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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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/pcm16b/audio_decoder_pcm16b.h" | 11 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h" |
12 | 12 |
13 #include "webrtc/base/checks.h" | 13 #include "webrtc/base/checks.h" |
14 #include "webrtc/modules/audio_coding/codecs/split_by_samples.h" | |
14 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" | 15 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" |
15 | 16 |
16 namespace webrtc { | 17 namespace webrtc { |
17 | 18 |
18 AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels) | 19 AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels) |
19 : sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) { | 20 : sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) { |
20 RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 || | 21 RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 || |
21 sample_rate_hz == 32000 || sample_rate_hz == 48000) | 22 sample_rate_hz == 32000 || sample_rate_hz == 48000) |
22 << "Unsupported sample rate " << sample_rate_hz; | 23 << "Unsupported sample rate " << sample_rate_hz; |
23 RTC_DCHECK_GE(num_channels, 1u); | 24 RTC_DCHECK_GE(num_channels, 1u); |
(...skipping 13 matching lines...) Expand all Loading... | |
37 size_t encoded_len, | 38 size_t encoded_len, |
38 int sample_rate_hz, | 39 int sample_rate_hz, |
39 int16_t* decoded, | 40 int16_t* decoded, |
40 SpeechType* speech_type) { | 41 SpeechType* speech_type) { |
41 RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz); | 42 RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz); |
42 size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded); | 43 size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded); |
43 *speech_type = ConvertSpeechType(1); | 44 *speech_type = ConvertSpeechType(1); |
44 return static_cast<int>(ret); | 45 return static_cast<int>(ret); |
45 } | 46 } |
46 | 47 |
48 std::vector<AudioDecoder::PacketSplit> AudioDecoderPcm16B::SplitPacket( | |
49 rtc::ArrayView<const uint8_t> payload) const { | |
50 // TODO(ossu): Investigate if we can ever get 44.1KHz audio here, in which | |
hlundin-webrtc
2016/09/09 12:52:52
Short answer: no. If codecs should be allowed to r
kwiberg-webrtc
2016/09/12 02:11:01
Also, space between the number and the unit. But I
ossu
2016/09/12 11:26:37
The comment is only here for this to get caught in
kwiberg-webrtc
2016/09/13 12:23:37
Well, if you didn't want pedantic complaints about
ossu
2016/09/13 14:25:55
Acknowledged.
hlundin-webrtc
2016/09/15 08:49:14
:)
| |
51 // case rounding will break. Consider replacing with | |
52 // CheckedDivExact to catch that happening. | |
53 return internal::SplitBySamples(payload, | |
54 sample_rate_hz_ * 2 * num_channels_ / 1000, | |
55 sample_rate_hz_ / 1000); | |
56 } | |
57 | |
47 int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded, | 58 int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded, |
48 size_t encoded_len) const { | 59 size_t encoded_len) const { |
49 // Two encoded byte per sample per channel. | 60 // Two encoded byte per sample per channel. |
50 return static_cast<int>(encoded_len / (2 * Channels())); | 61 return static_cast<int>(encoded_len / (2 * Channels())); |
51 } | 62 } |
52 | 63 |
53 } // namespace webrtc | 64 } // namespace webrtc |
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