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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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| 399 int ret = neteq_->GetAudio(&out_frame_, &muted); | 399 int ret = neteq_->GetAudio(&out_frame_, &muted); |
| 400 RTC_CHECK(!muted); | 400 RTC_CHECK(!muted); |
| 401 | 401 |
| 402 if (ret != NetEq::kOK) { | 402 if (ret != NetEq::kOK) { |
| 403 return -1; | 403 return -1; |
| 404 } else { | 404 } else { |
| 405 RTC_DCHECK_EQ(out_frame_.num_channels_, channels_); | 405 RTC_DCHECK_EQ(out_frame_.num_channels_, channels_); |
| 406 RTC_DCHECK_EQ(out_frame_.samples_per_channel_, | 406 RTC_DCHECK_EQ(out_frame_.samples_per_channel_, |
| 407 static_cast<size_t>(kOutputSizeMs * out_sampling_khz_)); | 407 static_cast<size_t>(kOutputSizeMs * out_sampling_khz_)); |
| 408 RTC_CHECK(output_->WriteArray( | 408 RTC_CHECK(output_->WriteArray( |
| 409 out_frame_.data_, | 409 out_frame_.data(), |
| 410 out_frame_.samples_per_channel_ * out_frame_.num_channels_)); | 410 out_frame_.samples_per_channel_ * out_frame_.num_channels_)); |
| 411 return static_cast<int>(out_frame_.samples_per_channel_); | 411 return static_cast<int>(out_frame_.samples_per_channel_); |
| 412 } | 412 } |
| 413 } | 413 } |
| 414 | 414 |
| 415 void NetEqQualityTest::Simulate() { | 415 void NetEqQualityTest::Simulate() { |
| 416 int audio_size_samples; | 416 int audio_size_samples; |
| 417 | 417 |
| 418 while (decoded_time_ms_ < FLAGS_runtime_ms) { | 418 while (decoded_time_ms_ < FLAGS_runtime_ms) { |
| 419 // Assume 10 packets in packets buffer. | 419 // Assume 10 packets in packets buffer. |
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| 432 } | 432 } |
| 433 } | 433 } |
| 434 Log() << "Average bit rate was " | 434 Log() << "Average bit rate was " |
| 435 << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms | 435 << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms |
| 436 << " kbps" | 436 << " kbps" |
| 437 << std::endl; | 437 << std::endl; |
| 438 } | 438 } |
| 439 | 439 |
| 440 } // namespace test | 440 } // namespace test |
| 441 } // namespace webrtc | 441 } // namespace webrtc |
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