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Revert 3170 - Added performance benchmarking in APM and iSAC-fix for Buildbots.

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Review URL: https://webrtc-codereview.appspot.com/929022

TBR=kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/969009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3172 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2012-11-26 23:57:38 +00:00
parent 4cd8f1f182
commit b43502e388
4 changed files with 14 additions and 26 deletions
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20
webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc → webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.c
View File

@ -8,17 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
/* kenny.c - Main function for the iSAC coder */
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <time.h>
#include <ctype.h>
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "isacfix.h"
// TODO(kma): Clean up the code and change benchmarking the whole codec to
// separate encoder and decoder.
/* Defines */
#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
@ -171,7 +170,7 @@ int main(int argc, char* argv[])
" in adaptive mode.\n\n");
printf("[-FL num] :Set (initial) frame length in msec. Valid length"
" are 30 and 60 msec.\n\n");
printf("[-FIXED_FL] :Frame length to be fixed to initial value.\n\n");
printf("[-FIXED_FL] :Frame length will be fixed to initial value.\n\n");
printf("[-MAX num] :Set the limit for the payload size of iSAC"
" in bytes. \n");
printf(" Minimum 100, maximum 400.\n\n");
@ -375,8 +374,7 @@ int main(int argc, char* argv[])
sscanf(argv[CodingMode+1], "%s", bottleneck_file);
f_bn = fopen(bottleneck_file, "rb");
if (f_bn == NULL) {
printf("No value provided for BottleNeck and cannot read file %s\n",
bottleneck_file);
printf("No value provided for BottleNeck and cannot read file %s\n", bottleneck_file);
exit(0);
} else {
int aux_var;
@ -567,8 +565,8 @@ int main(int argc, char* argv[])
shortdata,
(WebRtc_Word16*)streamdata);
/* If packet is ready, and CE testing, call the different API
functions from the internal API. */
/* If packet is ready, and CE testing, call the different API functions
from the internal API. */
if (stream_len>0) {
if (testCE == 1) {
err = WebRtcIsacfix_ReadBwIndex((WebRtc_Word16*)streamdata, &bwe);
@ -810,10 +808,6 @@ int main(int argc, char* argv[])
runtime, (100*runtime/length_file));
printf("\n\n_______________________________________________\n");
// Record the results with Perf test tools.
webrtc::test::PrintResult("time_per_10ms_frame", "", "isac",
(runtime * 10000) / length_file, "us", false);
fclose(inp);
fclose(outp);
fclose(outbits);

3
webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi
View File

@ -14,14 +14,13 @@
'type': 'executable',
'dependencies': [
'iSACFix',
'<(webrtc_root)/test/test.gyp:test_support',
],
'include_dirs': [
'./fix/test',
'./fix/interface',
],
'sources': [
'./fix/test/kenny.cc',
'./fix/test/kenny.c',
],
},
{

3
webrtc/modules/audio_processing/audio_processing_tests.gypi
View File

@ -58,9 +58,8 @@
'dependencies': [
'audio_processing',
'audioproc_debug_proto',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support',
'<(DEPTH)/testing/gtest.gyp:gtest',
],
'sources': [ 'test/process_test.cc', ],
},

14
webrtc/modules/audio_processing/test/process_test.cc
View File

@ -19,12 +19,11 @@
#include "gtest/gtest.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "audio_processing.h"
#include "cpu_features_wrapper.h"
#include "module_common_types.h"
#include "scoped_ptr.h"
#include "tick_util.h"
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
#else
@ -1033,9 +1032,6 @@ void void_main(int argc, char* argv[]) {
(exec_time * 1.0) / primary_count,
(max_time_us + max_time_reverse_us) / 1000.0,
(min_time_us + min_time_reverse_us) / 1000.0);
// Record the results with Perf test tools.
webrtc::test::PrintResult("time_per_10ms_frame", "", "audioproc",
(exec_time * 1000) / primary_count, "us", false);
} else {
printf("Warning: no capture frames\n");
}