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Side by Side Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 2247213005: Fixing config for Audio BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: on Stefan's suggestion Created 4 years, 1 month ago
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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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208 // and deleted before any other members. 208 // and deleted before any other members.
209 rtc::TaskQueue worker_queue_; 209 rtc::TaskQueue worker_queue_;
210 }; 210 };
211 } // namespace 211 } // namespace
212 212
213 TEST(AudioSendStreamTest, ConfigToString) { 213 TEST(AudioSendStreamTest, ConfigToString) {
214 AudioSendStream::Config config(nullptr); 214 AudioSendStream::Config config(nullptr);
215 config.rtp.ssrc = kSsrc; 215 config.rtp.ssrc = kSsrc;
216 config.rtp.c_name = kCName; 216 config.rtp.c_name = kCName;
217 config.voe_channel_id = kChannelId; 217 config.voe_channel_id = kChannelId;
218 config.min_bitrate_kbps = 12; 218 config.min_bitrate_bps = 12000;
219 config.max_bitrate_kbps = 34; 219 config.max_bitrate_bps = 34000;
220 config.send_codec_spec.nack_enabled = true; 220 config.send_codec_spec.nack_enabled = true;
221 config.send_codec_spec.transport_cc_enabled = false; 221 config.send_codec_spec.transport_cc_enabled = false;
222 config.send_codec_spec.enable_codec_fec = true; 222 config.send_codec_spec.enable_codec_fec = true;
223 config.send_codec_spec.enable_opus_dtx = false; 223 config.send_codec_spec.enable_opus_dtx = false;
224 config.send_codec_spec.opus_max_playback_rate = 32000; 224 config.send_codec_spec.opus_max_playback_rate = 32000;
225 config.send_codec_spec.cng_payload_type = 42; 225 config.send_codec_spec.cng_payload_type = 42;
226 config.send_codec_spec.cng_plfreq = 56; 226 config.send_codec_spec.cng_plfreq = 56;
227 config.send_codec_spec.min_ptime_ms = 20; 227 config.send_codec_spec.min_ptime_ms = 20;
228 config.send_codec_spec.max_ptime_ms = 60; 228 config.send_codec_spec.max_ptime_ms = 60;
229 config.send_codec_spec.codec_inst = kIsacCodec; 229 config.send_codec_spec.codec_inst = kIsacCodec;
230 config.rtp.extensions.push_back( 230 config.rtp.extensions.push_back(
231 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); 231 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
232 EXPECT_EQ( 232 EXPECT_EQ(
233 "{rtp: {ssrc: 1234, extensions: [{uri: " 233 "{rtp: {ssrc: 1234, extensions: [{uri: "
234 "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: " 234 "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: "
235 "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, " 235 "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
236 "voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, " 236 "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
237 "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " 237 "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
238 "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: " 238 "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
239 "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: " 239 "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: "
240 "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: " 240 "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: "
241 "320, channels: 1, rate: 32000}}}", 241 "320, channels: 1, rate: 32000}}}",
242 config.ToString()); 242 config.ToString());
243 } 243 }
244 244
245 TEST(AudioSendStreamTest, ConstructDestruct) { 245 TEST(AudioSendStreamTest, ConstructDestruct) {
246 ConfigHelper helper; 246 ConfigHelper helper;
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376 EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _)) 376 EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
377 .WillOnce(Return(0)); 377 .WillOnce(Return(0));
378 internal::AudioSendStream send_stream( 378 internal::AudioSendStream send_stream(
379 stream_config, helper.audio_state(), helper.worker_queue(), 379 stream_config, helper.audio_state(), helper.worker_queue(),
380 helper.congestion_controller(), helper.bitrate_allocator(), 380 helper.congestion_controller(), helper.bitrate_allocator(),
381 helper.event_log()); 381 helper.event_log());
382 } 382 }
383 383
384 } // namespace test 384 } // namespace test
385 } // namespace webrtc 385 } // namespace webrtc
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