1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
|
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include <utility>
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "rtc_base/checks.h"
namespace webrtc {
AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
: sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
RTC_DCHECK_GE(num_channels, 1);
}
void AudioDecoderPcm16B::Reset() {}
int AudioDecoderPcm16B::SampleRateHz() const {
return sample_rate_hz_;
}
size_t AudioDecoderPcm16B::Channels() const {
return num_channels_;
}
int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
// Adjust the encoded length down to ensure the same number of samples in each
// channel.
const size_t encoded_len_adjusted =
PacketDuration(encoded, encoded_len) * 2 *
Channels(); // 2 bytes per sample per channel
size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len_adjusted, decoded);
*speech_type = ConvertSpeechType(1);
return static_cast<int>(ret);
}
std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000);
return LegacyEncodedAudioFrame::SplitBySamples(
this, std::move(payload), timestamp, samples_per_ms * 2 * num_channels_,
samples_per_ms);
}
int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// Two encoded byte per sample per channel.
return static_cast<int>(encoded_len / (2 * Channels()));
}
int AudioDecoderPcm16B::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
return PacketDuration(encoded, encoded_len);
}
} // namespace webrtc
|
