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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 |
11 #include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h" | 11 #include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h" |
12 | 12 |
13 #include <limits> | 13 #include <limits> |
14 | 14 |
15 #include "webrtc/base/checks.h" | 15 #include "webrtc/base/checks.h" |
16 #include "webrtc/common_types.h" | 16 #include "webrtc/common_types.h" |
17 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" | 17 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" |
18 | 18 |
19 namespace webrtc { | 19 namespace webrtc { |
20 | 20 |
21 namespace { | 21 namespace { |
22 | |
22 int16_t NumSamplesPerFrame(int num_channels, | 23 int16_t NumSamplesPerFrame(int num_channels, |
23 int frame_size_ms, | 24 int frame_size_ms, |
24 int sample_rate_hz) { | 25 int sample_rate_hz) { |
25 int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000; | 26 int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000; |
26 CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max()) | 27 CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max()) |
27 << "Frame size too large."; | 28 << "Frame size too large."; |
28 return static_cast<int16_t>(samples_per_frame); | 29 return static_cast<int16_t>(samples_per_frame); |
29 } | 30 } |
31 | |
32 template <typename T> | |
33 typename T::Config CreateConfig(const CodecInst& codec_inst) { | |
34 typename T::Config config; | |
35 config.frame_size_ms = codec_inst.pacsize / 8; | |
36 config.num_channels = codec_inst.channels; | |
37 config.payload_type = codec_inst.pltype; | |
38 return config; | |
39 } | |
40 | |
30 } // namespace | 41 } // namespace |
31 | 42 |
32 bool AudioEncoderPcm::Config::IsOk() const { | 43 bool AudioEncoderPcm::Config::IsOk() const { |
33 return (frame_size_ms % 10 == 0) && (num_channels >= 1); | 44 return (frame_size_ms % 10 == 0) && (num_channels >= 1); |
34 } | 45 } |
35 | 46 |
36 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz) | 47 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz) |
37 : sample_rate_hz_(sample_rate_hz), | 48 : sample_rate_hz_(sample_rate_hz), |
38 num_channels_(config.num_channels), | 49 num_channels_(config.num_channels), |
39 payload_type_(config.payload_type), | 50 payload_type_(config.payload_type), |
40 num_10ms_frames_per_packet_( | 51 num_10ms_frames_per_packet_( |
41 static_cast<size_t>(config.frame_size_ms / 10)), | 52 static_cast<size_t>(config.frame_size_ms / 10)), |
42 full_frame_samples_(NumSamplesPerFrame(config.num_channels, | 53 full_frame_samples_(NumSamplesPerFrame(config.num_channels, |
43 config.frame_size_ms, | 54 config.frame_size_ms, |
44 sample_rate_hz_)), | 55 sample_rate_hz_)), |
45 first_timestamp_in_buffer_(0) { | 56 first_timestamp_in_buffer_(0) { |
46 CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz"; | 57 CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz"; |
47 CHECK_EQ(config.frame_size_ms % 10, 0) | 58 CHECK_EQ(config.frame_size_ms % 10, 0) |
48 << "Frame size must be an integer multiple of 10 ms."; | 59 << "Frame size must be an integer multiple of 10 ms."; |
49 speech_buffer_.reserve(full_frame_samples_); | 60 speech_buffer_.reserve(full_frame_samples_); |
50 } | 61 } |
51 | 62 |
52 AudioEncoderPcm::~AudioEncoderPcm() { | 63 AudioEncoderPcm::~AudioEncoderPcm() = default; |
64 | |
65 size_t AudioEncoderPcm::MaxEncodedBytes() const { | |
66 return full_frame_samples_ * BytesPerSample(); | |
53 } | 67 } |
54 | 68 |
55 int AudioEncoderPcm::SampleRateHz() const { | 69 int AudioEncoderPcm::SampleRateHz() const { |
56 return sample_rate_hz_; | 70 return sample_rate_hz_; |
57 } | 71 } |
58 | 72 |
59 int AudioEncoderPcm::NumChannels() const { | 73 int AudioEncoderPcm::NumChannels() const { |
60 return num_channels_; | 74 return num_channels_; |
61 } | 75 } |
62 | 76 |
63 size_t AudioEncoderPcm::MaxEncodedBytes() const { | |
64 return full_frame_samples_ * BytesPerSample(); | |
65 } | |
66 | |
67 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const { | 77 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const { |
68 return num_10ms_frames_per_packet_; | 78 return num_10ms_frames_per_packet_; |
69 } | 79 } |
70 | 80 |
71 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const { | 81 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const { |
72 return num_10ms_frames_per_packet_; | 82 return num_10ms_frames_per_packet_; |
73 } | 83 } |
74 | 84 |
75 int AudioEncoderPcm::GetTargetBitrate() const { | 85 int AudioEncoderPcm::GetTargetBitrate() const { |
76 return 8 * BytesPerSample() * SampleRateHz() * NumChannels(); | 86 return 8 * BytesPerSample() * SampleRateHz() * NumChannels(); |
(...skipping 18 matching lines...) Expand all Loading... | |
95 CHECK_GE(max_encoded_bytes, full_frame_samples_); | 105 CHECK_GE(max_encoded_bytes, full_frame_samples_); |
96 EncodedInfo info; | 106 EncodedInfo info; |
97 info.encoded_timestamp = first_timestamp_in_buffer_; | 107 info.encoded_timestamp = first_timestamp_in_buffer_; |
98 info.payload_type = payload_type_; | 108 info.payload_type = payload_type_; |
99 info.encoded_bytes = | 109 info.encoded_bytes = |
100 EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded); | 110 EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded); |
101 speech_buffer_.clear(); | 111 speech_buffer_.clear(); |
102 return info; | 112 return info; |
103 } | 113 } |
104 | 114 |
115 void AudioEncoderPcm::Reset() { | |
116 speech_buffer_.clear(); | |
117 } | |
118 | |
119 AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst) | |
hlundin-webrtc
2015/09/07 20:00:01
With this construct, there is no way we to verify
kwiberg-webrtc
2015/09/08 10:47:45
If we wanted to handle these errors, I guess the b
hlundin-webrtc
2015/09/08 11:21:26
Acknowledged.
| |
120 : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {} | |
121 | |
105 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, | 122 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, |
106 size_t input_len, | 123 size_t input_len, |
107 uint8_t* encoded) { | 124 uint8_t* encoded) { |
108 return WebRtcG711_EncodeA(audio, input_len, encoded); | 125 return WebRtcG711_EncodeA(audio, input_len, encoded); |
109 } | 126 } |
110 | 127 |
111 int AudioEncoderPcmA::BytesPerSample() const { | 128 int AudioEncoderPcmA::BytesPerSample() const { |
112 return 1; | 129 return 1; |
113 } | 130 } |
114 | 131 |
132 AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst) | |
133 : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {} | |
134 | |
115 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, | 135 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, |
116 size_t input_len, | 136 size_t input_len, |
117 uint8_t* encoded) { | 137 uint8_t* encoded) { |
118 return WebRtcG711_EncodeU(audio, input_len, encoded); | 138 return WebRtcG711_EncodeU(audio, input_len, encoded); |
119 } | 139 } |
120 | 140 |
121 int AudioEncoderPcmU::BytesPerSample() const { | 141 int AudioEncoderPcmU::BytesPerSample() const { |
122 return 1; | 142 return 1; |
123 } | 143 } |
124 | 144 |
125 namespace { | |
126 template <typename T> | |
127 typename T::Config CreateConfig(const CodecInst& codec_inst) { | |
128 typename T::Config config; | |
129 config.frame_size_ms = codec_inst.pacsize / 8; | |
130 config.num_channels = codec_inst.channels; | |
131 config.payload_type = codec_inst.pltype; | |
132 return config; | |
133 } | |
134 } // namespace | |
135 | |
136 AudioEncoderMutablePcmU::AudioEncoderMutablePcmU(const CodecInst& codec_inst) | |
137 : AudioEncoderMutableImpl<AudioEncoderPcmU>( | |
138 CreateConfig<AudioEncoderPcmU>(codec_inst)) { | |
139 } | |
140 | |
141 AudioEncoderMutablePcmA::AudioEncoderMutablePcmA(const CodecInst& codec_inst) | |
142 : AudioEncoderMutableImpl<AudioEncoderPcmA>( | |
143 CreateConfig<AudioEncoderPcmA>(codec_inst)) { | |
144 } | |
145 | |
146 } // namespace webrtc | 145 } // namespace webrtc |
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