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[Gnash-commit] gnash server/parser/sound_definition.cpp server...
|
From: |
Martin Guy |
|
Subject: |
[Gnash-commit] gnash server/parser/sound_definition.cpp server... |
|
Date: |
Sun, 13 May 2007 12:20:37 +0000 |
CVSROOT: /sources/gnash
Module name: gnash
Changes by: Martin Guy <martinwguy> 07/05/13 12:20:37
Modified files:
server/parser : sound_definition.cpp
server/swf : tag_loaders.cpp
. : ChangeLog
Log message:
* server/parser/sound_definition.cpp: Move ADPCM code elsewhere
* server/swf/tag_loaders.cpp: Decode all linear formats into
native 16,
move ADPCM code here where it's used.
CVSWeb URLs:
http://cvs.savannah.gnu.org/viewcvs/gnash/server/parser/sound_definition.cpp?cvsroot=gnash&r1=1.2&r2=1.3
http://cvs.savannah.gnu.org/viewcvs/gnash/server/swf/tag_loaders.cpp?cvsroot=gnash&r1=1.97&r2=1.98
http://cvs.savannah.gnu.org/viewcvs/gnash/ChangeLog?cvsroot=gnash&r1=1.3175&r2=1.3176
Patches:
Index: server/parser/sound_definition.cpp
===================================================================
RCS file: /sources/gnash/gnash/server/parser/sound_definition.cpp,v
retrieving revision 1.2
retrieving revision 1.3
diff -u -b -r1.2 -r1.3
--- server/parser/sound_definition.cpp 27 Feb 2007 09:54:49 -0000 1.2
+++ server/parser/sound_definition.cpp 13 May 2007 12:20:36 -0000 1.3
@@ -124,213 +124,6 @@
}
}
- //
- // ADPCM
- //
-
-
- // Data from Alexis' SWF reference
- static int s_index_update_table_2bits[2] = { -1, 2 };
- static int s_index_update_table_3bits[4] = { -1, -1, 2, 4 };
- static int s_index_update_table_4bits[8] = { -1, -1, -1, -1, 2,
4, 6, 8 };
- static int s_index_update_table_5bits[16] = { -1, -1, -1, -1, -1,
-1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16 };
-
- static int* s_index_update_tables[4] = {
- s_index_update_table_2bits,
- s_index_update_table_3bits,
- s_index_update_table_4bits,
- s_index_update_table_5bits,
- };
-
- // Data from Jansen. http://homepages.cwi.nl/~jack/
- // Check out his Dutch retro punk songs, heh heh :)
- const int STEPSIZE_CT = 89;
- static int s_stepsize[STEPSIZE_CT] = {
- 7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
- 19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
- 50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
- 130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
- 337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
- 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
- 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
- 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
- 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
- };
-
-
- // Algo from
http://www.circuitcellar.com/pastissues/articles/richey110/text.htm
- // And also Jansen.
- // Here's another reference:
http://www.geocities.com/SiliconValley/8682/aud3.txt
- // Original IMA spec doesn't seem to be on the web :(
-
-
- // @@ lots of macros here! It seems that VC6 can't correctly
- // handle integer template args, although it's happy to
- // compile them?!
-
-// void DO_SAMPLE(int n_bits, int& sample, int& stepsize_index, int
raw_code)
-#define DO_SAMPLE(n_bits, sample, stepsize_index, raw_code)
\
- {
\
- assert(raw_code >= 0 && raw_code < (1 << n_bits));
\
-
\
- static const int HI_BIT = (1 << (n_bits - 1));
\
- int* index_update_table = s_index_update_tables[n_bits - 2];
\
-
\
- /* Core of ADPCM. */
\
-
\
- int code_mag = raw_code & (HI_BIT - 1);
\
- bool code_sign_bit = (raw_code & HI_BIT) ? 1 : 0;
\
- int mag = (code_mag << 1) + 1; /* shift in LSB (they
do this so that pos & neg zero are different)*/ \
-
\
- int stepsize = s_stepsize[stepsize_index];
\
-
\
- /* Compute the new sample. It's the predicted value
*/ \
- /* (i.e. the previous value), plus a delta. The delta
*/ \
- /* comes from the code times the stepsize. going for
*/ \
- /* something like: delta = stepsize * (code * 2 + 1) >>
code_bits */ \
- int delta = (stepsize * mag) >> (n_bits - 1);
\
- if (code_sign_bit) delta = -delta;
\
-
\
- sample += delta;
\
- sample = iclamp(sample, -32768, 32767);
\
-
\
- /* Update our stepsize index. Use a lookup table. */
\
- stepsize_index += index_update_table[code_mag];
\
- stepsize_index = iclamp(stepsize_index, 0, STEPSIZE_CT - 1);
\
- }
-
-
- class in_stream
- {
- public:
- const unsigned char* m_in_data;
- int m_current_bits;
- int m_unused_bits;
-
- in_stream(const unsigned char* data)
- :
- m_in_data(data),
- m_current_bits(0),
- m_unused_bits(0)
- {
- }
- };
-
-
-// void DO_MONO_BLOCK(int16_t** out_data, int n_bits, int sample_count,
stream* in, int sample, int stepsize_index)
-#define DO_MONO_BLOCK(out_data, n_bits, sample_count, in, sample,
stepsize_index) \
- {
\
- /* First sample doesn't need to be decompressed. */
\
- sample_count--;
\
- *(*out_data)++ = (int16_t) sample;
\
-
\
- while (sample_count--)
\
- {
\
- int raw_code = in->read_uint(n_bits);
\
- DO_SAMPLE(n_bits, sample, stepsize_index, raw_code);
/* sample & stepsize_index are in/out params */ \
- *(*out_data)++ = (int16_t) sample;
\
- }
\
- }
-
-
-// void do_stereo_block(
-// int16_t** out_data, // in/out param
-// int n_bits,
-// int sample_count,
-// stream* in,
-// int left_sample,
-// int left_stepsize_index,
-// int right_sample,
-// int right_stepsize_index
-// )
-#define DO_STEREO_BLOCK(out_data, n_bits, sample_count, in, left_sample,
left_stepsize_index, right_sample, right_stepsize_index) \
- /* Uncompress 4096 stereo sample pairs of ADPCM. */
\
- {
\
- /* First samples don't need to be decompressed. */
\
- sample_count--;
\
- *(*out_data)++ = (int16_t) left_sample;
\
- *(*out_data)++ = (int16_t) right_sample;
\
-
\
- while (sample_count--)
\
- {
\
- int left_raw_code = in->read_uint(n_bits);
\
- DO_SAMPLE(n_bits, left_sample, left_stepsize_index,
left_raw_code); \
- *(*out_data)++ = (int16_t) left_sample;
\
-
\
- int right_raw_code = in->read_uint(n_bits);
\
- DO_SAMPLE(n_bits, right_sample, right_stepsize_index,
right_raw_code); \
- *(*out_data)++ = (int16_t) right_sample;
\
- }
\
- }
-
-
- // Utility function: uncompress ADPCM data from in stream to
- // out_data[]. The output buffer must have (sample_count*2)
- // bytes for mono, or (sample_count*4) bytes for stereo.
- void sound_handler::adpcm_expand(
- void* out_data_void,
- stream* in,
- int sample_count, // in stereo, this is number of *pairs*
of samples
- bool stereo)
- {
- int16_t* out_data = (int16_t*) out_data_void;
-
- // Read header.
- int n_bits = in->read_uint(2) + 2; // 2 to 5 bits
-
- while (sample_count)
- {
- // Read initial sample & index values.
- int sample = in->read_sint(16);
-
- int stepsize_index = in->read_uint(6);
- assert(STEPSIZE_CT >= (1 << 6)); // ensure we
don't need to clamp.
-
- int samples_this_block = imin(sample_count, 4096);
- sample_count -= samples_this_block;
-
- if (stereo == false)
- {
-#define DO_MONO(n) DO_MONO_BLOCK(&out_data, n, samples_this_block, in, sample,
stepsize_index)
-
- switch (n_bits)
- {
- default: assert(0); break;
- case 2: DO_MONO(2); break;
- case 3: DO_MONO(3); break;
- case 4: DO_MONO(4); break;
- case 5: DO_MONO(5); break;
- }
- }
- else
- {
- // Stereo.
-
- // Got values for left channel; now get initial
sample
- // & index for right channel.
- int right_sample = in->read_sint(16);
-
- int right_stepsize_index = in->read_uint(6);
- assert(STEPSIZE_CT >= (1 << 6)); //
ensure we don't need to clamp.
-
-#define DO_STEREO(n) \
- DO_STEREO_BLOCK( \
- &out_data, n, samples_this_block, \
- in, sample, stepsize_index, \
- right_sample, right_stepsize_index)
-
- switch (n_bits)
- {
- default: assert(0); break;
- case 2: DO_STEREO(2); break;
- case 3: DO_STEREO(3); break;
- case 4: DO_STEREO(4); break;
- case 5: DO_STEREO(5); break;
- }
- }
- }
- }
-
} // namespace gnash
Index: server/swf/tag_loaders.cpp
===================================================================
RCS file: /sources/gnash/gnash/server/swf/tag_loaders.cpp,v
retrieving revision 1.97
retrieving revision 1.98
diff -u -b -r1.97 -r1.98
--- server/swf/tag_loaders.cpp 12 May 2007 06:50:37 -0000 1.97
+++ server/swf/tag_loaders.cpp 13 May 2007 12:20:36 -0000 1.98
@@ -17,7 +17,7 @@
// Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
//
-/* $Id: tag_loaders.cpp,v 1.97 2007/05/12 06:50:37 strk Exp $ */
+/* $Id: tag_loaders.cpp,v 1.98 2007/05/13 12:20:36 martinwguy Exp $ */
#ifdef HAVE_CONFIG_H
#include "config.h"
@@ -69,6 +69,17 @@
}
namespace gnash {
+
+// Forward declaration for functions at end of file
+
+static void u8_expand(void* out_data_void, stream* in,
+ int sample_count, // in stereo, this is number of *pairs* of samples
+ bool stereo);
+
+static void adpcm_expand(void* out_data_void, stream* in,
+ int sample_count, // in stereo, this is number of *pairs* of samples
+ bool stereo);
+
namespace SWF {
namespace tag_loaders {
@@ -1149,6 +1160,9 @@
// Sound
//
+// Common data
+static int s_sample_rate_table[] = { 5512, 11025, 22050, 44100 };
+
// @@ There are two sets of code to decode/expand/byteswap audio here.
// @@ There should be one (search for ADPCM).
@@ -1168,8 +1182,6 @@
bool stereo = in->read_uint(1) ? true : false;
int sample_count = in->read_u32();
- static int s_sample_rate_table[] = { 5512, 11025, 22050, 44100 };
-
IF_VERBOSE_PARSE
(
log_parse(_("define sound: ch=%d, format=%d, "
@@ -1182,7 +1194,7 @@
if (handler)
{
- int data_bytes = 0;
+ unsigned data_bytes = 0;
unsigned char* data = NULL;
if (! (sample_rate >= 0 && sample_rate <= 3))
@@ -1193,57 +1205,99 @@
return;
}
- if (format == sound_handler::FORMAT_ADPCM)
- {
+ switch (format) {
+
+ case sound_handler::FORMAT_ADPCM:
// Uncompress the ADPCM before handing data to host.
data_bytes = sample_count * (stereo ? 4 : 2);
data = new unsigned char[data_bytes];
- sound_handler::adpcm_expand(data, in, sample_count, stereo);
+ adpcm_expand(data, in, sample_count, stereo);
format = sound_handler::FORMAT_NATIVE16;
- }
- else
- {
- // @@ This is pretty awful -- lots of copying, slow reading.
+ break;
+
+ case sound_handler::FORMAT_RAW:
+ // 8- or 16-bit mono or stereo host-endian audio
data_bytes = in->get_tag_end_position() - in->get_position();
data = new unsigned char[data_bytes];
- for (int i = 0; i < data_bytes; i++)
- {
- data[i] = in->read_u8();
+ // Convert to 16-bit host-endian
+ if (sample_16bit) {
+ // FORMAT_RAW 16-bit is exactly what we want!
+ in->read((char *)data, data_bytes);
+ } else {
+ // Convert 8-bit signed to 16-bit range
+ // Allocate as many shorts as there are samples
+ int16_t *newdata = new int16_t[data_bytes];
+ u8_expand(newdata, in, sample_count, stereo);
+
+ // Keep new data and dispose of old
+ unsigned char *tmp = data;
+ data = (unsigned char *) newdata;
+ delete [] tmp;
}
+ format = sound_handler::FORMAT_NATIVE16;
+ break;
- // Samples are always little-endian.
- // Swap bytes on behalf of the host, to make it easier for the
handler.
- // @@ I'm assuming this is a good idea? Most sound handlers
will prefer native endianness?
- if (format == sound_handler::FORMAT_UNCOMPRESSED
- && sample_16bit)
+ case sound_handler::FORMAT_UNCOMPRESSED:
+ // 8- or 16-bit mono or stereo little-endian audio
+ data_bytes = in->get_tag_end_position() - in->get_position();
+
+ // Convert to 16-bit host-endian.
+ if (!sample_16bit)
{
+ // Convert 8-bit signed to 16-bit range
+ // Allocate as many shorts as there are 8-bit samples
+ int16_t *newdata = new int16_t[data_bytes];
+ u8_expand(newdata, in, sample_count, stereo);
+ data = (unsigned char *) newdata;
+ } else {
+ // Read 16-bit data into buffer
+ data = new unsigned char[data_bytes];
+ in->read((char *)data, data_bytes);
+
+ // Convert 16-bit little-endian data to host-endian.
+
// Runtime detection of host endianness costs almost
// nothing and is less of a continual maintenance headache
- // than attempting compile-time detection.
+ // than compile-time detection.
union u {
uint16_t s;
struct {
uint8_t c0;
uint8_t c1;
} c;
- } u;
- u.s = 0x0001;
+ } u = { 0x0001 };
+
switch (u.c.c0) {
case 0x01: // Little-endian host: sample is already native.
break;
case 0x00: // Big-endian host
// Swap sample bytes to get big-endian format.
- for (int i = 0; i < data_bytes - 1; i += 2)
+ assert(data_bytes & 1 == 0);
+ for (unsigned i = 0; i < data_bytes; i+=2)
{
swap(&data[i], &data[i+1]);
}
break;
default: // Impossible
- log_error(_("Host endianness not detected in
define-sound_loader"));
- assert(0);
+ log_error(_("Host endianness not detected in
define_sound_loader"));
+ // Just carry on anyway...
}
- format = sound_handler::FORMAT_NATIVE16;
}
+ format = sound_handler::FORMAT_NATIVE16;
+ break;
+
+ case sound_handler::FORMAT_MP3:
+ // Handled elsewhere
+ break;
+
+ case sound_handler::FORMAT_NELLYMOSER:
+ // One day...
+ break;
+
+ // This is impossible as an input but stops fussy compilers
+ // complaining about unhandled enum values.
+ case sound_handler::FORMAT_NATIVE16:
+ break;
}
int handler_id = handler->create_sound(
@@ -1338,8 +1392,6 @@
int sample_count = in->read_u32();
if (format == 2) garbage = in->read_uint(16);
- static int s_sample_rate_table[] = { 5512, 11025, 22050, 44100 };
-
IF_VERBOSE_PARSE
(
log_parse(_("sound stream head: format=%d, rate=%d, 16=%d, stereo=%d,
ct=%d"),
@@ -1411,7 +1463,7 @@
sample_count = data_bytes / (stereo ? 4 : 2);
data_bytes = sample_count * (stereo ? 4 : 2);
data = new unsigned char[data_bytes];
- sound_handler::adpcm_expand(data, in, sample_count, stereo);
+ adpcm_expand(data, in, sample_count, stereo);
format = sound_handler::FORMAT_NATIVE16;
} else if (format == sound_handler::FORMAT_NATIVE16)
{
@@ -1566,13 +1618,260 @@
}
-
-
} // namespace gnash::SWF::tag_loaders
} // namespace gnash::SWF
-} // namespace gnash
// Local Variables:
// mode: C++
// indent-tabs-mode: t
// End:
+
+// Expand ADPCM, 8-bit and non-host-endian 16-bit audio to 16-bit host-endian
+//
+// Provides:
+//
+// void adpcm_expand(void* out_data_void, stream* in,
+// int sample_count, // in stereo, this is number of *pairs* of samples
+// bool stereo)
+//
+// Uncompress ADPCM data from in stream to out_data[].
+// The output buffer must have (sample_count*2) bytes for mono,
+// or (sample_count*4) bytes for stereo.
+//
+// This code has wandered round the Gnash source tree and has still not found
+// its proper home yet.
+
+
+//
+// Unsigned 8-bit expansion (128 is silence)
+//
+
+static void u8_expand(
+ void* out_data_void,
+ stream* in,
+ int sample_count, // in stereo, this is number of *pairs* of
samples
+ bool stereo)
+{
+ int16_t *out_data = (int16_t*) out_data_void;
+
+ unsigned total_samples = stereo ? sample_count*2 : sample_count;
+ uint8_t *in_data = new uint8_t[total_samples];
+ in->read((char *)in_data, total_samples); // Read 8-bit samples
+
+ // Convert 8-bit to 16
+ uint8_t *inp = in_data;
+ int16_t *outp = out_data;
+ for (unsigned i=total_samples; i>0; i--) {
+ *outp++ = ((int16_t)(*inp++) - 128) * 256;
+ }
+
+ delete [] in_data;
+}
+
+//
+// ADPCM
+//
+
+// Data from Alexis' SWF reference
+static int s_index_update_table_2bits[2] = { -1, 2 };
+static int s_index_update_table_3bits[4] = { -1, -1, 2, 4 };
+static int s_index_update_table_4bits[8] = { -1, -1, -1, -1, 2, 4, 6,
8 };
+static int s_index_update_table_5bits[16] = { -1, -1, -1, -1, -1, -1, -1,
-1, 1, 2, 4, 6, 8, 10, 13, 16 };
+
+static int* s_index_update_tables[4] = {
+ s_index_update_table_2bits,
+ s_index_update_table_3bits,
+ s_index_update_table_4bits,
+ s_index_update_table_5bits,
+};
+
+// Data from Jansen. http://homepages.cwi.nl/~jack/
+// Check out his Dutch retro punk songs, heh heh :)
+const int STEPSIZE_CT = 89;
+static int s_stepsize[STEPSIZE_CT] = {
+ 7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
+ 19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
+ 50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
+ 130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
+ 337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
+ 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
+ 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
+ 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
+ 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
+};
+
+
+// Algo from
http://www.circuitcellar.com/pastissues/articles/richey110/text.htm
+// And also Jansen.
+// Here's another reference:
http://www.geocities.com/SiliconValley/8682/aud3.txt
+// Original IMA spec doesn't seem to be on the web :(
+
+
+// @@ lots of macros here! It seems that VC6 can't correctly
+// handle integer template args, although it's happy to
+// compile them?!
+
+// void DO_SAMPLE(int n_bits, int& sample, int& stepsize_index, int
raw_code)
+#define DO_SAMPLE(n_bits, sample, stepsize_index, raw_code)
\
+{
\
+ assert(raw_code >= 0 && raw_code < (1 << n_bits));
\
+
\
+ static const int HI_BIT = (1 << (n_bits - 1));
\
+ int* index_update_table = s_index_update_tables[n_bits - 2];
\
+
\
+ /* Core of ADPCM. */
\
+
\
+ int code_mag = raw_code & (HI_BIT - 1);
\
+ bool code_sign_bit = (raw_code & HI_BIT) ? 1 : 0;
\
+ int mag = (code_mag << 1) + 1; /* shift in LSB (they do this
so that pos & neg zero are different)*/ \
+
\
+ int stepsize = s_stepsize[stepsize_index];
\
+
\
+ /* Compute the new sample. It's the predicted value
*/ \
+ /* (i.e. the previous value), plus a delta. The delta
*/ \
+ /* comes from the code times the stepsize. going for
*/ \
+ /* something like: delta = stepsize * (code * 2 + 1) >> code_bits
*/ \
+ int delta = (stepsize * mag) >> (n_bits - 1);
\
+ if (code_sign_bit) delta = -delta;
\
+
\
+ sample += delta;
\
+ sample = iclamp(sample, -32768, 32767);
\
+
\
+ /* Update our stepsize index. Use a lookup table. */
\
+ stepsize_index += index_update_table[code_mag];
\
+ stepsize_index = iclamp(stepsize_index, 0, STEPSIZE_CT - 1);
\
+}
+
+
+class in_stream
+{
+public:
+ const unsigned char* m_in_data;
+ int m_current_bits;
+ int m_unused_bits;
+
+ in_stream(const unsigned char* data)
+ :
+ m_in_data(data),
+ m_current_bits(0),
+ m_unused_bits(0)
+ {
+ }
+};
+
+
+// void DO_MONO_BLOCK(int16_t** out_data, int n_bits, int sample_count,
stream* in, int sample, int stepsize_index)
+#define DO_MONO_BLOCK(out_data, n_bits, sample_count, in, sample,
stepsize_index) \
+{
\
+ /* First sample doesn't need to be decompressed. */
\
+ sample_count--;
\
+ *(*out_data)++ = (int16_t) sample;
\
+
\
+ while (sample_count--)
\
+ {
\
+ int raw_code = in->read_uint(n_bits);
\
+ DO_SAMPLE(n_bits, sample, stepsize_index, raw_code); /*
sample & stepsize_index are in/out params */ \
+ *(*out_data)++ = (int16_t) sample;
\
+ }
\
+}
+
+
+// void do_stereo_block(
+// int16_t** out_data, // in/out param
+// int n_bits,
+// int sample_count,
+// stream* in,
+// int left_sample,
+// int left_stepsize_index,
+// int right_sample,
+// int right_stepsize_index
+// )
+#define DO_STEREO_BLOCK(out_data, n_bits, sample_count, in, left_sample,
left_stepsize_index, right_sample, right_stepsize_index) \
+/* Uncompress 4096 stereo sample pairs of ADPCM. */
\
+{
\
+ /* First samples don't need to be decompressed. */
\
+ sample_count--;
\
+ *(*out_data)++ = (int16_t) left_sample;
\
+ *(*out_data)++ = (int16_t) right_sample;
\
+
\
+ while (sample_count--)
\
+ {
\
+ int left_raw_code = in->read_uint(n_bits);
\
+ DO_SAMPLE(n_bits, left_sample, left_stepsize_index,
left_raw_code); \
+ *(*out_data)++ = (int16_t) left_sample;
\
+
\
+ int right_raw_code = in->read_uint(n_bits);
\
+ DO_SAMPLE(n_bits, right_sample, right_stepsize_index,
right_raw_code); \
+ *(*out_data)++ = (int16_t) right_sample;
\
+ }
\
+}
+
+
+// Utility function: uncompress ADPCM data from in stream to
+// out_data[]. The output buffer must have (sample_count*2)
+// bytes for mono, or (sample_count*4) bytes for stereo.
+static void adpcm_expand(
+ void* out_data_void,
+ stream* in,
+ int sample_count, // in stereo, this is number of *pairs* of
samples
+ bool stereo)
+{
+ int16_t* out_data = (int16_t*) out_data_void;
+
+ // Read header.
+ int n_bits = in->read_uint(2) + 2; // 2 to 5 bits
+
+ while (sample_count)
+ {
+ // Read initial sample & index values.
+ int sample = in->read_sint(16);
+
+ int stepsize_index = in->read_uint(6);
+ assert(STEPSIZE_CT >= (1 << 6)); // ensure we don't need
to clamp.
+
+ int samples_this_block = imin(sample_count, 4096);
+ sample_count -= samples_this_block;
+
+ if (stereo == false)
+ {
+#define DO_MONO(n) DO_MONO_BLOCK(&out_data, n, samples_this_block, in, sample,
stepsize_index)
+
+ switch (n_bits)
+ {
+ default: assert(0); break;
+ case 2: DO_MONO(2); break;
+ case 3: DO_MONO(3); break;
+ case 4: DO_MONO(4); break;
+ case 5: DO_MONO(5); break;
+ }
+ }
+ else
+ {
+ // Stereo.
+
+ // Got values for left channel; now get initial sample
+ // & index for right channel.
+ int right_sample = in->read_sint(16);
+
+ int right_stepsize_index = in->read_uint(6);
+ assert(STEPSIZE_CT >= (1 << 6)); // ensure we
don't need to clamp.
+
+#define DO_STEREO(n) \
+ DO_STEREO_BLOCK( \
+ &out_data, n, samples_this_block, \
+ in, sample, stepsize_index, \
+ right_sample, right_stepsize_index)
+
+ switch (n_bits)
+ {
+ default: assert(0); break;
+ case 2: DO_STEREO(2); break;
+ case 3: DO_STEREO(3); break;
+ case 4: DO_STEREO(4); break;
+ case 5: DO_STEREO(5); break;
+ }
+ }
+ }
+}
+
+} // namespace gnash
Index: ChangeLog
===================================================================
RCS file: /sources/gnash/gnash/ChangeLog,v
retrieving revision 1.3175
retrieving revision 1.3176
diff -u -b -r1.3175 -r1.3176
--- ChangeLog 13 May 2007 12:19:05 -0000 1.3175
+++ ChangeLog 13 May 2007 12:20:36 -0000 1.3176
@@ -2,6 +2,9 @@
* server/stream.{h.cpp}: Punch a block-read primitive through the
stream layer.
+ * server/parser/sound_definition.cpp: Move ADPCM code elsewhere
+ * server/swf/tag_loaders.cpp: Decode all linear formats into native 16,
+ move ADPCM code here where it's used.
2007-05-12 Tomas Groth Christensen <address@hidden>
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