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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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| 434 receive_blocks->push_back(block); // Duplicate SSRC, bad fraction_lost. | 434 receive_blocks->push_back(block); // Duplicate SSRC, bad fraction_lost. |
| 435 return 0; | 435 return 0; |
| 436 } | 436 } |
| 437 int SetNACKStatus(int channel, bool enable, int maxNoPackets) override { | 437 int SetNACKStatus(int channel, bool enable, int maxNoPackets) override { |
| 438 return -1; | 438 return -1; |
| 439 } | 439 } |
| 440 | 440 |
| 441 // VoEVideoSync | 441 // VoEVideoSync |
| 442 int GetPlayoutBufferSize(int& buffer_ms) override { return -1; } | 442 int GetPlayoutBufferSize(int& buffer_ms) override { return -1; } |
| 443 int SetMinimumPlayoutDelay(int channel, int delay_ms) override { return -1; } | 443 int SetMinimumPlayoutDelay(int channel, int delay_ms) override { return -1; } |
| 444 int SetInitialPlayoutDelay(int channel, int delay_ms) override { return -1; } | |
| 445 int GetDelayEstimate(int channel, | 444 int GetDelayEstimate(int channel, |
| 446 int* jitter_buffer_delay_ms, | 445 int* jitter_buffer_delay_ms, |
| 447 int* playout_buffer_delay_ms) override { | 446 int* playout_buffer_delay_ms) override { |
| 448 EXPECT_EQ(channel, kRecvChannelId); | 447 EXPECT_EQ(channel, kRecvChannelId); |
| 449 *jitter_buffer_delay_ms = kRecvJitterBufferDelay; | 448 *jitter_buffer_delay_ms = kRecvJitterBufferDelay; |
| 450 *playout_buffer_delay_ms = kRecvPlayoutBufferDelay; | 449 *playout_buffer_delay_ms = kRecvPlayoutBufferDelay; |
| 451 return 0; | 450 return 0; |
| 452 } | 451 } |
| 453 int GetLeastRequiredDelayMs(int channel) const override { return -1; } | 452 int GetLeastRequiredDelayMs(int channel) const override { return -1; } |
| 454 int SetInitTimestamp(int channel, unsigned int timestamp) override { | 453 int SetInitTimestamp(int channel, unsigned int timestamp) override { |
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| 495 return -1; | 494 return -1; |
| 496 } | 495 } |
| 497 int GetOutputVolumePan(int channel, float& left, float& right) override { | 496 int GetOutputVolumePan(int channel, float& left, float& right) override { |
| 498 return -1; | 497 return -1; |
| 499 } | 498 } |
| 500 }; | 499 }; |
| 501 } // namespace test | 500 } // namespace test |
| 502 } // namespace webrtc | 501 } // namespace webrtc |
| 503 | 502 |
| 504 #endif // WEBRTC_AUDIO_FAKE_VOICE_ENGINE_H_ | 503 #endif // WEBRTC_AUDIO_FAKE_VOICE_ENGINE_H_ |
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