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Side by Side Diff: webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h

Issue 1769883002: Remove the type parameter to NetEq::GetAudio (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@acm-rec-delete-vad
Patch Set: After review Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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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36 36
37 // Inserts a new packet with |rtp_header| and |payload| of 37 // Inserts a new packet with |rtp_header| and |payload| of
38 // |payload_size_bytes| bytes. The |receive_timestamp| is an indication 38 // |payload_size_bytes| bytes. The |receive_timestamp| is an indication
39 // of the time when the packet was received, and should be measured with 39 // of the time when the packet was received, and should be measured with
40 // the same tick rate as the RTP timestamp of the current payload. 40 // the same tick rate as the RTP timestamp of the current payload.
41 virtual void InsertPacket(WebRtcRTPHeader rtp_header, 41 virtual void InsertPacket(WebRtcRTPHeader rtp_header,
42 rtc::ArrayView<const uint8_t> payload, 42 rtc::ArrayView<const uint8_t> payload,
43 uint32_t receive_timestamp); 43 uint32_t receive_timestamp);
44 44
45 // Get 10 ms of audio data. 45 // Get 10 ms of audio data.
46 void GetOutputAudio(AudioFrame* output, NetEqOutputType* output_type); 46 void GetOutputAudio(AudioFrame* output);
47 47
48 NetEq* neteq() { return neteq_.get(); } 48 NetEq* neteq() { return neteq_.get(); }
49 49
50 private: 50 private:
51 NetEqDecoder codec_; 51 NetEqDecoder codec_;
52 std::string name_ = "dummy name"; 52 std::string name_ = "dummy name";
53 AudioDecoder* decoder_; 53 AudioDecoder* decoder_;
54 int sample_rate_hz_; 54 int sample_rate_hz_;
55 size_t channels_; 55 size_t channels_;
56 std::unique_ptr<NetEq> neteq_; 56 std::unique_ptr<NetEq> neteq_;
57 }; 57 };
58 58
59 } // namespace test 59 } // namespace test
60 } // namespace webrtc 60 } // namespace webrtc
61 61
62 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H _ 62 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H _
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