Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(601)

Side by Side Diff: webrtc/modules/audio_coding/acm2/acm_receiver.h

Issue 3007763002: Move array_view.h to webrtc/api/ (Closed)
Patch Set: rebase Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/modules/audio_coding/BUILD.gn ('k') | webrtc/modules/audio_coding/acm2/rent_a_codec.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
13 13
14 #include <map> 14 #include <map>
15 #include <memory> 15 #include <memory>
16 #include <string> 16 #include <string>
17 #include <vector> 17 #include <vector>
18 18
19 #include "webrtc/api/array_view.h"
19 #include "webrtc/common_audio/vad/include/webrtc_vad.h" 20 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
20 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" 21 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
21 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" 22 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
22 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" 23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
23 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" 24 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
24 #include "webrtc/modules/include/module_common_types.h" 25 #include "webrtc/modules/include/module_common_types.h"
25 #include "webrtc/rtc_base/array_view.h"
26 #include "webrtc/rtc_base/criticalsection.h" 26 #include "webrtc/rtc_base/criticalsection.h"
27 #include "webrtc/rtc_base/optional.h" 27 #include "webrtc/rtc_base/optional.h"
28 #include "webrtc/rtc_base/thread_annotations.h" 28 #include "webrtc/rtc_base/thread_annotations.h"
29 #include "webrtc/typedefs.h" 29 #include "webrtc/typedefs.h"
30 30
31 namespace webrtc { 31 namespace webrtc {
32 32
33 struct CodecInst; 33 struct CodecInst;
34 class NetEq; 34 class NetEq;
35 35
(...skipping 249 matching lines...) Expand 10 before | Expand all | Expand 10 after
285 const Clock* const clock_; 285 const Clock* const clock_;
286 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); 286 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
287 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); 287 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
288 }; 288 };
289 289
290 } // namespace acm2 290 } // namespace acm2
291 291
292 } // namespace webrtc 292 } // namespace webrtc
293 293
294 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ 294 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/BUILD.gn ('k') | webrtc/modules/audio_coding/acm2/rent_a_codec.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698