Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(343)

Side by Side Diff: webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc

Issue 1421013006: Delete a chain of methods in ViE, VoE and ACM (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month 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
OLDNEW
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
(...skipping 24 matching lines...) Expand all
35 35
36 // Timing files 36 // Timing files
37 DEFINE_string(seq_num, "seq_num", "Sequence number file."); 37 DEFINE_string(seq_num, "seq_num", "Sequence number file.");
38 DEFINE_string(send_ts, "send_timestamp", "Send timestamp file."); 38 DEFINE_string(send_ts, "send_timestamp", "Send timestamp file.");
39 DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file"); 39 DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
40 40
41 // Delay logging 41 // Delay logging
42 DEFINE_string(delay, "", "Log for delay."); 42 DEFINE_string(delay, "", "Log for delay.");
43 43
44 // Other setups 44 // Other setups
45 DEFINE_int32(init_delay, 0, "Initial delay.");
46 DEFINE_bool(verbose, false, "Verbosity."); 45 DEFINE_bool(verbose, false, "Verbosity.");
47 DEFINE_double(loss_rate, 0, "Rate of packet loss < 1"); 46 DEFINE_double(loss_rate, 0, "Rate of packet loss < 1");
48 47
49 const int32_t kAudioPlayedOut = 0x00000001; 48 const int32_t kAudioPlayedOut = 0x00000001;
50 const int32_t kPacketPushedIn = 0x00000001 << 1; 49 const int32_t kPacketPushedIn = 0x00000001 << 1;
51 const int kPlayoutPeriodMs = 10; 50 const int kPlayoutPeriodMs = 10;
52 51
53 namespace webrtc { 52 namespace webrtc {
54 53
55 class InsertPacketWithTiming { 54 class InsertPacketWithTiming {
(...skipping 59 matching lines...) Expand 10 before | Expand all | Expand 10 after
115 << " Hz in " << ((FLAGS_input_stereo) ? "stereo." : "mono.") 114 << " Hz in " << ((FLAGS_input_stereo) ? "stereo." : "mono.")
116 << std::endl; 115 << std::endl;
117 pcm_in_fid_.ReadStereo(FLAGS_input_stereo); 116 pcm_in_fid_.ReadStereo(FLAGS_input_stereo);
118 } 117 }
119 118
120 ASSERT_TRUE(pcm_out_fid_ != NULL); 119 ASSERT_TRUE(pcm_out_fid_ != NULL);
121 std::cout << "Output file " << FLAGS_output << " at " << FLAGS_output_fs_hz 120 std::cout << "Output file " << FLAGS_output << " at " << FLAGS_output_fs_hz
122 << " Hz." << std::endl; 121 << " Hz." << std::endl;
123 122
124 // Other setups 123 // Other setups
125 if (FLAGS_init_delay > 0)
126 EXPECT_EQ(0, receive_acm_->SetInitialPlayoutDelay(FLAGS_init_delay));
127
128 if (FLAGS_loss_rate > 0) 124 if (FLAGS_loss_rate > 0)
129 loss_threshold_ = RAND_MAX * FLAGS_loss_rate; 125 loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
130 else 126 else
131 loss_threshold_ = 0; 127 loss_threshold_ = 0;
132 } 128 }
133 129
134 void TickOneMillisecond(uint32_t* action) { 130 void TickOneMillisecond(uint32_t* action) {
135 // One millisecond passed. 131 // One millisecond passed.
136 time_to_insert_packet_ms_--; 132 time_to_insert_packet_ms_--;
137 time_to_playout_audio_ms_--; 133 time_to_playout_audio_ms_--;
(...skipping 164 matching lines...) Expand 10 before | Expand all | Expand 10 after
302 if (delay_log != NULL) { 298 if (delay_log != NULL) {
303 fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms); 299 fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
304 } 300 }
305 } 301 }
306 } 302 }
307 std::cout << std::endl; 303 std::cout << std::endl;
308 test.TearDown(); 304 test.TearDown();
309 if (delay_log != NULL) 305 if (delay_log != NULL)
310 fclose(delay_log); 306 fclose(delay_log);
311 } 307 }
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc ('k') | webrtc/modules/modules.gyp » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698