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-rw-r--r--src/post/audio/upmix.c100
1 files changed, 68 insertions, 32 deletions
diff --git a/src/post/audio/upmix.c b/src/post/audio/upmix.c
index f0e40c83a..168411fd3 100644
--- a/src/post/audio/upmix.c
+++ b/src/post/audio/upmix.c
@@ -23,7 +23,7 @@
* process. It simply paints the screen a solid color and rotates through
* colors on each iteration.
*
- * $Id: upmix.c,v 1.3 2004/05/16 02:56:35 jcdutton Exp $
+ * $Id: upmix.c,v 1.4 2004/05/16 14:44:42 jcdutton Exp $
*
*/
@@ -158,9 +158,10 @@ static int upmix_port_open(xine_audio_port_t *port_gen, xine_stream_t *stream,
this->ratio = (double)FOO_WIDTH/(double)FOO_HEIGHT;
this->channels = _x_ao_mode2channels(mode);
/* FIXME: Handle all desired output formats */
- if (capabilities & AO_CAP_MODE_5_1CHANNEL) {
+ if ((capabilities & AO_CAP_MODE_5_1CHANNEL) && (capabilities & AO_CAP_32BITS)) {
this->channels_out=6;
mode = AO_CAP_MODE_5_1CHANNEL;
+ bits = 32; /* Upmix to Floats */
} else {
this->channels_out=2;
}
@@ -204,36 +205,74 @@ static void upmix_port_close(xine_audio_port_t *port_gen, xine_stream_t *stream
_x_post_dec_usage(port);
}
-static int upmix_frames_2to51_16bit(uint8_t *dst8, uint8_t *src8, int num_frames, af_sub_t *sub) {
- int16_t *dst=(int16_t *)dst8;
- int16_t *src=(int16_t *)src8;
-
- int frame;
- int bytes_per_sample=1; /* Actually int16 per sample here */
+static int upmix_frames_2to51_any_to_float(uint8_t *dst8, uint8_t *src8, int num_frames, int step_channel_in, af_sub_t *sub) {
+ float *dst=(float *)dst8;
+ int16_t *src16=(int16_t *)src8;
+ float *src_float=(float *)src8;
int src_num_channels=2;
int dst_num_channels=6;
int src_frame;
int dst_frame;
+ int32_t sample24;
float sample;
- int32_t sum;
+ float left;
+ float right;
+ float sum;
+ int frame;
+ int src_units_per_sample=1;
+ if (step_channel_in == 3) src_units_per_sample=step_channel_in; /* Special handling for 24 bit 3byte input */
for (frame=0;frame < num_frames; frame++) {
- dst_frame=frame*dst_num_channels*bytes_per_sample;
- src_frame=frame*src_num_channels*bytes_per_sample;
- dst[dst_frame] = src[src_frame];
- dst[dst_frame+(1*bytes_per_sample)] = src[src_frame+(1*bytes_per_sample)];
+ dst_frame=frame*dst_num_channels;
+ src_frame=frame*src_num_channels*src_units_per_sample;
+ switch (step_channel_in) {
+ case 1:
+ left = src8[src_frame];
+ left = (left - 128 ) / 128; /* FIXME: Need to verify this is correct */
+ right = src8[src_frame+1];
+ right = (right - 128) / 128;
+ break;
+ case 2:
+ left = (1.0/SHRT_MAX)*((float)src16[src_frame]);
+ right = (1.0/SHRT_MAX)*((float)src16[src_frame+1]);
+ break;
+ case 3:
+#ifdef WORDS_BIGENDIAN
+ sample24 = (src8[src_frame] << 24) | (src8[src_frame+1] << 16) | ( src8[src_frame+2] << 8);
+#else
+ sample24 = (src8[src_frame] << 8) | (src8[src_frame+1] << 16) | ( src8[src_frame+2] << 24);
+#endif
+ left = (1.0/INT32_MAX)*((float)sample24);
+#ifdef WORDS_BIGENDIAN
+ sample24 = (src8[src_frame+3] << 24) | (src8[src_frame+4] << 16) | ( src8[src_frame+5] << 8);
+#else
+ sample24 = (src8[src_frame+3] << 8) | (src8[src_frame+4] << 16) | ( src8[src_frame+5] << 24);
+#endif
+ right = (1.0/INT32_MAX)*((float)sample24);
+ break;
+ case 4:
+ left = src_float[src_frame];
+ right = src_float[src_frame+1];
+ break;
+ default:
+ left = right = 0.0;
+ }
+
+ dst[dst_frame] = left;
+ dst[dst_frame+1] = right;
/* try a bit of dolby */
/* FIXME: Dobly surround is a bit more complicated than this, but this is a start. */
- dst[dst_frame+(2*bytes_per_sample)] = (src[src_frame] - src[src_frame+(1*bytes_per_sample)]) / 2;
- dst[dst_frame+(3*bytes_per_sample)] = (src[src_frame] - src[src_frame+(1*bytes_per_sample)]) / 2;
- sum = ((int32_t)src[src_frame] + (int32_t)src[src_frame+(1*bytes_per_sample)]) / 2;
- dst[dst_frame+(4*bytes_per_sample)] = (int16_t)sum;
+ dst[dst_frame+2] = (left - right) / 2;
+ dst[dst_frame+3] = (left - right) / 2;
+ sum = (left + right) / 2;
+ dst[dst_frame+4] = sum;
/* Create the LFE channel using a low pass filter */
/* filter feature ported from mplayer */
- sample = (1.0/SHRT_MAX)*((float)sum);
+ sample = sum;
IIR(sample * sub->k, sub->w[0], sub->q[0], sample);
IIR(sample , sub->w[1], sub->q[1], sample);
- dst[dst_frame+(5*bytes_per_sample)] = (int16_t)(sample * SHRT_MAX);
+ dst[dst_frame+5] = sample;
+
}
return frame;
}
@@ -243,15 +282,10 @@ static void upmix_port_put_buffer (xine_audio_port_t *port_gen,
post_audio_port_t *port = (post_audio_port_t *)port_gen;
post_plugin_upmix_t *this = (post_plugin_upmix_t *)port->post;
- vo_frame_t *frame;
- int16_t *data;
- int8_t *data8;
- int samples_used = 0;
- int64_t pts = buf->vpts;
- int i, j;
int src_step_frame;
int dst_step_frame;
- int step_channel;
+ int step_channel_in;
+ int step_channel_out;
uint8_t *data8src;
uint8_t *data8dst;
int num_bytes;
@@ -270,13 +304,14 @@ static void upmix_port_put_buffer (xine_audio_port_t *port_gen,
/* FIXME: The audio buffer should contain this info.
* We should not have to get it from the open call.
*/
- this->buf->format.bits = port->bits;
+ this->buf->format.bits = 32; /* Upmix to floats */
this->buf->format.rate = port->rate;
this->buf->format.mode = AO_CAP_MODE_5_1CHANNEL;
_x_extra_info_merge( this->buf->extra_info, buf->extra_info);
- step_channel = this->buf->format.bits>>3;
- dst_step_frame = this->channels_out*step_channel;
- src_step_frame = this->channels*step_channel;
+ step_channel_in = port->bits>>3;
+ step_channel_out = this->buf->format.bits>>3;
+ dst_step_frame = this->channels_out*step_channel_out;
+ src_step_frame = this->channels*step_channel_in;
num_bytes=(buf->num_frames-num_frames_processed)*dst_step_frame;
if (num_bytes > this->buf->mem_size) {
num_bytes = this->buf->mem_size;
@@ -285,7 +320,7 @@ static void upmix_port_put_buffer (xine_audio_port_t *port_gen,
data8src=(int8_t*)buf->mem;
data8src+=num_frames_processed*src_step_frame;
data8dst=(int8_t*)this->buf->mem;
- num_frames_done = upmix_frames_2to51_16bit(data8dst, data8src, num_frames, this->sub);
+ num_frames_done = upmix_frames_2to51_any_to_float(data8dst, data8src, num_frames, step_channel_in, this->sub);
this->buf->num_frames = num_frames_done;
num_frames_processed+= num_frames_done;
/* pass data to original port */
@@ -387,7 +422,8 @@ static void *upmix_init_plugin(xine_t *xine, void *data)
}
/* plugin catalog information */
-post_info_t upmix_special_info = { XINE_POST_TYPE_AUDIO_VISUALIZATION };
+/* post_info_t upmix_special_info = { XINE_POST_TYPE_AUDIO_VISUALIZATION }; */
+post_info_t upmix_special_info = { XINE_POST_TYPE_AUDIO_FILTER };
plugin_info_t xine_plugin_info[] = {
/* type, API, "name", version, special_info, init_function */
generated by cgit v1.2.3 (git 2.25.1) at 2025-04-24 14:47:30 +0000