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Side by Side Diff: talk/media/webrtc/fakewebrtcvoiceengine.h

Issue 1316523002: Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Fix compile Created 4 years, 11 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2010 Google Inc. 3 * Copyright 2010 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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70 WEBRTC_STUB(Initialize, ( 70 WEBRTC_STUB(Initialize, (
71 const webrtc::ProcessingConfig& processing_config)); 71 const webrtc::ProcessingConfig& processing_config));
72 72
73 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { 73 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
74 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; 74 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
75 } 75 }
76 76
77 WEBRTC_STUB_CONST(input_sample_rate_hz, ()); 77 WEBRTC_STUB_CONST(input_sample_rate_hz, ());
78 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); 78 WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
79 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); 79 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
80 WEBRTC_STUB_CONST(num_input_channels, ()); 80 size_t num_input_channels() const override { return 0; }
81 WEBRTC_STUB_CONST(num_proc_channels, ()); 81 size_t num_proc_channels() const override { return 0; }
82 WEBRTC_STUB_CONST(num_output_channels, ()); 82 size_t num_output_channels() const override { return 0; }
83 WEBRTC_STUB_CONST(num_reverse_channels, ()); 83 size_t num_reverse_channels() const override { return 0; }
84 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); 84 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
85 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); 85 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
86 WEBRTC_STUB(ProcessStream, ( 86 WEBRTC_STUB(ProcessStream, (
87 const float* const* src, 87 const float* const* src,
88 size_t samples_per_channel, 88 size_t samples_per_channel,
89 int input_sample_rate_hz, 89 int input_sample_rate_hz,
90 webrtc::AudioProcessing::ChannelLayout input_layout, 90 webrtc::AudioProcessing::ChannelLayout input_layout,
91 int output_sample_rate_hz, 91 int output_sample_rate_hz,
92 webrtc::AudioProcessing::ChannelLayout output_layout, 92 webrtc::AudioProcessing::ChannelLayout output_layout,
93 float* const* dest)); 93 float* const* dest));
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815 int playout_fail_channel_; 815 int playout_fail_channel_;
816 int send_fail_channel_; 816 int send_fail_channel_;
817 int recording_sample_rate_; 817 int recording_sample_rate_;
818 int playout_sample_rate_; 818 int playout_sample_rate_;
819 FakeAudioProcessing audio_processing_; 819 FakeAudioProcessing audio_processing_;
820 }; 820 };
821 821
822 } // namespace cricket 822 } // namespace cricket
823 823
824 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ 824 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
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