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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 |
11 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h" | 11 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h" |
12 | 12 |
| 13 #include "webrtc/base/checks.h" |
13 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" | 14 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" |
14 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" | 15 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" |
15 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" | 16 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" |
16 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" | 17 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" |
17 #include "webrtc/modules/include/module_common_types.h" | 18 #include "webrtc/modules/include/module_common_types.h" |
18 #include "webrtc/system_wrappers/include/clock.h" | 19 #include "webrtc/system_wrappers/include/clock.h" |
19 #include "webrtc/test/testsupport/fileutils.h" | 20 #include "webrtc/test/testsupport/fileutils.h" |
20 #include "webrtc/typedefs.h" | 21 #include "webrtc/typedefs.h" |
21 | 22 |
22 using webrtc::NetEq; | 23 using webrtc::NetEq; |
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98 &rtp_header); | 99 &rtp_header); |
99 input_samples = audio_loop.GetNextBlock(); | 100 input_samples = audio_loop.GetNextBlock(); |
100 if (input_samples.empty()) | 101 if (input_samples.empty()) |
101 return -1; | 102 return -1; |
102 payload_len = WebRtcPcm16b_Encode(input_samples.data(), | 103 payload_len = WebRtcPcm16b_Encode(input_samples.data(), |
103 input_samples.size(), input_payload); | 104 input_samples.size(), input_payload); |
104 assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t)); | 105 assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t)); |
105 } | 106 } |
106 | 107 |
107 // Get output audio, but don't do anything with it. | 108 // Get output audio, but don't do anything with it. |
108 int error = neteq->GetAudio(&out_frame); | 109 bool muted; |
| 110 int error = neteq->GetAudio(&out_frame, &muted); |
| 111 RTC_CHECK(!muted); |
109 if (error != NetEq::kOK) | 112 if (error != NetEq::kOK) |
110 return -1; | 113 return -1; |
111 | 114 |
112 assert(out_frame.samples_per_channel_ == | 115 assert(out_frame.samples_per_channel_ == |
113 static_cast<size_t>(kSampRateHz * 10 / 1000)); | 116 static_cast<size_t>(kSampRateHz * 10 / 1000)); |
114 | 117 |
115 static const int kOutputBlockSizeMs = 10; | 118 static const int kOutputBlockSizeMs = 10; |
116 time_now_ms += kOutputBlockSizeMs; | 119 time_now_ms += kOutputBlockSizeMs; |
117 if (time_now_ms >= runtime_ms / 2 && !drift_flipped) { | 120 if (time_now_ms >= runtime_ms / 2 && !drift_flipped) { |
118 // Apply negative drift second half of simulation. | 121 // Apply negative drift second half of simulation. |
119 rtp_gen.set_drift_factor(-drift_factor); | 122 rtp_gen.set_drift_factor(-drift_factor); |
120 drift_flipped = true; | 123 drift_flipped = true; |
121 } | 124 } |
122 } | 125 } |
123 int64_t end_time_ms = clock->TimeInMilliseconds(); | 126 int64_t end_time_ms = clock->TimeInMilliseconds(); |
124 delete neteq; | 127 delete neteq; |
125 return end_time_ms - start_time_ms; | 128 return end_time_ms - start_time_ms; |
126 } | 129 } |
127 | 130 |
128 } // namespace test | 131 } // namespace test |
129 } // namespace webrtc | 132 } // namespace webrtc |
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