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1 /* | 1 /* |
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2017 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/api/audio_codecs/L16/audio_encoder_L16.h" | 11 #include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h" |
12 | 12 |
13 #include "webrtc/common_types.h" | 13 #include "webrtc/common_types.h" |
14 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" | 14 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" |
15 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h" | 15 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h" |
16 #include "webrtc/rtc_base/ptr_util.h" | 16 #include "webrtc/rtc_base/ptr_util.h" |
17 | 17 |
18 namespace webrtc { | 18 namespace webrtc { |
19 | 19 |
20 rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig( | 20 rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig( |
21 const SdpAudioFormat& format) { | 21 const SdpAudioFormat& format) { |
22 Config config; | 22 Config config; |
23 config.sample_rate_hz = format.clockrate_hz; | 23 config.sample_rate_hz = format.clockrate_hz; |
24 config.num_channels = format.num_channels; | 24 config.num_channels = rtc::dchecked_cast<size_t>(format.num_channels); |
kwiberg-webrtc
2017/08/22 17:40:24
checked_cast is probably better here, since this v
| |
25 return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk() | 25 return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk() |
26 ? rtc::Optional<Config>(config) | 26 ? rtc::Optional<Config>(config) |
27 : rtc::Optional<Config>(); | 27 : rtc::Optional<Config>(); |
28 } | 28 } |
29 | 29 |
30 void AudioEncoderL16::AppendSupportedEncoders( | 30 void AudioEncoderL16::AppendSupportedEncoders( |
31 std::vector<AudioCodecSpec>* specs) { | 31 std::vector<AudioCodecSpec>* specs) { |
32 Pcm16BAppendSupportedCodecSpecs(specs); | 32 Pcm16BAppendSupportedCodecSpecs(specs); |
33 } | 33 } |
34 | 34 |
(...skipping 11 matching lines...) Expand all Loading... | |
46 RTC_DCHECK(config.IsOk()); | 46 RTC_DCHECK(config.IsOk()); |
47 AudioEncoderPcm16B::Config c; | 47 AudioEncoderPcm16B::Config c; |
48 c.sample_rate_hz = config.sample_rate_hz; | 48 c.sample_rate_hz = config.sample_rate_hz; |
49 c.num_channels = config.num_channels; | 49 c.num_channels = config.num_channels; |
50 c.frame_size_ms = config.frame_size_ms; | 50 c.frame_size_ms = config.frame_size_ms; |
51 c.payload_type = payload_type; | 51 c.payload_type = payload_type; |
52 return rtc::MakeUnique<AudioEncoderPcm16B>(c); | 52 return rtc::MakeUnique<AudioEncoderPcm16B>(c); |
53 } | 53 } |
54 | 54 |
55 } // namespace webrtc | 55 } // namespace webrtc |
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