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Side by Side Diff: webrtc/modules/audio_processing/test/aec_dump_based_simulator.h

Issue 1907223003: Extension and refactoring of the audioproc_f tool to be a fully fledged tool for audio processing m… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Changes in response to reviewer comments Created 4 years, 7 months ago
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1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
13
14 #include "webrtc/modules/audio_processing/test/audio_processing_simulator.h"
15
16 #include "webrtc/base/constructormagic.h"
17
18 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
19 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
20 #else
21 #include "webrtc/modules/audio_processing/debug.pb.h"
22 #endif
23
24 namespace webrtc {
25 namespace test {
26
27 // Used to perform an audio processing simulation from an aec dump.
28 class AecDumpBasedSimulator final : public AudioProcessingSimulator {
29 public:
30 explicit AecDumpBasedSimulator(const SimulationSettings& settings)
31 : AudioProcessingSimulator(settings) {}
32 virtual ~AecDumpBasedSimulator() {}
33
34 // Processes the messages in the aecdump file.
35 void Process() override;
36
37 private:
38 void HandleMessage(const webrtc::audioproc::Init& msg);
39 void HandleMessage(const webrtc::audioproc::Stream& msg);
40 void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
41 void HandleMessage(const webrtc::audioproc::Config& msg);
42 void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg);
43 void PrepareReverseProcessStreamCall(
44 const webrtc::audioproc::ReverseStream& msg);
45 void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);
46
47 enum InterfaceType {
48 kFixedInterface,
49 kFloatInterface,
50 kNotSpecified,
51 };
52
53 FILE* dump_input_file_;
54 InterfaceType interface_used_ = InterfaceType::kNotSpecified;
55
56 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator);
57 };
58
59 } // namespace test
60 } // namespace webrtc
61
62 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
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