OLD | NEW |
1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 |
(...skipping 19 matching lines...) Expand all Loading... |
30 } // namespace | 30 } // namespace |
31 | 31 |
32 bool AudioEncoderPcm::Config::IsOk() const { | 32 bool AudioEncoderPcm::Config::IsOk() const { |
33 return (frame_size_ms % 10 == 0) && (num_channels >= 1); | 33 return (frame_size_ms % 10 == 0) && (num_channels >= 1); |
34 } | 34 } |
35 | 35 |
36 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz) | 36 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz) |
37 : sample_rate_hz_(sample_rate_hz), | 37 : sample_rate_hz_(sample_rate_hz), |
38 num_channels_(config.num_channels), | 38 num_channels_(config.num_channels), |
39 payload_type_(config.payload_type), | 39 payload_type_(config.payload_type), |
40 num_10ms_frames_per_packet_(config.frame_size_ms / 10), | 40 num_10ms_frames_per_packet_( |
| 41 static_cast<size_t>(config.frame_size_ms / 10)), |
41 full_frame_samples_(NumSamplesPerFrame(config.num_channels, | 42 full_frame_samples_(NumSamplesPerFrame(config.num_channels, |
42 config.frame_size_ms, | 43 config.frame_size_ms, |
43 sample_rate_hz_)), | 44 sample_rate_hz_)), |
44 first_timestamp_in_buffer_(0) { | 45 first_timestamp_in_buffer_(0) { |
45 CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz"; | 46 CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz"; |
46 CHECK_EQ(config.frame_size_ms % 10, 0) | 47 CHECK_EQ(config.frame_size_ms % 10, 0) |
47 << "Frame size must be an integer multiple of 10 ms."; | 48 << "Frame size must be an integer multiple of 10 ms."; |
48 speech_buffer_.reserve(full_frame_samples_); | 49 speech_buffer_.reserve(full_frame_samples_); |
49 } | 50 } |
50 | 51 |
51 AudioEncoderPcm::~AudioEncoderPcm() { | 52 AudioEncoderPcm::~AudioEncoderPcm() { |
52 } | 53 } |
53 | 54 |
54 int AudioEncoderPcm::SampleRateHz() const { | 55 int AudioEncoderPcm::SampleRateHz() const { |
55 return sample_rate_hz_; | 56 return sample_rate_hz_; |
56 } | 57 } |
57 | 58 |
58 int AudioEncoderPcm::NumChannels() const { | 59 int AudioEncoderPcm::NumChannels() const { |
59 return num_channels_; | 60 return num_channels_; |
60 } | 61 } |
61 | 62 |
62 size_t AudioEncoderPcm::MaxEncodedBytes() const { | 63 size_t AudioEncoderPcm::MaxEncodedBytes() const { |
63 return full_frame_samples_ * BytesPerSample(); | 64 return full_frame_samples_ * BytesPerSample(); |
64 } | 65 } |
65 | 66 |
66 int AudioEncoderPcm::Num10MsFramesInNextPacket() const { | 67 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const { |
67 return num_10ms_frames_per_packet_; | 68 return num_10ms_frames_per_packet_; |
68 } | 69 } |
69 | 70 |
70 int AudioEncoderPcm::Max10MsFramesInAPacket() const { | 71 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const { |
71 return num_10ms_frames_per_packet_; | 72 return num_10ms_frames_per_packet_; |
72 } | 73 } |
73 | 74 |
74 int AudioEncoderPcm::GetTargetBitrate() const { | 75 int AudioEncoderPcm::GetTargetBitrate() const { |
75 return 8 * BytesPerSample() * SampleRateHz() * NumChannels(); | 76 return 8 * BytesPerSample() * SampleRateHz() * NumChannels(); |
76 } | 77 } |
77 | 78 |
78 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal( | 79 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal( |
79 uint32_t rtp_timestamp, | 80 uint32_t rtp_timestamp, |
80 const int16_t* audio, | 81 const int16_t* audio, |
81 size_t max_encoded_bytes, | 82 size_t max_encoded_bytes, |
82 uint8_t* encoded) { | 83 uint8_t* encoded) { |
83 const int num_samples = SampleRateHz() / 100 * NumChannels(); | 84 const int num_samples = SampleRateHz() / 100 * NumChannels(); |
84 if (speech_buffer_.empty()) { | 85 if (speech_buffer_.empty()) { |
85 first_timestamp_in_buffer_ = rtp_timestamp; | 86 first_timestamp_in_buffer_ = rtp_timestamp; |
86 } | 87 } |
87 for (int i = 0; i < num_samples; ++i) { | 88 for (int i = 0; i < num_samples; ++i) { |
88 speech_buffer_.push_back(audio[i]); | 89 speech_buffer_.push_back(audio[i]); |
89 } | 90 } |
90 if (speech_buffer_.size() < full_frame_samples_) { | 91 if (speech_buffer_.size() < full_frame_samples_) { |
91 return EncodedInfo(); | 92 return EncodedInfo(); |
92 } | 93 } |
93 CHECK_EQ(speech_buffer_.size(), full_frame_samples_); | 94 CHECK_EQ(speech_buffer_.size(), full_frame_samples_); |
94 CHECK_GE(max_encoded_bytes, full_frame_samples_); | 95 CHECK_GE(max_encoded_bytes, full_frame_samples_); |
95 EncodedInfo info; | 96 EncodedInfo info; |
96 info.encoded_timestamp = first_timestamp_in_buffer_; | 97 info.encoded_timestamp = first_timestamp_in_buffer_; |
97 info.payload_type = payload_type_; | 98 info.payload_type = payload_type_; |
98 int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded); | 99 info.encoded_bytes = |
99 CHECK_GE(ret, 0); | 100 EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded); |
100 info.encoded_bytes = static_cast<size_t>(ret); | |
101 speech_buffer_.clear(); | 101 speech_buffer_.clear(); |
102 return info; | 102 return info; |
103 } | 103 } |
104 | 104 |
105 int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, | 105 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, |
106 size_t input_len, | 106 size_t input_len, |
107 uint8_t* encoded) { | 107 uint8_t* encoded) { |
108 return WebRtcG711_EncodeA(audio, static_cast<int16_t>(input_len), encoded); | 108 return WebRtcG711_EncodeA(audio, input_len, encoded); |
109 } | 109 } |
110 | 110 |
111 int AudioEncoderPcmA::BytesPerSample() const { | 111 int AudioEncoderPcmA::BytesPerSample() const { |
112 return 1; | 112 return 1; |
113 } | 113 } |
114 | 114 |
115 int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, | 115 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, |
116 size_t input_len, | 116 size_t input_len, |
117 uint8_t* encoded) { | 117 uint8_t* encoded) { |
118 return WebRtcG711_EncodeU(audio, static_cast<int16_t>(input_len), encoded); | 118 return WebRtcG711_EncodeU(audio, input_len, encoded); |
119 } | 119 } |
120 | 120 |
121 int AudioEncoderPcmU::BytesPerSample() const { | 121 int AudioEncoderPcmU::BytesPerSample() const { |
122 return 1; | 122 return 1; |
123 } | 123 } |
124 | 124 |
125 namespace { | 125 namespace { |
126 template <typename T> | 126 template <typename T> |
127 typename T::Config CreateConfig(const CodecInst& codec_inst) { | 127 typename T::Config CreateConfig(const CodecInst& codec_inst) { |
128 typename T::Config config; | 128 typename T::Config config; |
129 config.frame_size_ms = codec_inst.pacsize / 8; | 129 config.frame_size_ms = codec_inst.pacsize / 8; |
130 config.num_channels = codec_inst.channels; | 130 config.num_channels = codec_inst.channels; |
131 config.payload_type = codec_inst.pltype; | 131 config.payload_type = codec_inst.pltype; |
132 return config; | 132 return config; |
133 } | 133 } |
134 } // namespace | 134 } // namespace |
135 | 135 |
136 AudioEncoderMutablePcmU::AudioEncoderMutablePcmU(const CodecInst& codec_inst) | 136 AudioEncoderMutablePcmU::AudioEncoderMutablePcmU(const CodecInst& codec_inst) |
137 : AudioEncoderMutableImpl<AudioEncoderPcmU>( | 137 : AudioEncoderMutableImpl<AudioEncoderPcmU>( |
138 CreateConfig<AudioEncoderPcmU>(codec_inst)) { | 138 CreateConfig<AudioEncoderPcmU>(codec_inst)) { |
139 } | 139 } |
140 | 140 |
141 AudioEncoderMutablePcmA::AudioEncoderMutablePcmA(const CodecInst& codec_inst) | 141 AudioEncoderMutablePcmA::AudioEncoderMutablePcmA(const CodecInst& codec_inst) |
142 : AudioEncoderMutableImpl<AudioEncoderPcmA>( | 142 : AudioEncoderMutableImpl<AudioEncoderPcmA>( |
143 CreateConfig<AudioEncoderPcmA>(codec_inst)) { | 143 CreateConfig<AudioEncoderPcmA>(codec_inst)) { |
144 } | 144 } |
145 | 145 |
146 } // namespace webrtc | 146 } // namespace webrtc |
OLD | NEW |