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Side by Side Diff: webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc

Issue 1750353002: Change NetEq::GetAudio to use AudioFrame (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 9 months ago
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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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152 sample_rate_hz, num_channels, name); 152 sample_rate_hz, num_channels, name);
153 } 153 }
154 154
155 void AcmReceiveTestOldApi::Run() { 155 void AcmReceiveTestOldApi::Run() {
156 for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet; 156 for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
157 packet.reset(packet_source_->NextPacket())) { 157 packet.reset(packet_source_->NextPacket())) {
158 // Pull audio until time to insert packet. 158 // Pull audio until time to insert packet.
159 while (clock_.TimeInMilliseconds() < packet->time_ms()) { 159 while (clock_.TimeInMilliseconds() < packet->time_ms()) {
160 AudioFrame output_frame; 160 AudioFrame output_frame;
161 EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame)); 161 EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
162 EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_); 162 ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
163 const size_t samples_per_block = 163 const size_t samples_per_block =
164 static_cast<size_t>(output_freq_hz_ * 10 / 1000); 164 static_cast<size_t>(output_freq_hz_ * 10 / 1000);
165 EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_); 165 EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
166 if (exptected_output_channels_ != kArbitraryChannels) { 166 if (exptected_output_channels_ != kArbitraryChannels) {
167 if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) { 167 if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
168 // Don't check number of channels for PLC output, since each test run 168 // Don't check number of channels for PLC output, since each test run
169 // usually starts with a short period of mono PLC before decoding the 169 // usually starts with a short period of mono PLC before decoding the
170 // first packet. 170 // first packet.
171 } else { 171 } else {
172 EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_); 172 EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
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215 if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) { 215 if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
216 output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_) 216 output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
217 ? output_freq_hz_2_ 217 ? output_freq_hz_2_
218 : output_freq_hz_1_; 218 : output_freq_hz_1_;
219 last_toggle_time_ms_ = clock_.TimeInMilliseconds(); 219 last_toggle_time_ms_ = clock_.TimeInMilliseconds();
220 } 220 }
221 } 221 }
222 222
223 } // namespace test 223 } // namespace test
224 } // namespace webrtc 224 } // namespace webrtc
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