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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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| 174 output_config_.set_sample_rate_hz(rate_hz); | 174 output_config_.set_sample_rate_hz(rate_hz); |
| 175 MaybeResetBuffer(&output_, output_config_); | 175 MaybeResetBuffer(&output_, output_config_); |
| 176 } | 176 } |
| 177 | 177 |
| 178 void DebugDumpGenerator::SetOutputChannels(int channels) { | 178 void DebugDumpGenerator::SetOutputChannels(int channels) { |
| 179 output_config_.set_num_channels(channels); | 179 output_config_.set_num_channels(channels); |
| 180 MaybeResetBuffer(&output_, output_config_); | 180 MaybeResetBuffer(&output_, output_config_); |
| 181 } | 181 } |
| 182 | 182 |
| 183 void DebugDumpGenerator::StartRecording() { | 183 void DebugDumpGenerator::StartRecording() { |
| 184 apm_->StartDebugRecording(dump_file_name_.c_str(), -1); | 184 apm_->StartDebugRecording(dump_file_name_.c_str()); |
| 185 } | 185 } |
| 186 | 186 |
| 187 void DebugDumpGenerator::Process(size_t num_blocks) { | 187 void DebugDumpGenerator::Process(size_t num_blocks) { |
| 188 for (size_t i = 0; i < num_blocks; ++i) { | 188 for (size_t i = 0; i < num_blocks; ++i) { |
| 189 ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_, | 189 ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_, |
| 190 reverse_config_, reverse_->channels()); | 190 reverse_config_, reverse_->channels()); |
| 191 ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_, | 191 ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_, |
| 192 input_->channels()); | 192 input_->channels()); |
| 193 RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100)); | 193 RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100)); |
| 194 apm_->set_stream_key_pressed(i % 10 == 9); | 194 apm_->set_stream_key_pressed(i % 10 == 9); |
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| 600 config.Set<ExperimentalNs>(new ExperimentalNs(true)); | 600 config.Set<ExperimentalNs>(new ExperimentalNs(true)); |
| 601 DebugDumpGenerator generator(config); | 601 DebugDumpGenerator generator(config); |
| 602 generator.StartRecording(); | 602 generator.StartRecording(); |
| 603 generator.Process(100); | 603 generator.Process(100); |
| 604 generator.StopRecording(); | 604 generator.StopRecording(); |
| 605 VerifyDebugDump(generator.dump_file_name()); | 605 VerifyDebugDump(generator.dump_file_name()); |
| 606 } | 606 } |
| 607 | 607 |
| 608 } // namespace test | 608 } // namespace test |
| 609 } // namespace webrtc | 609 } // namespace webrtc |
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