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

Issue 2078313002: Remove header files for the AEC and the APM test program that are no longer used. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 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_PROCESSOR_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_PROCESSOR_H_
13
14 #include <algorithm>
15 #include <limits>
16 #include <memory>
17 #include <string>
18 #include <vector>
19
20 #include "webrtc/base/timeutils.h"
21 #include "webrtc/base/optional.h"
22 #include "webrtc/common_audio/channel_buffer.h"
23 #include "webrtc/common_audio/wav_file.h"
24 #include "webrtc/modules/audio_processing/include/audio_processing.h"
25 #include "webrtc/modules/audio_processing/test/audio_file_processor.h"
26 #include "webrtc/modules/audio_processing/test/test_utils.h"
27
28 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
29 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
30 #else
31 #include "webrtc/modules/audio_processing/debug.pb.h"
32 #endif
33
34 namespace webrtc {
35 namespace test {
36
37 // Used to read from an aecdump file and write to a WavWriter.
38 class AecDumpFileProcessor final : public AudioFileProcessor {
39 public:
40 AecDumpFileProcessor(std::unique_ptr<AudioProcessing> ap,
41 FILE* dump_file,
42 std::string out_filename,
43 std::string reverse_out_filename,
44 rtc::Optional<int> out_sample_rate_hz,
45 rtc::Optional<int> out_num_channels,
46 rtc::Optional<int> reverse_out_sample_rate_hz,
47 rtc::Optional<int> reverse_out_num_channels,
48 bool override_config_message);
49
50 virtual ~AecDumpFileProcessor();
51
52 // Processes the messages in the aecdump file and returns
53 // the number of forward stream chunks processed.
54 size_t Process(bool verbose_logging) override;
55
56 private:
57 void HandleMessage(const webrtc::audioproc::Init& msg);
58 void HandleMessage(const webrtc::audioproc::Stream& msg);
59 void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
60 void HandleMessage(const webrtc::audioproc::Config& msg);
61
62 enum InterfaceType {
63 kIntInterface,
64 kFloatInterface,
65 kNotSpecified,
66 };
67
68 std::unique_ptr<AudioProcessing> ap_;
69 FILE* dump_file_;
70 std::string out_filename_;
71 std::string reverse_out_filename_;
72 rtc::Optional<int> out_sample_rate_hz_;
73 rtc::Optional<int> out_num_channels_;
74 rtc::Optional<int> reverse_out_sample_rate_hz_;
75 rtc::Optional<int> reverse_out_num_channels_;
76 bool override_config_message_;
77
78 std::unique_ptr<ChannelBuffer<float>> in_buf_;
79 std::unique_ptr<ChannelBuffer<float>> reverse_buf_;
80 std::unique_ptr<ChannelBuffer<float>> out_buf_;
81 std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_;
82 std::unique_ptr<WavWriter> out_file_;
83 std::unique_ptr<WavWriter> reverse_out_file_;
84 StreamConfig input_config_;
85 StreamConfig reverse_config_;
86 StreamConfig output_config_;
87 StreamConfig reverse_output_config_;
88 std::unique_ptr<ChannelBufferWavWriter> buffer_writer_;
89 std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
90 AudioFrame far_frame_;
91 AudioFrame near_frame_;
92 InterfaceType interface_used_ = InterfaceType::kNotSpecified;
93 };
94
95 } // namespace test
96 } // namespace webrtc
97
98 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_PROCESSOR_H_
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