admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Move H264BitstreamParser to video_coding.

Browse Source 
Moves parser, used in video_coding/ from rtp_rtcp where it is unused.

BUG=webrtc:5678
R=asapersson@webrtc.org
TBR=glaznev@webrt.org

Review URL: https://codereview.webrtc.org/2007553003 .

Cr-Commit-Position: refs/heads/master@{#12866}
This commit is contained in:
Peter Boström 2016-05-24 12:16:26 +02:00
parent 3ede7be00a
commit cc1543abf3
9 changed files with 11 additions and 11 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/api/java/jni/androidmediaencoder_jni.cc
View File

@ -29,8 +29,8 @@
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/modules/video_coding/utility/h264_bitstream_parser.h"
#include "webrtc/modules/video_coding/utility/quality_scaler.h"
#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
#include "webrtc/system_wrappers/include/field_trial.h"

2
webrtc/modules/modules.gyp
View File

@ -297,7 +297,6 @@
'rtp_rtcp/source/fec_test_helper.cc',
'rtp_rtcp/source/fec_test_helper.h',
'rtp_rtcp/source/h264_sps_parser_unittest.cc',
'rtp_rtcp/source/h264_bitstream_parser_unittest.cc',
'rtp_rtcp/source/nack_rtx_unittest.cc',
'rtp_rtcp/source/packet_loss_stats_unittest.cc',
'rtp_rtcp/source/producer_fec_unittest.cc',
@ -382,6 +381,7 @@
'video_coding/test/stream_generator.cc',
'video_coding/test/stream_generator.h',
'video_coding/utility/frame_dropper_unittest.cc',
'video_coding/utility/h264_bitstream_parser_unittest.cc',
'video_coding/utility/ivf_file_writer_unittest.cc',
'video_coding/utility/quality_scaler_unittest.cc',
'video_processing/test/denoiser_test.cc',

2
webrtc/modules/rtp_rtcp/BUILD.gn
View File

@ -32,8 +32,6 @@ source_set("rtp_rtcp") {
"source/forward_error_correction.h",
"source/forward_error_correction_internal.cc",
"source/forward_error_correction_internal.h",
"source/h264_bitstream_parser.cc",
"source/h264_bitstream_parser.h",
"source/h264_sps_parser.cc",
"source/h264_sps_parser.h",
"source/mock/mock_rtp_payload_strategy.h",

2
webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
View File

@ -135,8 +135,6 @@
'source/forward_error_correction.h',
'source/forward_error_correction_internal.cc',
'source/forward_error_correction_internal.h',
'source/h264_bitstream_parser.cc',
'source/h264_bitstream_parser.h',
'source/h264_sps_parser.cc',
'source/h264_sps_parser.h',
'source/producer_fec.cc',

2
webrtc/modules/video_coding/BUILD.gn
View File

@ -104,6 +104,8 @@ source_set("video_coding_utility") {
sources = [
"utility/frame_dropper.cc",
"utility/frame_dropper.h",
"utility/h264_bitstream_parser.cc",
"utility/h264_bitstream_parser.h",
"utility/ivf_file_writer.cc",
"utility/ivf_file_writer.h",
"utility/moving_average.h",

2
webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc → webrtc/modules/video_coding/utility/h264_bitstream_parser.cc
View File

@ -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 "webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h"
#include "webrtc/modules/video_coding/utility/h264_bitstream_parser.h"
#include <memory>
#include <vector>

6
webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h → webrtc/modules/video_coding/utility/h264_bitstream_parser.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_H264_BITSTREAM_PARSER_H_
#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_H264_BITSTREAM_PARSER_H_
#include <stddef.h>
#include <stdint.h>
@ -77,4 +77,4 @@ class H264BitstreamParser {
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
#endif // WEBRTC_MODULES_VIDEO_CODING_UTILITY_H264_BITSTREAM_PARSER_H_

2
webrtc/modules/rtp_rtcp/source/h264_bitstream_parser_unittest.cc → webrtc/modules/video_coding/utility/h264_bitstream_parser_unittest.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h"
#include "webrtc/modules/video_coding/utility/h264_bitstream_parser.h"
#include "testing/gtest/include/gtest/gtest.h"

2
webrtc/modules/video_coding/utility/video_coding_utility.gyp
View File

@ -20,6 +20,8 @@
'sources': [
'frame_dropper.cc',
'frame_dropper.h',
'h264_bitstream_parser.cc',
'h264_bitstream_parser.h',
'ivf_file_writer.cc',
'ivf_file_writer.h',
'moving_average.h',