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 int16_t temp_type = 1; // Default is speech.
44 size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
45 *speech_type = ConvertSpeechType(temp_type);
46 return static_cast<int>(ret);
47 }
48
PacketDuration(const uint8_t * encoded,size_t encoded_len) const49 int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
50 size_t encoded_len) const {
51 // One encoded byte per sample per channel.
52 return static_cast<int>(encoded_len / Channels());
53 }
54
Reset()55 void AudioDecoderPcmA::Reset() {}
56
ParsePayload(rtc::Buffer && payload,uint32_t timestamp)57 std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
58 rtc::Buffer&& payload,
59 uint32_t timestamp) {
60 return LegacyEncodedAudioFrame::SplitBySamples(
61 this, std::move(payload), timestamp, 8 * num_channels_, 8);
62 }
63
SampleRateHz() const64 int AudioDecoderPcmA::SampleRateHz() const {
65 return 8000;
66 }
67
Channels() const68 size_t AudioDecoderPcmA::Channels() const {
69 return num_channels_;
70 }
71
DecodeInternal(const uint8_t * encoded,size_t encoded_len,int sample_rate_hz,int16_t * decoded,SpeechType * speech_type)72 int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
73 size_t encoded_len,
74 int sample_rate_hz,
75 int16_t* decoded,
76 SpeechType* speech_type) {
77 RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
78 int16_t temp_type = 1; // Default is speech.
79 size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
80 *speech_type = ConvertSpeechType(temp_type);
81 return static_cast<int>(ret);
82 }
83
PacketDuration(const uint8_t * encoded,size_t encoded_len) const84 int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
85 size_t encoded_len) const {
86 // One encoded byte per sample per channel.
87 return static_cast<int>(encoded_len / Channels());
88 }
89
90 } // namespace webrtc
91