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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 |
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24 class AudioEncoderOpus final : public AudioEncoder { | 24 class AudioEncoderOpus final : public AudioEncoder { |
25 public: | 25 public: |
26 enum ApplicationMode { | 26 enum ApplicationMode { |
27 kVoip = 0, | 27 kVoip = 0, |
28 kAudio = 1, | 28 kAudio = 1, |
29 }; | 29 }; |
30 | 30 |
31 struct Config { | 31 struct Config { |
32 bool IsOk() const; | 32 bool IsOk() const; |
33 int frame_size_ms = 20; | 33 int frame_size_ms = 20; |
34 int num_channels = 1; | 34 size_t num_channels = 1; |
35 int payload_type = 120; | 35 int payload_type = 120; |
36 ApplicationMode application = kVoip; | 36 ApplicationMode application = kVoip; |
37 int bitrate_bps = 64000; | 37 int bitrate_bps = 64000; |
38 bool fec_enabled = false; | 38 bool fec_enabled = false; |
39 int max_playback_rate_hz = 48000; | 39 int max_playback_rate_hz = 48000; |
40 int complexity = kDefaultComplexity; | 40 int complexity = kDefaultComplexity; |
41 bool dtx_enabled = false; | 41 bool dtx_enabled = false; |
42 | 42 |
43 private: | 43 private: |
44 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) | 44 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) |
45 // If we are on Android, iOS and/or ARM, use a lower complexity setting as | 45 // If we are on Android, iOS and/or ARM, use a lower complexity setting as |
46 // default, to save encoder complexity. | 46 // default, to save encoder complexity. |
47 static const int kDefaultComplexity = 5; | 47 static const int kDefaultComplexity = 5; |
48 #else | 48 #else |
49 static const int kDefaultComplexity = 9; | 49 static const int kDefaultComplexity = 9; |
50 #endif | 50 #endif |
51 }; | 51 }; |
52 | 52 |
53 explicit AudioEncoderOpus(const Config& config); | 53 explicit AudioEncoderOpus(const Config& config); |
54 explicit AudioEncoderOpus(const CodecInst& codec_inst); | 54 explicit AudioEncoderOpus(const CodecInst& codec_inst); |
55 ~AudioEncoderOpus() override; | 55 ~AudioEncoderOpus() override; |
56 | 56 |
57 size_t MaxEncodedBytes() const override; | 57 size_t MaxEncodedBytes() const override; |
58 int SampleRateHz() const override; | 58 int SampleRateHz() const override; |
59 int NumChannels() const override; | 59 size_t NumChannels() const override; |
60 size_t Num10MsFramesInNextPacket() const override; | 60 size_t Num10MsFramesInNextPacket() const override; |
61 size_t Max10MsFramesInAPacket() const override; | 61 size_t Max10MsFramesInAPacket() const override; |
62 int GetTargetBitrate() const override; | 62 int GetTargetBitrate() const override; |
63 | 63 |
64 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, | 64 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
65 rtc::ArrayView<const int16_t> audio, | 65 rtc::ArrayView<const int16_t> audio, |
66 size_t max_encoded_bytes, | 66 size_t max_encoded_bytes, |
67 uint8_t* encoded) override; | 67 uint8_t* encoded) override; |
68 | 68 |
69 void Reset() override; | 69 void Reset() override; |
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93 double packet_loss_rate_; | 93 double packet_loss_rate_; |
94 std::vector<int16_t> input_buffer_; | 94 std::vector<int16_t> input_buffer_; |
95 OpusEncInst* inst_; | 95 OpusEncInst* inst_; |
96 uint32_t first_timestamp_in_buffer_; | 96 uint32_t first_timestamp_in_buffer_; |
97 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); | 97 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); |
98 }; | 98 }; |
99 | 99 |
100 } // namespace webrtc | 100 } // namespace webrtc |
101 | 101 |
102 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ | 102 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ |
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