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Side by Side Diff: webrtc/voice_engine/mock/fake_voe_external_media.h

Issue 1224163002: Update audio code to use size_t more correctly, webrtc/voice_engine/ portion. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Checkpoint Created 5 years, 5 months ago
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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
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35 callback_map_.erase(type); 35 callback_map_.erase(type);
36 return 0; 36 return 0;
37 } 37 }
38 WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz, 38 WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz,
39 AudioFrame* frame)); 39 AudioFrame* frame));
40 WEBRTC_STUB(SetExternalMixing, (int channel, bool enable)); 40 WEBRTC_STUB(SetExternalMixing, (int channel, bool enable));
41 41
42 // Use this to trigger the Process() callback to a registered media processor. 42 // Use this to trigger the Process() callback to a registered media processor.
43 // If |audio| is NULL, a zero array of the correct length will be forwarded. 43 // If |audio| is NULL, a zero array of the correct length will be forwarded.
44 void CallProcess(ProcessingTypes type, int16_t* audio, 44 void CallProcess(ProcessingTypes type, int16_t* audio,
45 int samples_per_channel, int sample_rate_hz, 45 size_t samples_per_channel, int sample_rate_hz,
46 int num_channels) { 46 int num_channels) {
pbos-webrtc 2015/07/14 08:15:10 num_channels size_t as well?
47 const int length = samples_per_channel * num_channels; 47 const size_t length = samples_per_channel * num_channels;
48 rtc::scoped_ptr<int16_t[]> data; 48 rtc::scoped_ptr<int16_t[]> data;
49 if (!audio) { 49 if (!audio) {
50 data.reset(new int16_t[length]); 50 data.reset(new int16_t[length]);
51 memset(data.get(), 0, length * sizeof(data[0])); 51 memset(data.get(), 0, length * sizeof(data[0]));
52 audio = data.get(); 52 audio = data.get();
53 } 53 }
54 54
55 std::map<ProcessingTypes, VoEMediaProcess*>::const_iterator it = 55 std::map<ProcessingTypes, VoEMediaProcess*>::const_iterator it =
56 callback_map_.find(type); 56 callback_map_.find(type);
57 if (it != callback_map_.end()) { 57 if (it != callback_map_.end()) {
58 it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz, 58 it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
59 num_channels == 2 ? true : false); 59 num_channels == 2 ? true : false);
60 } 60 }
61 } 61 }
62 62
63 private: 63 private:
64 std::map<ProcessingTypes, VoEMediaProcess*> callback_map_; 64 std::map<ProcessingTypes, VoEMediaProcess*> callback_map_;
65 }; 65 };
66 66
67 } // namespace webrtc 67 } // namespace webrtc
68 68
69 #endif // WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_ 69 #endif // WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
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