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Side by Side Diff: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

Issue 1438663003: modules/audio_coding: Remove some codec include dirs (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase again Created 5 years, 1 month ago
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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())
(...skipping 105 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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