Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(314)

Side by Side Diff: webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc

Issue 1782803002: Fix NetEq performance test regression (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « no previous file | webrtc/modules/include/module_common_types.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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
(...skipping 57 matching lines...) Expand 10 before | Expand all | Expand 10 after
68 if (input_samples.empty()) 68 if (input_samples.empty())
69 exit(1); 69 exit(1);
70 uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)]; 70 uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
71 size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(), 71 size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
72 input_samples.size(), input_payload); 72 input_samples.size(), input_payload);
73 RTC_CHECK_EQ(sizeof(input_payload), payload_len); 73 RTC_CHECK_EQ(sizeof(input_payload), payload_len);
74 74
75 // Main loop. 75 // Main loop.
76 webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock(); 76 webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
77 int64_t start_time_ms = clock->TimeInMilliseconds(); 77 int64_t start_time_ms = clock->TimeInMilliseconds();
78 AudioFrame out_frame;
78 while (time_now_ms < runtime_ms) { 79 while (time_now_ms < runtime_ms) {
79 while (packet_input_time_ms <= time_now_ms) { 80 while (packet_input_time_ms <= time_now_ms) {
80 // Drop every N packets, where N = FLAGS_lossrate. 81 // Drop every N packets, where N = FLAGS_lossrate.
81 bool lost = false; 82 bool lost = false;
82 if (lossrate > 0) { 83 if (lossrate > 0) {
83 lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0; 84 lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
84 } 85 }
85 if (!lost) { 86 if (!lost) {
86 // Insert packet. 87 // Insert packet.
87 int error = 88 int error =
88 neteq->InsertPacket(rtp_header, input_payload, 89 neteq->InsertPacket(rtp_header, input_payload,
89 packet_input_time_ms * kSampRateHz / 1000); 90 packet_input_time_ms * kSampRateHz / 1000);
90 if (error != NetEq::kOK) 91 if (error != NetEq::kOK)
91 return -1; 92 return -1;
92 } 93 }
93 94
94 // Get next packet. 95 // Get next packet.
95 packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType, 96 packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
96 kInputBlockSizeSamples, 97 kInputBlockSizeSamples,
97 &rtp_header); 98 &rtp_header);
98 input_samples = audio_loop.GetNextBlock(); 99 input_samples = audio_loop.GetNextBlock();
99 if (input_samples.empty()) 100 if (input_samples.empty())
100 return -1; 101 return -1;
101 payload_len = WebRtcPcm16b_Encode(input_samples.data(), 102 payload_len = WebRtcPcm16b_Encode(input_samples.data(),
102 input_samples.size(), input_payload); 103 input_samples.size(), input_payload);
103 assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t)); 104 assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
104 } 105 }
105 106
106 // Get output audio, but don't do anything with it. 107 // Get output audio, but don't do anything with it.
107 AudioFrame out_frame;
108 int error = neteq->GetAudio(&out_frame); 108 int error = neteq->GetAudio(&out_frame);
109 if (error != NetEq::kOK) 109 if (error != NetEq::kOK)
110 return -1; 110 return -1;
111 111
112 assert(out_frame.samples_per_channel_ == 112 assert(out_frame.samples_per_channel_ ==
113 static_cast<size_t>(kSampRateHz * 10 / 1000)); 113 static_cast<size_t>(kSampRateHz * 10 / 1000));
114 114
115 static const int kOutputBlockSizeMs = 10; 115 static const int kOutputBlockSizeMs = 10;
116 time_now_ms += kOutputBlockSizeMs; 116 time_now_ms += kOutputBlockSizeMs;
117 if (time_now_ms >= runtime_ms / 2 && !drift_flipped) { 117 if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
118 // Apply negative drift second half of simulation. 118 // Apply negative drift second half of simulation.
119 rtp_gen.set_drift_factor(-drift_factor); 119 rtp_gen.set_drift_factor(-drift_factor);
120 drift_flipped = true; 120 drift_flipped = true;
121 } 121 }
122 } 122 }
123 int64_t end_time_ms = clock->TimeInMilliseconds(); 123 int64_t end_time_ms = clock->TimeInMilliseconds();
124 delete neteq; 124 delete neteq;
125 return end_time_ms - start_time_ms; 125 return end_time_ms - start_time_ms;
126 } 126 }
127 127
128 } // namespace test 128 } // namespace test
129 } // namespace webrtc 129 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | webrtc/modules/include/module_common_types.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698