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Fixing WebRTC after moving from src/webrtc to src/

Browse Source 
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:47:31 +02:00
committed by Commit Bot
parent bb547203bf
commit 92ea95e34a
3635 changed files with 19692 additions and 19645 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
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@ -13,8 +13,8 @@
#include <map>
#include <stdio.h>
#include "webrtc/typedefs.h"
#include "webrtc/modules/include/module_common_types.h"
#include "typedefs.h"
#include "modules/include/module_common_types.h"
enum stereoModes {
stereoModeMono,

4
modules/audio_coding/neteq/test/RTPchange.cc
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@ -13,8 +13,8 @@
#include <algorithm>
#include <vector>
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h"
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#include "modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h"
#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#define FIRSTLINELEN 40
//#define WEBRTC_DUMMY_RTP

24
modules/audio_coding/neteq/test/RTPencode.cc
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@ -25,12 +25,12 @@
#include <algorithm>
#include "webrtc/rtc_base/checks.h"
#include "webrtc/typedefs.h"
#include "rtc_base/checks.h"
#include "typedefs.h"
// needed for NetEqDecoder
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/audio_decoder_impl.h"
#include "modules/audio_coding/neteq/include/neteq.h"
/************************/
/* Define payload types */
@ -132,10 +132,10 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
#include "webrtc_vad.h"
#if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC))
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#endif
#ifdef CODEC_G711
#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
#include "modules/audio_coding/codecs/g711/g711_interface.h"
#endif
#ifdef CODEC_G729
#include "G729Interface.h"
@ -152,19 +152,19 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
#include "AMRWBCreation.h"
#endif
#ifdef CODEC_ILBC
#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
#include "modules/audio_coding/codecs/ilbc/ilbc.h"
#endif
#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
#include "modules/audio_coding/codecs/isac/main/include/isac.h"
#endif
#ifdef NETEQ_ISACFIX_CODEC
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#ifdef CODEC_ISAC
#error Cannot have both ISAC and ISACfix defined. Please de-select one.
#endif
#endif
#ifdef CODEC_G722
#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
#include "modules/audio_coding/codecs/g722/g722_interface.h"
#endif
#ifdef CODEC_G722_1_24
#include "G722_1Interface.h"
@ -194,10 +194,10 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
#endif
#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
#endif
#ifdef CODEC_OPUS
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#endif
/***********************************/

4
modules/audio_coding/neteq/test/RTPjitter.cc
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@ -25,8 +25,8 @@
#include <assert.h>
#include "webrtc/test/gtest.h"
#include "webrtc/typedefs.h"
#include "test/gtest.h"
#include "typedefs.h"
/*********************/
/* Misc. definitions */

4
modules/audio_coding/neteq/test/RTPtimeshift.cc
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@ -12,8 +12,8 @@
#include <algorithm>
#include <vector>
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#include "webrtc/test/gtest.h"
#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#include "test/gtest.h"
#define FIRSTLINELEN 40

12
modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
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@ -10,12 +10,12 @@
#include <memory>
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "rtc_base/safe_conversions.h"
#include "test/testsupport/fileutils.h"
using testing::InitGoogleTest;

6
modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
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@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/rtc_base/flags.h"
#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/flags.h"
using testing::InitGoogleTest;

8
modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/rtc_base/flags.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/codecs/opus/opus_inst.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/flags.h"
using testing::InitGoogleTest;

12
modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
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@ -10,12 +10,12 @@
#include <memory>
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "rtc_base/safe_conversions.h"
#include "test/testsupport/fileutils.h"
using testing::InitGoogleTest;

10
modules/audio_coding/neteq/test/neteq_performance_unittest.cc
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/typedefs.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "test/gtest.h"
#include "test/testsupport/perf_test.h"
#include "typedefs.h"
#include "system_wrappers/include/field_trial.h"
// Runs a test with 10% packet losses and 10% clock drift, to exercise
// both loss concealment and time-stretching code.

8
modules/audio_coding/neteq/test/neteq_speed_test.cc
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@ -12,10 +12,10 @@
#include <iostream>
#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "rtc_base/flags.h"
#include "test/testsupport/fileutils.h"
#include "typedefs.h"
// Define command line flags.
DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms.");