generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Change gflags and gmock includes to be full paths.

Browse Source 
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kjellander@webrtc.org 2014-09-04 09:55:40 +00:00
parent 51bb33cc18
commit 3c0aae17f0
73 changed files with 211 additions and 197 deletions
DEPS
webrtc
base
gunit.h
modules
audio_coding
codecs
cng
cng_unittest.cc
isac/fix/source
filterbanks_unittest.ccfilters_unittest.cclpc_masking_model_unittest.cctransform_unittest.cc
main
acm2
acm_opus_unittest.ccacm_receiver_unittest.cccall_statistics_unittest.ccinitial_delay_manager_unittest.ccnack_unittest.cc
test
PCMFile.ccRTPFile.ccTestAllCodecs.ccTestStereo.ccTwoWayCommunication.ccdual_stream_unittest.ccinitial_delay_unittest.ccinsert_packet_with_timing.cctarget_delay_unittest.cc
neteq
audio_classifier_unittest.ccaudio_decoder_unittest.ccaudio_multi_vector_unittest.ccaudio_vector_unittest.ccbackground_noise_unittest.ccbuffer_level_filter_unittest.cccomfort_noise_unittest.ccdecision_logic_unittest.ccdecoder_database_unittest.ccdelay_manager_unittest.ccdelay_peak_detector_unittest.ccdsp_helper_unittest.ccdtmf_buffer_unittest.ccdtmf_tone_generator_unittest.ccexpand_unittest.ccmerge_unittest.cc
mock
mock_audio_decoder.hmock_audio_vector.hmock_buffer_level_filter.hmock_decoder_database.hmock_delay_manager.hmock_delay_peak_detector.hmock_dtmf_buffer.hmock_dtmf_tone_generator.hmock_external_decoder_pcm16b.hmock_packet_buffer.hmock_payload_splitter.h
neteq_external_decoder_unittest.ccneteq_impl_unittest.ccneteq_stereo_unittest.ccneteq_unittest.ccnormal_unittest.ccpacket_buffer_unittest.ccpayload_splitter_unittest.ccpost_decode_vad_unittest.ccrandom_vector_unittest.ccsync_buffer_unittest.cc
test
RTPcat.ccRTPjitter.ccRTPtimeshift.cc
time_stretch_unittest.cctimestamp_scaler_unittest.cc
tools
input_audio_file_unittest.ccpacket_unittest.cc
desktop_capture
desktop_and_cursor_composer_unittest.ccdesktop_region_unittest.ccmouse_cursor_monitor_unittest.ccwindow_capturer_unittest.cc
remote_bitrate_estimator
bwe_simulations.cc
test
bwe_test.hbwe_test_framework_unittest.cc
tools/e2e_quality/audio
audio_e2e_harness.cc

1
DEPS

@ -49,6 +49,7 @@ include_rules = [
# checkdeps.py shouldn't check include paths for files in these dirs:
skip_child_includes = [
'gflags',
'testing',
'third_party',
]

2
webrtc/base/gunit.h

@ -14,7 +14,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
#if defined(WEBRTC_ANDROID) || defined(GTEST_RELATIVE_PATH)
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#else
#include "testing/base/public/gunit.h"
#endif

2
webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc

@ -9,7 +9,7 @@
*/
#include <string>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc_cng.h"

2
webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h"

2
webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc

@ -7,7 +7,7 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc

@ -7,7 +7,7 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"

2
webrtc/modules/audio_coding/main/acm2/acm_opus_unittest.cc

@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/acm2/acm_opus.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc

@ -12,7 +12,7 @@
#include <algorithm> // std::min
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"

2
webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc

@ -10,7 +10,7 @@
#include <string.h>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/acm2/nack_unittest.cc

@ -14,7 +14,7 @@
#include <algorithm>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/typedefs.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"

4
webrtc/modules/audio_coding/main/test/PCMFile.cc

@ -14,8 +14,8 @@
#include <stdio.h>
#include <string.h>
#include "gtest/gtest.h"
#include "module_common_types.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/interface/module_common_types.h"
namespace webrtc {

3
webrtc/modules/audio_coding/main/test/RTPFile.cc

@ -20,8 +20,9 @@
#include "audio_coding_module.h"
#include "engine_configurations.h"
#include "gtest/gtest.h" // TODO (tlegrand): Consider removing usage of gtest.
#include "rw_lock_wrapper.h"
// TODO(tlegrand): Consider removing usage of gtest.
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/test/TestAllCodecs.cc

@ -13,7 +13,7 @@
#include <stdio.h>
#include <string>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"

2
webrtc/modules/audio_coding/main/test/TestStereo.cc

@ -14,7 +14,7 @@
#include <string>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"

2
webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc

@ -18,7 +18,7 @@
#include <Windows.h>
#endif
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"

2
webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"

2
webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc

@ -15,7 +15,7 @@
#include <iostream>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"

2
webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc

@ -11,7 +11,7 @@
#include <stdio.h>
#include "gflags/gflags.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"

2
webrtc/modules/audio_coding/main/test/target_delay_unittest.cc

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"

2
webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc

@ -16,7 +16,7 @@
#include <string.h>
#include <string>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

@ -15,7 +15,7 @@
#include <string>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_audio/resampler/include/resampler.h"
#ifdef WEBRTC_CODEC_CELT
#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"

2
webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc

@ -15,7 +15,7 @@
#include <string>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc

@ -15,7 +15,7 @@
#include <string>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/background_noise_unittest.cc

@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc

@ -14,7 +14,7 @@
#include <math.h> // Access to pow function.
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc

@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/comfort_noise.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"

2
webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc

@ -10,7 +10,7 @@
// Unit tests for DecisionLogic class and derived classes.
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/decision_logic.h"

4
webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc

@ -15,8 +15,8 @@
#include <string>
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
#include "webrtc/test/testsupport/gtest_disable.h"

4
webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc

@ -14,8 +14,8 @@
#include <math.h>
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc

@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc

@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc

@ -18,7 +18,7 @@
#include <iostream>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
// Modify the tests so that they pass with the modifications done to DtmfBuffer
// for backwards bit-exactness. Once bit-exactness is no longer required, this

2
webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc

@ -14,7 +14,7 @@
#include <math.h>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/expand_unittest.cc

@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"

2
webrtc/modules/audio_coding/neteq/merge_unittest.cc

@ -14,7 +14,7 @@
#include <vector>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/random_vector.h"

2
webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_audio_vector.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/audio_vector.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/payload_splitter.h"
#include "gmock/gmock.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {

4
webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc

@ -13,8 +13,8 @@
#include <string>
#include <list>
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"

4
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

@ -11,8 +11,8 @@
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/accelerate.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"

2
webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc

@ -14,7 +14,7 @@
#include <string>
#include <list>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"

2
webrtc/modules/audio_coding/neteq/neteq_unittest.cc

@ -24,7 +24,7 @@
#include <vector>
#include "gflags/gflags.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"

2
webrtc/modules/audio_coding/neteq/normal_unittest.cc

@ -14,7 +14,7 @@
#include <vector>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"

4
webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc

@ -12,8 +12,8 @@
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"

2
webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc

@ -16,7 +16,7 @@
#include <utility> // pair
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"

2
webrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc

@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/random_vector_unittest.cc

@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc

@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/test/RTPcat.cc

@ -13,7 +13,7 @@
#include <algorithm>
#include <vector>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#define FIRSTLINELEN 40

250
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

@ -25,7 +25,7 @@
#include <assert.h>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/typedefs.h"
/*********************/
@ -33,11 +33,13 @@
/*********************/
#define FIRSTLINELEN 40
#define CHECK_NOT_NULL(a) if((a)==NULL){fprintf(stderr,"\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
#define CHECK_NOT_NULL(a) if((a)==NULL){ \
fprintf(stderr,"\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a ); \
return(-1);}
struct arr_time {
float time;
uint32_t ix;
float time;
uint32_t ix;
};
int filelen(FILE *fid)
@ -61,157 +63,167 @@ int compare_arr_time(const void *x, const void *y);
int main(int argc, char* argv[])
{
unsigned int dat_len, rtp_len, Npack, k;
arr_time *time_vec;
char firstline[FIRSTLINELEN];
unsigned char* rtp_vec = NULL;
unsigned int dat_len, rtp_len, Npack, k;
arr_time *time_vec;
char firstline[FIRSTLINELEN];
unsigned char* rtp_vec = NULL;
unsigned char** packet_ptr = NULL;
unsigned char* temp_packet = NULL;
const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
uint16_t len;
uint32_t *offset;
const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
uint16_t len;
uint32_t *offset;
/* check number of parameters */
if (argc != 4) {
/* print help text and exit */
printf("Apply jitter on RTP stream.\n");
printf("The program reads an RTP stream and packet timing from two files.\n");
printf("The RTP stream is modified to have the same jitter as described in the timing files.\n");
printf("The format of the RTP stream file should be the same as for rtpplay,\n");
printf("and can be obtained e.g., from Ethereal by using\n");
printf("Statistics -> RTP -> Show All Streams -> [select a stream] -> Save As\n\n");
printf("Usage:\n\n");
printf("%s RTP_infile dat_file RTP_outfile\n", argv[0]);
printf("where:\n");
if (argc != 4) {
/* print help text and exit */
printf("Apply jitter on RTP stream.\n");
printf("Reads an RTP stream and packet timing from two files.\n");
printf("The RTP stream is modified to have the same jitter as described in "
"the timing files.\n");
printf("The format of the RTP stream file should be the same as for \n");
printf("rtpplay, and can be obtained e.g., from Ethereal by using\n");
printf("Statistics -> RTP -> Show All Streams -> [select a stream] -> "
"Save As\n\n");
printf("Usage:\n\n");
printf("%s RTP_infile dat_file RTP_outfile\n", argv[0]);
printf("where:\n");
printf("RTP_infile : RTP stream input file\n\n");
printf("RTP_infile : RTP stream input file\n\n");
printf("dat_file : file with packet arrival times in ms\n\n");
printf("dat_file : file with packet arrival times in ms\n\n");
printf("RTP_outfile : RTP stream output file\n\n");
printf("RTP_outfile : RTP stream output file\n\n");
return(0);
}
return(0);
}
FILE* in_file=fopen(argv[1],"rb");
CHECK_NOT_NULL(in_file);
printf("Input file: %s\n",argv[1]);
FILE* dat_file=fopen(argv[2],"rb");
CHECK_NOT_NULL(dat_file);
printf("Dat-file: %s\n",argv[2]);
FILE* out_file=fopen(argv[3],"wb");
CHECK_NOT_NULL(out_file);
printf("Output file: %s\n\n",argv[3]);
time_vec = (arr_time *) malloc(sizeof(arr_time)*(filelen(dat_file)/sizeof(float)) + 1000); // add 1000 bytes to avoid (rare) strange error
if (time_vec==NULL) {
fprintf(stderr, "Error: could not allocate memory for reading dat file\n");
goto closing;
}
FILE* in_file=fopen(argv[1],"rb");
CHECK_NOT_NULL(in_file);
printf("Input file: %s\n",argv[1]);
FILE* dat_file=fopen(argv[2],"rb");
CHECK_NOT_NULL(dat_file);
printf("Dat-file: %s\n",argv[2]);
FILE* out_file=fopen(argv[3],"wb");
CHECK_NOT_NULL(out_file);
printf("Output file: %s\n\n",argv[3]);
// add 1000 bytes to avoid (rare) strange error.
time_vec = (arr_time *) malloc(sizeof(arr_time)
*(filelen(dat_file)/sizeof(float)) + 1000);
if (time_vec==NULL) {
fprintf(stderr, "Error: could not allocate memory for reading dat file\n");
goto closing;
}
dat_len=0;
while(fread(&(time_vec[dat_len].time),sizeof(float),1,dat_file)>0) {
time_vec[dat_len].ix=dat_len;
dat_len++;
}
dat_len=0;
while(fread(&(time_vec[dat_len].time),sizeof(float),1,dat_file)>0) {
time_vec[dat_len].ix=dat_len;
dat_len++;
}
if (dat_len == 0) {
fprintf(stderr, "Error: dat_file is empty, no arrival time is given.\n");
goto closing;
}
qsort(time_vec,dat_len,sizeof(arr_time),compare_arr_time);
qsort(time_vec,dat_len,sizeof(arr_time),compare_arr_time);
rtp_vec = (unsigned char *) malloc(sizeof(unsigned char)*filelen(in_file));
if (rtp_vec==NULL) {
fprintf(stderr,"Error: could not allocate memory for reading rtp file\n");
goto closing;
}
rtp_vec = (unsigned char *) malloc(sizeof(unsigned char)*filelen(in_file));
if (rtp_vec==NULL) {
fprintf(stderr,"Error: could not allocate memory for reading rtp file\n");
goto closing;
}
// read file header and write directly to output file
EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, in_file) != NULL);
EXPECT_GT(fputs(firstline, out_file), 0);
EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
in_file));
EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
out_file));
// read file header and write directly to output file
EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, in_file) != NULL);
EXPECT_GT(fputs(firstline, out_file), 0);
EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
in_file));
EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
out_file));
// read all RTP packets into vector
rtp_len=0;
Npack=0;
len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of first packet
while(len==2) {
len = ntohs(*((uint16_t *)(rtp_vec + rtp_len)));
rtp_len += 2;
if(fread(&rtp_vec[rtp_len], sizeof(unsigned char), len-2, in_file)!=(unsigned) (len-2)) {
fprintf(stderr,"Error: currupt packet length\n");
goto closing;
}
rtp_len += len-2;
Npack++;
len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of next packet
}
// read all RTP packets into vector
rtp_len=0;
Npack=0;
if (Npack == 0) {
fprintf(stderr, "Error: No RTP packet found.\n");
goto closing;
}
// read length of first packet.
len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file);
while(len==2) {
len = ntohs(*((uint16_t *)(rtp_vec + rtp_len)));
rtp_len += 2;
if(fread(&rtp_vec[rtp_len], sizeof(unsigned char),
len-2, in_file)!=(unsigned) (len-2)) {
fprintf(stderr,"Error: currupt packet length\n");
goto closing;
}
rtp_len += len-2;
Npack++;
packet_ptr = (unsigned char **) malloc(Npack*sizeof(unsigned char*));
// read length of next packet.
len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file);
}
packet_ptr[0]=rtp_vec;
k=1;
while(k<Npack) {
len = ntohs(*((uint16_t *) packet_ptr[k-1]));
packet_ptr[k]=packet_ptr[k-1]+len;
k++;
}
if (Npack == 0) {
fprintf(stderr, "Error: No RTP packet found.\n");
goto closing;
}
for(k=0; k<dat_len && k<Npack; k++) {
if(time_vec[k].time < FLT_MAX && time_vec[k].ix < Npack){
temp_packet = packet_ptr[time_vec[k].ix];
offset = (uint32_t *) (temp_packet+4);
if ( time_vec[k].time >= 0 ) {
*offset = htonl((uint32_t) time_vec[k].time);
}
else {
*offset = htonl((uint32_t) 0);
fprintf(stderr, "Warning: negative receive time in dat file transformed to 0.\n");
}
packet_ptr = (unsigned char **) malloc(Npack*sizeof(unsigned char*));
// write packet to file
if (fwrite(temp_packet, sizeof(unsigned char),
ntohs(*((uint16_t*) temp_packet)),
out_file) !=
ntohs(*((uint16_t*) temp_packet))) {
return -1;
}
}
}
packet_ptr[0]=rtp_vec;
k=1;
while(k<Npack) {
len = ntohs(*((uint16_t *) packet_ptr[k-1]));
packet_ptr[k]=packet_ptr[k-1]+len;
k++;
}
for(k=0; k<dat_len && k<Npack; k++) {
if(time_vec[k].time < FLT_MAX && time_vec[k].ix < Npack){
temp_packet = packet_ptr[time_vec[k].ix];
offset = (uint32_t *) (temp_packet+4);
if ( time_vec[k].time >= 0 ) {
*offset = htonl((uint32_t) time_vec[k].time);
}
else {
*offset = htonl((uint32_t) 0);
fprintf(stderr, "Warning: negative receive time in dat file transformed"
" to 0.\n");
}
// write packet to file
if (fwrite(temp_packet, sizeof(unsigned char),
ntohs(*((uint16_t*) temp_packet)),
out_file) !=
ntohs(*((uint16_t*) temp_packet))) {
return -1;
}
}
}
closing:
free(time_vec);
free(rtp_vec);
free(time_vec);
free(rtp_vec);
if (packet_ptr != NULL) {
free(packet_ptr);
free(packet_ptr);
}
fclose(in_file);
fclose(dat_file);
fclose(out_file);
fclose(dat_file);
fclose(out_file);
return(0);
return(0);
}
int compare_arr_time(const void *xp, const void *yp) {
if(((arr_time *)xp)->time == ((arr_time *)yp)->time)
return(0);
else if(((arr_time *)xp)->time > ((arr_time *)yp)->time)
return(1);
if(((arr_time *)xp)->time == ((arr_time *)yp)->time)
return(0);
else if(((arr_time *)xp)->time > ((arr_time *)yp)->time)
return(1);
return(-1);
return(-1);
}

2
webrtc/modules/audio_coding/neteq/test/RTPtimeshift.cc

@ -13,7 +13,7 @@
#include <vector>
#include "NETEQTEST_RTPpacket.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
/*********************/
/* Misc. definitions */

2
webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_coding/neteq/accelerate.h"
#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
namespace webrtc {

4
webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc

@ -10,8 +10,8 @@
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"

2
webrtc/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc

@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {
namespace test {

2
webrtc/modules/audio_coding/neteq/tools/packet_unittest.cc

@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {
namespace test {

2
webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc

@ -10,7 +10,7 @@
#include "webrtc/modules/desktop_capture/desktop_and_cursor_composer.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/desktop_capture/desktop_capture_options.h"
#include "webrtc/modules/desktop_capture/desktop_frame.h"
#include "webrtc/modules/desktop_capture/mouse_cursor.h"

2
webrtc/modules/desktop_capture/desktop_region_unittest.cc

@ -12,7 +12,7 @@
#include <algorithm>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace webrtc {

2
webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc

@ -10,7 +10,7 @@
#include "webrtc/modules/desktop_capture/mouse_cursor_monitor.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/desktop_capture/desktop_capture_options.h"
#include "webrtc/modules/desktop_capture/desktop_frame.h"
#include "webrtc/modules/desktop_capture/mouse_cursor.h"

2
webrtc/modules/desktop_capture/window_capturer_unittest.cc

@ -10,7 +10,7 @@
#include "webrtc/modules/desktop_capture/window_capturer.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/desktop_capture/desktop_capture_options.h"
#include "webrtc/modules/desktop_capture/desktop_frame.h"
#include "webrtc/modules/desktop_capture/desktop_region.h"

2
webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
#include "webrtc/test/testsupport/fileutils.h"

2
webrtc/modules/remote_bitrate_estimator/test/bwe_test.h

@ -11,7 +11,7 @@
#include <map>
#include <string>
#include <vector>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"

2
webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc

@ -12,7 +12,7 @@
#include <numeric>
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/test/testsupport/fileutils.h"

2
webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc

@ -13,7 +13,7 @@
// flags.
#include "gflags/gflags.h"
#include "gtest/gtest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/channel_transport/include/channel_transport.h"