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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 "webrtc/common_audio/audio_converter.h" | 11 #include "webrtc/common_audio/audio_converter.h" |
12 | 12 |
13 #include <cstring> | 13 #include <cstring> |
14 | 14 |
15 #include "webrtc/base/checks.h" | 15 #include "webrtc/base/checks.h" |
16 #include "webrtc/base/safe_conversions.h" | 16 #include "webrtc/base/safe_conversions.h" |
17 #include "webrtc/common_audio/channel_buffer.h" | 17 #include "webrtc/common_audio/channel_buffer.h" |
18 #include "webrtc/common_audio/resampler/push_sinc_resampler.h" | 18 #include "webrtc/common_audio/resampler/push_sinc_resampler.h" |
19 #include "webrtc/system_wrappers/interface/scoped_vector.h" | 19 #include "webrtc/system_wrappers/include/scoped_vector.h" |
20 | 20 |
21 using rtc::checked_cast; | 21 using rtc::checked_cast; |
22 | 22 |
23 namespace webrtc { | 23 namespace webrtc { |
24 | 24 |
25 class CopyConverter : public AudioConverter { | 25 class CopyConverter : public AudioConverter { |
26 public: | 26 public: |
27 CopyConverter(int src_channels, size_t src_frames, int dst_channels, | 27 CopyConverter(int src_channels, size_t src_frames, int dst_channels, |
28 size_t dst_frames) | 28 size_t dst_frames) |
29 : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {} | 29 : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {} |
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191 RTC_CHECK(dst_channels == src_channels || dst_channels == 1 || | 191 RTC_CHECK(dst_channels == src_channels || dst_channels == 1 || |
192 src_channels == 1); | 192 src_channels == 1); |
193 } | 193 } |
194 | 194 |
195 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const { | 195 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const { |
196 RTC_CHECK_EQ(src_size, src_channels() * src_frames()); | 196 RTC_CHECK_EQ(src_size, src_channels() * src_frames()); |
197 RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames()); | 197 RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames()); |
198 } | 198 } |
199 | 199 |
200 } // namespace webrtc | 200 } // namespace webrtc |
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