h"
 #include "speaker-protocol.h"
 
 /** @brief Encoding lookup table type */
@@ -57,7 +59,7 @@ static FILE *outputfp;
 static const char *path;
 
 /** @brief Input buffer */
-static unsigned char buffer[1048576];
+static char buffer[1048576];
 
 /** @brief Open the input file */
 static void open_input(void) {
@@ -90,13 +92,14 @@ static inline void output_16(uint16_t n) {
 static void output_header(int rate,
 			  int channels,
 			  int bits,
-                          int nbytes) {
+                          int nbytes,
+                          int endian) {
   struct stream_header header;
 
   header.rate = rate;
   header.bits = bits;
   header.channels = channels;
-  header.endian = ENDIAN_BIG;
+  header.endian = endian;
   header.nbytes = nbytes;
   if(fwrite(&header, sizeof header, 1, outputfp) < 1)
     fatal(errno, "decoding %s: writing format header", path);
@@ -183,7 +186,8 @@ static enum mad_flow mp3_output(void attribute((unused)) *data,
   output_header(header->samplerate,
 		pcm->channels,
 		16,
-                2 * pcm->channels * pcm->length);
+                2 * pcm->channels * pcm->length,
+                ENDIAN_BIG);
   switch(pcm->channels) {
   case 1:
     while(n--)
@@ -205,10 +209,10 @@ static enum mad_flow mp3_output(void attribute((unused)) *data,
 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)
     return MAD_FLOW_STOP;
-  mad_stream_buffer(stream, buffer, n);
+  mad_stream_buffer(stream, (unsigned char *)buffer, n);
   return MAD_FLOW_CONTINUE;
 }
 
@@ -236,10 +240,67 @@ static void decode_mp3(void) {
   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, buffer, sizeof 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(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);
+  if(f->bits % 8)
+    fatal(err, "%s: unsupported byte size %d", path, f->bits);
+  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 Lookup table of decoders */
 static const struct decoder decoders[] = {
   { "*.mp3", decode_mp3 },
   { "*.MP3", decode_mp3 },
+  { "*.ogg", decode_ogg },
+  { "*.OGG", decode_ogg },
+  { "*.wav", decode_wav },
+  { "*.WAV", decode_wav },
   { 0, 0 }
 };