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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/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h" | 11 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h" |
12 | 12 |
13 #include <limits> | 13 #include <limits> |
14 | 14 |
15 #include "webrtc/base/checks.h" | 15 #include "webrtc/base/checks.h" |
16 #include "webrtc/common_types.h" | 16 #include "webrtc/common_types.h" |
17 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" | 17 #include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h" |
18 | 18 |
19 namespace webrtc { | 19 namespace webrtc { |
20 | 20 |
21 namespace { | 21 namespace { |
22 | 22 |
23 int16_t NumSamplesPerFrame(int num_channels, | 23 int16_t NumSamplesPerFrame(int num_channels, |
24 int frame_size_ms, | 24 int frame_size_ms, |
25 int sample_rate_hz) { | 25 int sample_rate_hz) { |
26 int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000; | 26 int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000; |
27 RTC_CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max()) | 27 RTC_CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max()) |
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133 size_t input_len, | 133 size_t input_len, |
134 uint8_t* encoded) { | 134 uint8_t* encoded) { |
135 return WebRtcG711_EncodeU(audio, input_len, encoded); | 135 return WebRtcG711_EncodeU(audio, input_len, encoded); |
136 } | 136 } |
137 | 137 |
138 int AudioEncoderPcmU::BytesPerSample() const { | 138 int AudioEncoderPcmU::BytesPerSample() const { |
139 return 1; | 139 return 1; |
140 } | 140 } |
141 | 141 |
142 } // namespace webrtc | 142 } // namespace webrtc |
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