audioconvert inefficient when converting only endianness / buffer passthrough method
Endejan, Edward
Edward.Endejan at dmh-global.com
Thu Feb 2 18:57:58 PST 2012
Hi again,
I'll try to restate and simplify my questions.
1. Is it preferable to optimize the existing audioconvert
plugin for a special case where the only operation
to be performed is endianness conversion or
to create a new plugin for this particular purpose?
The 'if' clause to see if the special case applies is
fairly quick and lightweight and the overall
modification does not add much to the code size.
2. Is there a recommended method to passthrough a
buffer with zero copy when no modification of the
buffer contents are required, but allow modification
in place when the data needs to be modified by
the filter?
More details are listed in my original post below for those
who might be interested.
Thanks,
Ed
________________________________________
From: gstreamer-devel-bounces+edward.endejan=dmh-global.com at lists.freedesktop.org [gstreamer-devel-bounces+edward.endejan=dmh-global.com at lists.freedesktop.org] On Behalf Of Endejan, Edward [Edward.Endejan at dmh-global.com]
Sent: Wednesday, February 01, 2012 10:58 AM
To: gstreamer-devel at lists.freedesktop.org
Subject: audioconvert inefficient when converting only endianness
Hi,
I have two main questions listed near the end, but first here is the
background information ...
I have been evaluating the CPU load of various GStreamer pipelines.
In one case of interest, I needed to use audioconvert simply to
change the endianness of a PCM audio stream.
Server pipeline:
alsasrc ! audioconvert ! \
"audio/x-raw-int, endianness=(int)4321, \
signed=(boolean)true, width=(int)16, depth=(int)16, \
rate=(int)44100, channels=(int)2" ! \
rtpL16pay ! udpsink
Client pipeline:
udpsrc ! rtpL16depay ! queue2 ! audioconvert !
"audio/x-raw-int, endianness=(int)1234, \
signed=(boolean)true, width=(int)16, depth=(int)16, \
rate=(int)44100, channels=(int)2" ! \
alsasink
I found that the audioconvert plugin was consuming more than 50%
of the processor budget for the entire chain. When I investigated
this I noticed that audioconvert was always up-converting the
input stream to at least 32-bit signed, presumably to retain
precision for the potential conversions that it would do. But in
the case that I need, where endianness conversion is the only
change required, this up-conversion is unnecessary and wasteful
in terms of CPU load and potentially memory consumption as well.
I implemented a modification (see the end of my post for a diff
showing my current, rough proposed changes) which first checks
if endianness conversion is the only required change, and if so
avoids the type conversion step. The performance improvement is
dramatic for the 16-bit width case that I've tested. I also
verified proper operation for 8, 32, and 64-bit width cases, but
have not compared performance for these cases.
I tried to test the 24-bit width case, which supposedly is a
valid setting for width, but I always get the error
"size 4096 is not a multiple of unit size 6" as the pipeline is
being set up and so have not been able to verify that case.
I also tried to use "width=24, depth=32" but it did not help.
(For all the other tests I used settings with depth == width.)
If someone could explain how to test the 24-bit width case,
please let me know and I'll give it a try.
Though the 8-bit case works as implemented, I think that it
could be further optimized if the input buffer could simply be
passed along as the output buffer rather than being copied.
I attempted to use gst_buffer_replace(), but that causes the
gst-launch process to hang. I have seen the
gst_base_transform_is_passthrough flag in the GstBaseTransform
object which looks promising, but I'm not sure how or if I could
use it for this case. I would appreciate a recommendation or a
reference to example code on the proper method for passing along
a buffer without change from input to output.
Notice that I had to add a special clause to detect the 8-bit width
case since the endianness comparison alone was not catching it.
Endianness really doesn't matter for the 8-bit width case, since
we are only talking about byte endianness, not bit endianness.
So, here are my main questions:
1. Is it better to optimize endianness conversion
within the audioconvert plugin or
to create a new plugin for this explicit purpose?
2. What is the proper way to pass a buffer from input
to output without doing a copy?
Best Regards,
Ed Endejan
P.S. Below are the changes I've tested thus far:
---------------
--- gst-plugins-base-0.10.35/gst/audioconvert/audioconvert.c.org
+++ gst-plugins-base-0.10.35/gst/audioconvert/audioconvert.c
@@ -709,6 +709,81 @@
return TRUE;
}
+static inline void
+audio_convert_endianness_only (gpointer src, gpointer dst,
+ gint width, gint samples)
+{
+ /* assumes src & dst already checked for NULL and samples > 0 */
+ /* which is a valid assumption since this is only called from */
+ /* within audio_convert_convert after such checks */
+ switch(width) {
+ case 64: {
+ uint64_t *sp = (uint64_t *)src;
+ uint64_t *dp = (uint64_t *)dst;
+ for(; samples; samples--) {
+ *dp = (*sp << 56)
+ | ((*sp & 0x000000000000FF00) << 40)
+ | ((*sp & 0x0000000000FF0000) << 24)
+ | ((*sp & 0x00000000FF000000) << 8)
+ | ((*sp & 0x000000FF00000000) >> 8)
+ | ((*sp & 0x0000FF0000000000) >> 24)
+ | ((*sp & 0x00FF000000000000) >> 40)
+ | (*sp >> 56);
+ sp++;
+ dp++;
+ }
+ break;
+ }
+ case 32: {
+ uint32_t *sp = (uint32_t *)src;
+ uint32_t *dp = (uint32_t *)dst;
+ for(; samples; samples--) {
+ *dp = (*sp << 24)
+ | ((*sp & 0x0000FF00) << 8)
+ | ((*sp & 0x00FF0000) >> 8)
+ | (*sp >> 24);
+ sp++;
+ dp++;
+ }
+ break;
+ }
+#if 0
+ case 24: {
+ /* 24-bit case is not real */
+ /* attempting to use width=24 causes error "size 4096 is not a multiple of unit size 6" */
+ GST_WARNING ("24-bit endianness conversion ignored\n");
+ (void)memcpy((void *)dst, (void *)src, (size_t)(width/8 * samples));
+ /* gst_buffer_replace(&dst, src); */ /* just pass along the unmodified buffer */
+ break;
+ }
+#endif
+ case 16: {
+ uint16_t *sp = (uint16_t *)src;
+ uint16_t *dp = (uint16_t *)dst;
+ for(; samples; samples--) {
+ *dp = (*sp << 8) | (*sp >> 8);
+ sp++;
+ dp++;
+ }
+ break;
+ }
+ case 8: {
+ /* endianness conversion is a nop for 8-bit width */
+ (void)memcpy((void *)dst, (void *)src, (size_t)samples);
+ /* gst_buffer_replace(&dst, src); */ /* just pass along the unmodified buffer */
+ break;
+ }
+#if 0
+ default: {
+ GST_WARNING ("Unexpected width - endianness conversion ignored\n");
+ (void)memcpy((void *)dst, (void *)src, (size_t)(width/8 * samples));
+ /* gst_buffer_replace(&dst, src); */ /* just pass along the unmodified buffer */
+ break;
+ }
+#endif
+ }
+}
+
gboolean
audio_convert_convert (AudioConvertCtx * ctx, gpointer src,
gpointer dst, gint samples, gboolean src_writable)
@@ -725,6 +800,14 @@
if (samples == 0)
return TRUE;
+ if (ctx->mix_passthrough && (ctx->in.width == ctx->out.width)
+ && ((ctx->in.width == 8) || (ctx->in.endianness != ctx->out.endianness))) {
+ /* no need for type conversion if only endianness conversion is needed */
+ audio_convert_endianness_only (src, dst, ctx->out.width, samples * ctx->out.channels);
+
+ return TRUE;
+ }
+
insize = ctx->in.unit_size * samples;
outsize = ctx->out.unit_size * samples;
---------------
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