OLD | NEW |
---|---|
1 /* | 1 /* |
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2017 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 #include <utility> | |
12 | |
11 #include "webrtc/modules/audio_processing/aec_dump/aec_dump.h" | 13 #include "webrtc/modules/audio_processing/aec_dump/aec_dump.h" |
12 | 14 |
13 namespace webrtc { | 15 namespace webrtc { |
14 | 16 |
15 InternalAPMConfig::InternalAPMConfig() = default; | 17 InternalAPMConfig::InternalAPMConfig() = default; |
16 InternalAPMConfig::InternalAPMConfig(const InternalAPMConfig&) = default; | 18 InternalAPMConfig::InternalAPMConfig(const InternalAPMConfig&) = default; |
17 InternalAPMConfig::InternalAPMConfig(InternalAPMConfig&&) = default; | 19 InternalAPMConfig::InternalAPMConfig(InternalAPMConfig&&) = default; |
18 | 20 |
19 AecDump::AecDump(int64_t max_log_size_bytes, rtc::TaskQueue* worker_queue) {} | 21 AecDump::AecDump(int64_t max_log_size_bytes, rtc::TaskQueue* worker_queue) {} |
20 | 22 |
21 AecDump::AecDump(rtc::PlatformFile file, | 23 AecDump::AecDump(rtc::PlatformFile file, |
22 int64_t max_log_size_bytes, | 24 int64_t max_log_size_bytes, |
23 rtc::TaskQueue* worker_queue) {} | 25 rtc::TaskQueue* worker_queue) {} |
24 | 26 |
25 AecDump::AecDump(std::string file_name, | 27 AecDump::AecDump(std::string file_name, |
26 int64_t max_log_size_bytes, | 28 int64_t max_log_size_bytes, |
27 rtc::TaskQueue* worker_queue) {} | 29 rtc::TaskQueue* worker_queue) {} |
28 | 30 |
29 AecDump::AecDump(FILE* handle, | 31 AecDump::AecDump(FILE* handle, |
30 int64_t max_log_size_bytes, | 32 int64_t max_log_size_bytes, |
31 rtc::TaskQueue* worker_queue) {} | 33 rtc::TaskQueue* worker_queue) {} |
32 | 34 |
33 AecDump::~AecDump() {} | 35 AecDump::~AecDump() {} |
34 | 36 |
35 AecDump::CaptureStreamInfo::CaptureStreamInfo() = default; | 37 AecDump::CaptureStreamInfo::CaptureStreamInfo() = default; |
36 AecDump::CaptureStreamInfo::~CaptureStreamInfo() = default; | 38 AecDump::CaptureStreamInfo::~CaptureStreamInfo() = default; |
37 | 39 |
38 void AecDump::CaptureStreamInfo::AddInput(FloatAudioFrame src) {} | 40 void AecDump::CaptureStreamInfo::AddInput(FloatAudioFrame src) {} |
peah-webrtc
2017/05/09 07:14:57
Would it make sense to put RTC_NOTREACHED() here?
| |
39 void AecDump::CaptureStreamInfo::AddOutput(FloatAudioFrame src) {} | 41 void AecDump::CaptureStreamInfo::AddOutput(FloatAudioFrame src) {} |
40 | 42 |
41 void AecDump::CaptureStreamInfo::AddInput(const AudioFrame& frame) {} | 43 void AecDump::CaptureStreamInfo::AddInput(const AudioFrame& frame) {} |
42 void AecDump::CaptureStreamInfo::AddOutput(const AudioFrame& frame) {} | 44 void AecDump::CaptureStreamInfo::AddOutput(const AudioFrame& frame) {} |
43 | 45 |
44 void AecDump::CaptureStreamInfo::set_delay(int delay) {} | 46 void AecDump::CaptureStreamInfo::set_delay(int delay) {} |
45 void AecDump::CaptureStreamInfo::set_drift(int drift) {} | 47 void AecDump::CaptureStreamInfo::set_drift(int drift) {} |
46 void AecDump::CaptureStreamInfo::set_level(int level) {} | 48 void AecDump::CaptureStreamInfo::set_level(int level) {} |
47 void AecDump::CaptureStreamInfo::set_keypress(bool keypress) {} | 49 void AecDump::CaptureStreamInfo::set_keypress(bool keypress) {} |
48 | 50 |
49 std::unique_ptr<audioproc::Event> GetEventMsg(); | 51 std::unique_ptr<audioproc::Event> GetEventMsg(); |
50 | 52 |
51 void AecDump::WriteInitMessage(const InternalAPMStreamsConfig& streams_config) { | 53 void AecDump::WriteInitMessage(const InternalAPMStreamsConfig& streams_config) { |
52 } | 54 } |
53 | 55 |
54 void AecDump::WriteRenderStreamMessage(const AudioFrame& frame) {} | 56 void AecDump::WriteRenderStreamMessage(const AudioFrame& frame) {} |
55 | 57 |
56 void AecDump::WriteRenderStreamMessage(FloatAudioFrame src) {} | 58 void AecDump::WriteRenderStreamMessage(FloatAudioFrame src) {} |
57 | 59 |
58 void AecDump::WriteCaptureStreamMessage( | 60 void AecDump::WriteCaptureStreamMessage( |
59 CaptureStreamInfo* capture_stream_info) {} | 61 CaptureStreamInfo* capture_stream_info) {} |
60 | 62 |
61 void AecDump::WriteConfig(const InternalAPMConfig& config, bool forced) {} | 63 void AecDump::WriteConfig(const InternalAPMConfig& config, bool forced) {} |
62 | 64 |
63 void AecDump::PostTask(std::unique_ptr<audioproc::Event> event) {} | 65 void AecDump::PostTask(std::unique_ptr<audioproc::Event> event) {} |
64 } // namespace webrtc | 66 } // namespace webrtc |
OLD | NEW |