Patch is attached.
I also noticed that the logic for checking the global
sound_play_utility variable in sound.m wasn't working for me.
The existing code uses the 'exist' function to see if the users has
already set sound_play_utility:
function sound()
## ...
if (exist("sound_play_utility", "var"))
## The user is always right, so just declare the set variable as global
global sound_play_utility;
else ## check path for known audio players...
exist always returns false for me inside the function, regardless of
whether I declare sound_play_utility global in my octaverc or
interpreter. After reading the docs on global variables, I think it
should be:
function sound()
## ...
global sound_play_utility;
if ~isempty(sound_play_utility),
## User has defined a play command
else ## check path for known audio players ...
--Roman
Index: sound.m
===================================================================
--- sound.m (revision 5480)
+++ sound.m (working copy)
@@ -13,11 +13,13 @@
## You should have received a copy of the GNU General Public License
## along with this program; If not, see <http://www.gnu.org/licenses/>.
-## usage: sound(x [, fs])
+## usage: sound(x [, fs, bs])
##
## Play the signal through the speakers. Data is a matrix with
## one column per channel. Rate fs defaults to 8000 Hz. The signal
-## is clipped to [-1, 1].
+## is clipped to [-1, 1]. Buffer size bs controls how many audio samples
+## are clipped and buffered before sending them to the audio player. bs
+## defaults to fs, which is equivalent to 1 second of audio.
##
## Note that if $DISPLAY != $HOSTNAME:n then a remote shell is opened
## to the host specified in $HOSTNAME to play the audio. See manual
@@ -55,12 +57,20 @@
## as your sound_play_utility. Things you may want to do are resample
## so that the rate is appropriate for your machine and convert the data
## to mulaw and output as bytes.
-function sound(data, rate)
+##
+## If you experience buffer underruns while playing audio data, the bs
+## buffer size parameter can be increased to tradeoff interactivity
+## for smoother playback. If bs=Inf, then all the data is clipped and
+## buffered before sending it to the audio player pipe. By default, 1
+## sec of audio is buffered.
- if nargin<1 || nargin>2
- usage("sound(x [, fs])");
+function sound(data, rate, buffer_size)
+
+ if nargin<1 || nargin>3
+ usage("sound(x [, fs, bs])");
endif
if nargin<2 || isempty(rate), rate = 8000; endif
+ if nargin<3 || isempty(buffer_size), buffer_size = rate; endif
if rows(data) != length(data), data=data'; endif
[samples, channels] = size(data);
@@ -114,7 +124,19 @@
fwrite(fid, 3, 'int32', 0, 'ieee-be');
fwrite(fid, rate, 'int32', 0, 'ieee-be');
fwrite(fid, channels, 'int32', 0, 'ieee-be');
- fwrite(fid, 32767*clip(data,[-1, 1])', 'int16', 0, 'ieee-be');
+
+ if isinf(buffer_size),
+ fwrite(fid, 32767*clip(data,[-1, 1])', 'int16', 0, 'ieee-be');
+ else
+ ## write data in blocks rather than all at once
+ nblocks = ceil(samples/buffer_size);
+ block_start = 1;
+ for i=1:nblocks,
+ block_end = min(size(data,1), block_start+buffer_size-1);
+ fwrite(fid, 32767*clip(data(block_start:block_end,:),[-1, 1])', 'int16', 0, 'ieee-be');
+ block_start = block_end + 1;
+ end
+ endif
pclose(fid);
endif
end
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