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Side by Side Diff: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h

Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed use of <functional> and moved MockAudioEncoderHelper things into the respective MockAudioEn… Created 4 years, 9 months ago
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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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34 }; 34 };
35 35
36 ~AudioEncoderPcm() override; 36 ~AudioEncoderPcm() override;
37 37
38 size_t MaxEncodedBytes() const override; 38 size_t MaxEncodedBytes() const override;
39 int SampleRateHz() const override; 39 int SampleRateHz() const override;
40 size_t NumChannels() const override; 40 size_t NumChannels() const override;
41 size_t Num10MsFramesInNextPacket() const override; 41 size_t Num10MsFramesInNextPacket() const override;
42 size_t Max10MsFramesInAPacket() const override; 42 size_t Max10MsFramesInAPacket() const override;
43 int GetTargetBitrate() const override; 43 int GetTargetBitrate() const override;
44 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
45 rtc::ArrayView<const int16_t> audio,
46 size_t max_encoded_bytes,
47 uint8_t* encoded) override;
48 void Reset() override; 44 void Reset() override;
49 45
50 protected: 46 protected:
51 AudioEncoderPcm(const Config& config, int sample_rate_hz); 47 AudioEncoderPcm(const Config& config, int sample_rate_hz);
52 48
49 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
50 rtc::ArrayView<const int16_t> audio,
51 rtc::Buffer* encoded) override;
52
53 virtual size_t EncodeCall(const int16_t* audio, 53 virtual size_t EncodeCall(const int16_t* audio,
54 size_t input_len, 54 size_t input_len,
55 uint8_t* encoded) = 0; 55 uint8_t* encoded) = 0;
56 56
57 virtual size_t BytesPerSample() const = 0; 57 virtual size_t BytesPerSample() const = 0;
58 58
59 private: 59 private:
60 const int sample_rate_hz_; 60 const int sample_rate_hz_;
61 const size_t num_channels_; 61 const size_t num_channels_;
62 const int payload_type_; 62 const int payload_type_;
(...skipping 45 matching lines...) Expand 10 before | Expand all | Expand 10 after
108 size_t BytesPerSample() const override; 108 size_t BytesPerSample() const override;
109 109
110 private: 110 private:
111 static const int kSampleRateHz = 8000; 111 static const int kSampleRateHz = 8000;
112 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU); 112 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
113 }; 113 };
114 114
115 } // namespace webrtc 115 } // namespace webrtc
116 116
117 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_ 117 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
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