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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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| 197 "curr buff-lev = %4u opt buff-lev = %4u \n", | 197 "curr buff-lev = %4u opt buff-lev = %4u \n", |
| 198 statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs, | 198 statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs, |
| 199 statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs, | 199 statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs, |
| 200 average_delay, statistics.currentBufferSize, | 200 average_delay, statistics.currentBufferSize, |
| 201 statistics.preferredBufferSize); | 201 statistics.preferredBufferSize); |
| 202 fflush (stdout); | 202 fflush (stdout); |
| 203 } | 203 } |
| 204 | 204 |
| 205 in_file_a_.Read10MsData(audio_frame); | 205 in_file_a_.Read10MsData(audio_frame); |
| 206 ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0); | 206 ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0); |
| 207 ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame)); | 207 bool muted; |
| 208 ASSERT_EQ(0, |
| 209 acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted)); |
| 210 RTC_DCHECK(!muted); |
| 208 out_file_b_.Write10MsData( | 211 out_file_b_.Write10MsData( |
| 209 audio_frame.data_, | 212 audio_frame.data_, |
| 210 audio_frame.samples_per_channel_ * audio_frame.num_channels_); | 213 audio_frame.samples_per_channel_ * audio_frame.num_channels_); |
| 211 received_ts = channel_a2b_->LastInTimestamp(); | 214 received_ts = channel_a2b_->LastInTimestamp(); |
| 212 rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp(); | 215 rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp(); |
| 213 ASSERT_TRUE(playout_timestamp); | 216 ASSERT_TRUE(playout_timestamp); |
| 214 inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) / | 217 inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) / |
| 215 static_cast<double>(encoding_sample_rate_hz_); | 218 static_cast<double>(encoding_sample_rate_hz_); |
| 216 | 219 |
| 217 if (num_frames > 10) | 220 if (num_frames > 10) |
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| 256 test_setting.codec.num_channels = FLAGS_num_channels; | 259 test_setting.codec.num_channels = FLAGS_num_channels; |
| 257 test_setting.acm.dtx = FLAGS_dtx; | 260 test_setting.acm.dtx = FLAGS_dtx; |
| 258 test_setting.acm.fec = FLAGS_fec; | 261 test_setting.acm.fec = FLAGS_fec; |
| 259 test_setting.packet_loss = FLAGS_packet_loss; | 262 test_setting.packet_loss = FLAGS_packet_loss; |
| 260 | 263 |
| 261 webrtc::DelayTest delay_test; | 264 webrtc::DelayTest delay_test; |
| 262 delay_test.Initialize(); | 265 delay_test.Initialize(); |
| 263 delay_test.Perform(&test_setting, 1, 240, "delay_test"); | 266 delay_test.Perform(&test_setting, 1, 240, "delay_test"); |
| 264 return 0; | 267 return 0; |
| 265 } | 268 } |
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