1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
12
13 #include <utility>
14
15 #include "modules/audio_coding/codecs/g711/g711_interface.h"
16 #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
17
18 namespace webrtc {
19
Reset()20 void AudioDecoderPcmU::Reset() {}
21
ParsePayload(rtc::Buffer && payload,uint32_t timestamp)22 std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
23 rtc::Buffer&& payload,
24 uint32_t timestamp) {
25 return LegacyEncodedAudioFrame::SplitBySamples(
26 this, std::move(payload), timestamp, 8 * num_channels_, 8);
27 }
28
SampleRateHz() const29 int AudioDecoderPcmU::SampleRateHz() const {
30 return 8000;
31 }
32
Channels() const33 size_t AudioDecoderPcmU::Channels() const {
34 return num_channels_;
35 }
36
DecodeInternal(const uint8_t * encoded,size_t encoded_len,int sample_rate_hz,int16_t * decoded,SpeechType * speech_type)37 int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
38 size_t encoded_len,
39 int sample_rate_hz,
40 int16_t* decoded,
41 SpeechType* speech_type) {
42 RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
43 // Adjust the encoded length down to ensure the same number of samples in each
44 // channel.
45 const size_t encoded_len_adjusted =
46 PacketDuration(encoded, encoded_len) *
47 Channels(); // 1 byte per sample per channel
48 int16_t temp_type = 1; // Default is speech.
49 size_t ret =
50 WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type);
51 *speech_type = ConvertSpeechType(temp_type);
52 return static_cast<int>(ret);
53 }
54
PacketDuration(const uint8_t * encoded,size_t encoded_len) const55 int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
56 size_t encoded_len) const {
57 // One encoded byte per sample per channel.
58 return static_cast<int>(encoded_len / Channels());
59 }
60
Reset()61 void AudioDecoderPcmA::Reset() {}
62
ParsePayload(rtc::Buffer && payload,uint32_t timestamp)63 std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
64 rtc::Buffer&& payload,
65 uint32_t timestamp) {
66 return LegacyEncodedAudioFrame::SplitBySamples(
67 this, std::move(payload), timestamp, 8 * num_channels_, 8);
68 }
69
SampleRateHz() const70 int AudioDecoderPcmA::SampleRateHz() const {
71 return 8000;
72 }
73
Channels() const74 size_t AudioDecoderPcmA::Channels() const {
75 return num_channels_;
76 }
77
DecodeInternal(const uint8_t * encoded,size_t encoded_len,int sample_rate_hz,int16_t * decoded,SpeechType * speech_type)78 int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
79 size_t encoded_len,
80 int sample_rate_hz,
81 int16_t* decoded,
82 SpeechType* speech_type) {
83 RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
84 // Adjust the encoded length down to ensure the same number of samples in each
85 // channel.
86 const size_t encoded_len_adjusted =
87 PacketDuration(encoded, encoded_len) *
88 Channels(); // 1 byte per sample per channel
89 int16_t temp_type = 1; // Default is speech.
90 size_t ret =
91 WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type);
92 *speech_type = ConvertSpeechType(temp_type);
93 return static_cast<int>(ret);
94 }
95
PacketDuration(const uint8_t * encoded,size_t encoded_len) const96 int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
97 size_t encoded_len) const {
98 // One encoded byte per sample per channel.
99 return static_cast<int>(encoded_len / Channels());
100 }
101
102 } // namespace webrtc
103