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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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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| 329 rtp_timestamp_ += static_cast<uint32_t>(frame_length); | 329 rtp_timestamp_ += static_cast<uint32_t>(frame_length); |
| 330 read_samples += frame_length * channels; | 330 read_samples += frame_length * channels; |
| 331 } | 331 } |
| 332 if (read_samples == written_samples) { | 332 if (read_samples == written_samples) { |
| 333 read_samples = 0; | 333 read_samples = 0; |
| 334 written_samples = 0; | 334 written_samples = 0; |
| 335 } | 335 } |
| 336 } | 336 } |
| 337 | 337 |
| 338 // Run received side of ACM. | 338 // Run received side of ACM. |
| 339 ASSERT_EQ(0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame)); | 339 bool muted; |
| 340 ASSERT_EQ( |
| 341 0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted)); |
| 342 ASSERT_FALSE(muted); |
| 340 | 343 |
| 341 // Write output speech to file. | 344 // Write output speech to file. |
| 342 out_file_.Write10MsData( | 345 out_file_.Write10MsData( |
| 343 audio_frame.data_, | 346 audio_frame.data_, |
| 344 audio_frame.samples_per_channel_ * audio_frame.num_channels_); | 347 audio_frame.samples_per_channel_ * audio_frame.num_channels_); |
| 345 | 348 |
| 346 // Write stand-alone speech to file. | 349 // Write stand-alone speech to file. |
| 347 out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels); | 350 out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels); |
| 348 | 351 |
| 349 if (audio_frame.timestamp_ > start_time_stamp) { | 352 if (audio_frame.timestamp_ > start_time_stamp) { |
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| 374 out_file_.Open(file_name, 48000, "wb"); | 377 out_file_.Open(file_name, 48000, "wb"); |
| 375 file_stream.str(""); | 378 file_stream.str(""); |
| 376 file_name = file_stream.str(); | 379 file_name = file_stream.str(); |
| 377 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" | 380 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" |
| 378 << test_number << ".pcm"; | 381 << test_number << ".pcm"; |
| 379 file_name = file_stream.str(); | 382 file_name = file_stream.str(); |
| 380 out_file_standalone_.Open(file_name, 48000, "wb"); | 383 out_file_standalone_.Open(file_name, 48000, "wb"); |
| 381 } | 384 } |
| 382 | 385 |
| 383 } // namespace webrtc | 386 } // namespace webrtc |
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