Channel count first seems...nicer, somehow.

5 files changed, 39 insertions, 39 deletions

diff --git a/src/audio.c b/src/audio.c
index 641bb93..f5dc5ee 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -8,23 +8,23 @@ static xas_audio_stream *stream_new(enum xas_audio_stream_type type,
                                       void *callback,
                                       xas_audio_cleanup cleanup,
                                       void *ctx,
+                                      size_t channels,
                                       size_t sample_size,
                                       size_t sample_rate,
-                                      size_t channels,
                                       size_t buffer_size) {
     xas_audio_stream *stream;
 
     size_t total = sizeof(xas_audio_stream)
-                 + sample_size * channels * buffer_size;
+                 + channels * sample_size * buffer_size;
 
     if ((stream = malloc(total)) == NULL) {
         goto error_malloc_stream;
     }
 
     stream->type         = type;
+    stream->channels     = channels;
     stream->sample_size  = sample_size;
     stream->sample_rate  = sample_rate;
-    stream->channels     = channels;
     stream->buffer_size  = buffer_size;
     stream->buffer_count = 0;
     stream->callback     = callback;
@@ -40,34 +40,34 @@ error_malloc_stream:
 xas_audio_stream *xas_audio_stream_new_sink(xas_audio_drain drain,
                                                 xas_audio_cleanup cleanup,
                                                 void *ctx,
+                                                size_t channels,
                                                 size_t sample_size,
                                                 size_t sample_rate,
-                                                size_t channels,
                                                 size_t buffer_size) {
     return stream_new(XAS_AUDIO_STREAM_SINK,
                       drain,
                       cleanup,
                       ctx,
+                      channels,
                       sample_size,
                       sample_rate,
-                      channels,
                       buffer_size);
 }
 
 xas_audio_stream *xas_audio_stream_new_source(xas_audio_fill fill,
                                                   xas_audio_cleanup cleanup,
                                                   void *ctx,
+                                                  size_t channels,
                                                   size_t sample_size,
                                                   size_t sample_rate,
-                                                  size_t channels,
                                                   size_t buffer_size) {
     return stream_new(XAS_AUDIO_STREAM_SOURCE,
                       fill,
                       cleanup,
                       ctx,
+                      channels,
                       sample_size,
                       sample_rate,
-                      channels,
                       buffer_size);
 }
 
@@ -100,7 +100,7 @@ error_stream_drain:
 }
 
 static inline void *ptr(xas_audio_stream *stream, void *buf, size_t index) {
-    return ((uint8_t *)buf) + stream->sample_size * stream->channels * index;
+    return ((uint8_t *)buf) + stream->channels * stream->sample_size * index;
 }
 
 ssize_t xas_audio_stream_write(xas_audio_stream *stream,
@@ -118,7 +118,7 @@ ssize_t xas_audio_stream_write(xas_audio_stream *stream,
 
         memcpy(ptr(stream, stream + 1, stream->buffer_count),
                ptr(stream, samples,    index_i),
-               remaining * stream->sample_size * stream->channels);
+               remaining * stream->channels * stream->sample_size);
 
         if (stream->drain(stream->ctx,
                           ptr(stream, stream + 1, 0),
@@ -159,7 +159,7 @@ ssize_t xas_audio_stream_write(xas_audio_stream *stream,
              */
             memcpy(ptr(stream, stream + 1, stream->buffer_count),
                    ptr(stream, samples,    index_i),
-                   remaining * stream->sample_size * stream->channels);
+                   remaining * stream->channels * stream->sample_size);
 
             index_i += remaining;
 
diff --git a/src/mixer.c b/src/mixer.c
index 32c557d..9706b09 100644
--- a/src/mixer.c
+++ b/src/mixer.c
@@ -71,7 +71,7 @@ static ssize_t mixer_fill(xas_mixer *mixer,
 
     memset(mixer->buf,
            '\0',
-           output->sample_size * output->channels * output->buffer_size);
+           output->channels * output->sample_size * output->buffer_size);
 
     while (input) {
         void (*mixer_apply)(int16_t *,
@@ -120,7 +120,7 @@ static ssize_t mixer_fill(xas_mixer *mixer,
 
     memcpy(samples,
            mixer->buf,
-           total * output->sample_size * output->channels);
+           total * output->channels * output->sample_size);
 
     return total;
 
@@ -129,9 +129,9 @@ error_audio_read_stream:
     return -1;
 }
 
-xas_mixer *xas_mixer_new(size_t sample_size,
+xas_mixer *xas_mixer_new(size_t channels,
+                             size_t sample_size,
                              size_t sample_rate,
-                             size_t channels,
                              size_t buffer_size) {
     xas_mixer *mixer;
 
@@ -139,16 +139,16 @@ xas_mixer *xas_mixer_new(size_t sample_size,
         goto error_malloc_mixer;
     }
 
-    if ((mixer->buf = malloc(sample_size * channels * buffer_size)) == NULL) {
+    if ((mixer->buf = malloc(channels * sample_size * buffer_size)) == NULL) {
         goto error_malloc_buf;
     }
 
     if ((mixer->output = xas_audio_stream_new_source((xas_audio_fill)mixer_fill,
                                                        NULL,
                                                        mixer,
+                                                       channels,
                                                        sample_size,
                                                        sample_rate,
-                                                       channels,
                                                        buffer_size)) == NULL) {
         goto error_audio_stream_new_source;
     }
diff --git a/src/riff.c b/src/riff.c
index a920d10..0184a30 100644
--- a/src/riff.c
+++ b/src/riff.c
@@ -12,9 +12,9 @@ struct _xas_riff {
     int fd;
     size_t size;
 
-    size_t sample_size,
-           sample_rate,
-           channels;
+    size_t channels,
+           sample_size,
+           sample_rate;
 };
 
 static int header_write(xas_riff *riff) {
@@ -70,9 +70,9 @@ error_io:
 }
 
 static xas_riff *new_file(const char *path,
+                            size_t channels,
                             size_t sample_size,
                             size_t sample_rate,
-                            size_t channels,
                             int flags) {
     xas_riff *riff;
 
@@ -88,9 +88,9 @@ static xas_riff *new_file(const char *path,
 
     riff->file        = path;
     riff->size        = 0;
+    riff->channels    = channels;
     riff->sample_size = sample_size;
     riff->sample_rate = sample_rate;
-    riff->channels    = channels;
 
     if (header_write(riff) < 0) {
         goto error_header_write;
@@ -167,9 +167,9 @@ static int wave_header_parse(xas_riff *riff,
         goto error_invalid_wave_data_format_id;
     }
 
+    riff->channels    = header->wave.channels;
     riff->sample_size = header->wave.sample_size;
     riff->sample_rate = header->wave.sample_rate;
-    riff->channels    = header->wave.channels;
 
     return 0;
 
@@ -199,9 +199,9 @@ static xas_riff *open_fd(int fd) {
     riff->file        = NULL;
     riff->fd          = fd;
     riff->size        = 0;
+    riff->channels    = 0;
     riff->sample_size = 0;
     riff->sample_rate = 0;
-    riff->channels    = 0;
 
     if ((readlen = read(fd, &header, sizeof(header))) < 0) {
         goto error_read;
@@ -246,9 +246,9 @@ static xas_riff *open_file(const char *path, int flags) {
 
     riff->file        = path;
     riff->size        = 0;
+    riff->channels    = 0;
     riff->sample_size = 0;
     riff->sample_rate = 0;
-    riff->channels    = 0;
 
     if (lseek(riff->fd, 0, SEEK_SET) < 0) {
         goto error_lseek;
@@ -299,8 +299,8 @@ static ssize_t audio_drain(xas_riff *riff,
                            void *samples,
                            size_t count,
                            xas_audio_stream *stream) {
-    size_t len = count * stream->sample_size
-                       * stream->channels;
+    size_t len = count * stream->channels
+                       * stream->sample_size;
 
     ssize_t wrlen;
 
@@ -310,7 +310,7 @@ static ssize_t audio_drain(xas_riff *riff,
 
     riff->size += wrlen;
 
-    return wrlen / stream->sample_size / stream->channels;
+    return wrlen / stream->channels / stream->sample_size;
 
 error_write:
     return -1;
@@ -320,8 +320,8 @@ static ssize_t audio_fill(xas_riff *riff,
                           void *samples,
                           size_t count,
                           xas_audio_stream *stream) {
-    size_t len = count * stream->sample_size
-                       * stream->channels;
+    size_t len = count * stream->channels
+                       * stream->sample_size;
 
     ssize_t rdlen;
 
@@ -329,24 +329,24 @@ static ssize_t audio_fill(xas_riff *riff,
         goto error_read;
     }
 
-    return rdlen / stream->sample_size / stream->channels;
+    return rdlen / stream->channels / stream->sample_size;
 
 error_read:
     return -1;
 }
 
 xas_audio_stream *xas_riff_new_file(const char *path,
+                                        size_t channels,
                                         size_t sample_size,
                                         size_t sample_rate,
-                                        size_t channels,
                                         int flags) {
     xas_audio_stream *stream;
     xas_riff *riff;
 
     if ((riff = new_file(path,
+                         channels,
                          sample_size,
                          sample_rate,
-                         channels,
                          flags)) == NULL) {
         goto error_new_file;
     }
@@ -355,9 +355,9 @@ xas_audio_stream *xas_riff_new_file(const char *path,
         if ((stream = xas_audio_stream_new_sink((xas_audio_drain)audio_drain,
                                                   (xas_audio_cleanup)close_file,
                                                   riff,
+                                                  channels,
                                                   sample_size,
                                                   sample_rate,
-                                                  channels,
                                                   4096)) == NULL) {
             goto error_audio_stream_new_sink;
         }
@@ -365,9 +365,9 @@ xas_audio_stream *xas_riff_new_file(const char *path,
         if ((stream = xas_audio_stream_new_source((xas_audio_fill)audio_fill,
                                                     (xas_audio_cleanup)close_file,
                                                     riff,
+                                                    channels,
                                                     sample_size,
                                                     sample_rate,
-                                                    channels,
                                                     4096)) == NULL) {
             goto error_audio_stream_new_sink;
         }
@@ -394,9 +394,9 @@ xas_audio_stream *xas_riff_open_file(const char *path, int flags) {
         if ((stream = xas_audio_stream_new_sink((xas_audio_drain)audio_drain,
                                                   (xas_audio_cleanup)close_file,
                                                   riff,
+                                                  riff->channels,
                                                   riff->sample_size,
                                                   riff->sample_rate,
-                                                  riff->channels,
                                                   4096)) == NULL) {
             goto error_audio_stream_new_sink;
         }
@@ -404,9 +404,9 @@ xas_audio_stream *xas_riff_open_file(const char *path, int flags) {
         if ((stream = xas_audio_stream_new_source((xas_audio_fill)audio_fill,
                                                     (xas_audio_cleanup)close_file,
                                                     riff,
+                                                    riff->channels,
                                                     riff->sample_size,
                                                     riff->sample_rate,
-                                                    riff->channels,
                                                     4096)) == NULL) {
             goto error_audio_stream_new_sink;
         }
@@ -432,9 +432,9 @@ xas_audio_stream *xas_riff_open_fd(int fd) {
     if ((stream = xas_audio_stream_new_source((xas_audio_fill)audio_fill,
                                                 (xas_audio_cleanup)close_file,
                                                 riff,
+                                                riff->channels,
                                                 riff->sample_size,
                                                 riff->sample_rate,
-                                                riff->channels,
                                                 4096)) == NULL) {
         goto error_audio_stream_new_sink;
     }
diff --git a/src/synth.c b/src/synth.c
index a482875..2e77748 100644
--- a/src/synth.c
+++ b/src/synth.c
@@ -45,9 +45,9 @@ xas_audio_stream *xas_synth_new(size_t sample_size,
     if ((stream = xas_audio_stream_new_source((xas_audio_fill)synth_fill,
                                                 (xas_audio_cleanup)synth_cleanup,
                                                 synth,
+                                                XAS_AUDIO_STREAM_MONO,
                                                 sample_size,
                                                 sample_rate,
-                                                XAS_AUDIO_STREAM_MONO,
                                                 buffer_size)) == NULL) {
         goto error_audio_stream_new_source;
     }
diff --git a/src/vox.c b/src/vox.c
index df3e084..7fec432 100644
--- a/src/vox.c
+++ b/src/vox.c
@@ -275,8 +275,8 @@ xas_audio_stream *xas_vox_stream_new(xas_vox *vox) {
     return xas_audio_stream_new_source((xas_audio_fill)vox_fill,
                                          (xas_audio_cleanup)vox_cleanup,
                                          vox,
+                                         XAS_AUDIO_STREAM_MONO,
                                          vox->sample_size,
                                          vox->sample_rate,
-                                         XAS_AUDIO_STREAM_MONO,
                                          vox->buffer_size);
 }