1 /*
2 * Copyright (c) 2014 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 "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
12
13 #include <cstring>
14 #include <limits>
15 #include "webrtc/base/checks.h"
16 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
17
18 namespace webrtc {
19
20 namespace {
21
22 const int kSampleRateHz = 8000;
23
24 } // namespace
25
AudioEncoderIlbc(const Config & config)26 AudioEncoderIlbc::AudioEncoderIlbc(const Config& config)
27 : payload_type_(config.payload_type),
28 num_10ms_frames_per_packet_(config.frame_size_ms / 10),
29 num_10ms_frames_buffered_(0) {
30 CHECK(config.frame_size_ms == 20 || config.frame_size_ms == 30 ||
31 config.frame_size_ms == 40 || config.frame_size_ms == 60)
32 << "Frame size must be 20, 30, 40, or 60 ms.";
33 DCHECK_LE(kSampleRateHz / 100 * num_10ms_frames_per_packet_,
34 kMaxSamplesPerPacket);
35 CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
36 const int encoder_frame_size_ms = config.frame_size_ms > 30
37 ? config.frame_size_ms / 2
38 : config.frame_size_ms;
39 CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
40 }
41
~AudioEncoderIlbc()42 AudioEncoderIlbc::~AudioEncoderIlbc() {
43 CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
44 }
45
SampleRateHz() const46 int AudioEncoderIlbc::SampleRateHz() const {
47 return kSampleRateHz;
48 }
49
NumChannels() const50 int AudioEncoderIlbc::NumChannels() const {
51 return 1;
52 }
53
MaxEncodedBytes() const54 size_t AudioEncoderIlbc::MaxEncodedBytes() const {
55 return RequiredOutputSizeBytes();
56 }
57
Num10MsFramesInNextPacket() const58 int AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
59 return num_10ms_frames_per_packet_;
60 }
61
Max10MsFramesInAPacket() const62 int AudioEncoderIlbc::Max10MsFramesInAPacket() const {
63 return num_10ms_frames_per_packet_;
64 }
65
EncodeInternal(uint32_t rtp_timestamp,const int16_t * audio,size_t max_encoded_bytes,uint8_t * encoded)66 AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
67 uint32_t rtp_timestamp,
68 const int16_t* audio,
69 size_t max_encoded_bytes,
70 uint8_t* encoded) {
71 DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
72
73 // Save timestamp if starting a new packet.
74 if (num_10ms_frames_buffered_ == 0)
75 first_timestamp_in_buffer_ = rtp_timestamp;
76
77 // Buffer input.
78 std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_,
79 audio,
80 kSampleRateHz / 100 * sizeof(audio[0]));
81
82 // If we don't yet have enough buffered input for a whole packet, we're done
83 // for now.
84 if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
85 return EncodedInfo();
86 }
87
88 // Encode buffered input.
89 DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
90 num_10ms_frames_buffered_ = 0;
91 const int output_len = WebRtcIlbcfix_Encode(
92 encoder_,
93 input_buffer_,
94 kSampleRateHz / 100 * num_10ms_frames_per_packet_,
95 encoded);
96 CHECK_GE(output_len, 0);
97 EncodedInfo info;
98 info.encoded_bytes = output_len;
99 DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
100 info.encoded_timestamp = first_timestamp_in_buffer_;
101 info.payload_type = payload_type_;
102 return info;
103 }
104
RequiredOutputSizeBytes() const105 size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const {
106 switch (num_10ms_frames_per_packet_) {
107 case 2: return 38;
108 case 3: return 50;
109 case 4: return 2 * 38;
110 case 6: return 2 * 50;
111 default: FATAL();
112 }
113 }
114
115 } // namespace webrtc
116