File: BUILD.gn

package info (click to toggle)
chromium 138.0.7204.183-1
  • links: PTS, VCS
  • area: main
  • in suites: trixie
  • size: 6,071,908 kB
  • sloc: cpp: 34,937,088; ansic: 7,176,967; javascript: 4,110,704; python: 1,419,953; asm: 946,768; xml: 739,971; pascal: 187,324; sh: 89,623; perl: 88,663; objc: 79,944; sql: 50,304; cs: 41,786; fortran: 24,137; makefile: 21,806; php: 13,980; tcl: 13,166; yacc: 8,925; ruby: 7,485; awk: 3,720; lisp: 3,096; lex: 1,327; ada: 727; jsp: 228; sed: 36
file content (42 lines) | stat: -rw-r--r-- 1,096 bytes parent folder | download | duplicates (20)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS.  All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.

import("../../webrtc.gni")

rtc_library("async_audio_processing") {
  sources = [
    "async_audio_processing.cc",
    "async_audio_processing.h",
  ]

  public = [ "async_audio_processing.h" ]

  deps = [
    "../../api:scoped_refptr",
    "../../api:sequence_checker",
    "../../api/audio:audio_frame_api",
    "../../api/audio:audio_frame_processor",
    "../../api/task_queue:task_queue",
    "../../rtc_base:checks",
    "../../rtc_base:refcount",
  ]
}

if (rtc_include_tests) {
  rtc_library("async_audio_processing_test") {
    testonly = true

    sources = []

    deps = [
      ":async_audio_processing",
      "../../api/audio:audio_frame_api",
      "../../rtc_base:checks",
    ]
  }
}