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Issue 2060403002: Add task queue to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@move_getpadding
Patch Set: Fix audio thread check when adding audio to bitrateallocator. Created 4 years, 4 months ago
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1 # Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
2 # 2 #
3 # Use of this source code is governed by a BSD-style license 3 # Use of this source code is governed by a BSD-style license
4 # that can be found in the LICENSE file in the root of the source 4 # that can be found in the LICENSE file in the root of the source
5 # tree. An additional intellectual property rights grant can be found 5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may 6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree. 7 # be found in the AUTHORS file in the root of the source tree.
8 { 8 {
9 'targets': [ 9 'targets': [
10 { 10 {
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369 'video/call_stats_unittest.cc', 369 'video/call_stats_unittest.cc',
370 'video/encoder_state_feedback_unittest.cc', 370 'video/encoder_state_feedback_unittest.cc',
371 'video/end_to_end_tests.cc', 371 'video/end_to_end_tests.cc',
372 'video/overuse_frame_detector_unittest.cc', 372 'video/overuse_frame_detector_unittest.cc',
373 'video/payload_router_unittest.cc', 373 'video/payload_router_unittest.cc',
374 'video/report_block_stats_unittest.cc', 374 'video/report_block_stats_unittest.cc',
375 'video/send_delay_stats_unittest.cc', 375 'video/send_delay_stats_unittest.cc',
376 'video/send_statistics_proxy_unittest.cc', 376 'video/send_statistics_proxy_unittest.cc',
377 'video/stats_counter_unittest.cc', 377 'video/stats_counter_unittest.cc',
378 'video/stream_synchronization_unittest.cc', 378 'video/stream_synchronization_unittest.cc',
379 'video/video_capture_input_unittest.cc',
380 'video/video_decoder_unittest.cc', 379 'video/video_decoder_unittest.cc',
381 'video/video_encoder_unittest.cc', 380 'video/video_encoder_unittest.cc',
382 'video/video_send_stream_tests.cc', 381 'video/video_send_stream_tests.cc',
382 'video/vie_encoder_unittest.cc',
383 'video/vie_remb_unittest.cc', 383 'video/vie_remb_unittest.cc',
384 ], 384 ],
385 'dependencies': [ 385 'dependencies': [
386 '<(DEPTH)/testing/gmock.gyp:gmock', 386 '<(DEPTH)/testing/gmock.gyp:gmock',
387 '<(DEPTH)/testing/gtest.gyp:gtest', 387 '<(DEPTH)/testing/gtest.gyp:gtest',
388 '<(webrtc_root)/common.gyp:webrtc_common', 388 '<(webrtc_root)/common.gyp:webrtc_common',
389 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', 389 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
390 '<(webrtc_root)/modules/modules.gyp:video_capture', 390 '<(webrtc_root)/modules/modules.gyp:video_capture',
391 '<(webrtc_root)/test/test.gyp:channel_transport', 391 '<(webrtc_root)/test/test.gyp:channel_transport',
392 '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine', 392 '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
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660 'build/isolate.gypi', 660 'build/isolate.gypi',
661 ], 661 ],
662 'sources': [ 662 'sources': [
663 'webrtc_perf_tests.isolate', 663 'webrtc_perf_tests.isolate',
664 ], 664 ],
665 }, 665 },
666 ], 666 ],
667 }], 667 }],
668 ], 668 ],
669 } 669 }
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