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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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212 }; | 212 }; |
213 } // namespace | 213 } // namespace |
214 | 214 |
215 TEST(AudioSendStreamTest, ConfigToString) { | 215 TEST(AudioSendStreamTest, ConfigToString) { |
216 AudioSendStream::Config config(nullptr); | 216 AudioSendStream::Config config(nullptr); |
217 config.rtp.ssrc = kSsrc; | 217 config.rtp.ssrc = kSsrc; |
218 config.rtp.extensions.push_back( | 218 config.rtp.extensions.push_back( |
219 RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); | 219 RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); |
220 config.rtp.c_name = kCName; | 220 config.rtp.c_name = kCName; |
221 config.voe_channel_id = kChannelId; | 221 config.voe_channel_id = kChannelId; |
222 config.min_bitrate_kbps = 12; | |
223 config.max_bitrate_kbps = 34; | |
224 config.send_codec_spec.nack_enabled = true; | |
225 config.send_codec_spec.transport_cc_enabled = false; | |
226 config.send_codec_spec.enable_codec_fec = true; | |
227 config.send_codec_spec.enable_opus_dtx = false; | |
228 config.send_codec_spec.opus_max_playback_rate = 32000; | |
229 config.send_codec_spec.cng_payload_type = 42; | 222 config.send_codec_spec.cng_payload_type = 42; |
230 config.send_codec_spec.cng_plfreq = 56; | |
231 config.send_codec_spec.codec_inst = kIsacCodec; | |
232 EXPECT_EQ( | 223 EXPECT_EQ( |
233 "{rtp: {ssrc: 1234, extensions: [{uri: " | 224 "{rtp: {ssrc: 1234, extensions: [{uri: " |
234 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], " | 225 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], " |
235 "nack: {rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, " | 226 "nack: {rtp_history_ms: 0}, c_name: foo_name}, voe_channel_id: 1, " |
236 "voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, " | 227 "cng_payload_type: 42}", |
237 "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " | |
238 "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: " | |
239 "32000, cng_payload_type: 42, cng_plfreq: 56, codec_inst: {pltype: " | |
240 "103, plname: \"isac\", plfreq: 16000, pacsize: 320, channels: 1, rate: " | |
241 "32000}}}", | |
242 config.ToString()); | 228 config.ToString()); |
243 } | 229 } |
244 | 230 |
245 TEST(AudioSendStreamTest, ConstructDestruct) { | 231 TEST(AudioSendStreamTest, ConstructDestruct) { |
246 ConfigHelper helper; | 232 ConfigHelper helper; |
247 internal::AudioSendStream send_stream( | 233 internal::AudioSendStream send_stream( |
248 helper.config(), helper.audio_state(), helper.worker_queue(), | 234 helper.config(), helper.audio_state(), helper.worker_queue(), |
249 helper.congestion_controller(), helper.bitrate_allocator(), | 235 helper.congestion_controller(), helper.bitrate_allocator(), |
250 helper.event_log()); | 236 helper.event_log()); |
251 } | 237 } |
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365 EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _)) | 351 EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _)) |
366 .WillOnce(Return(0)); | 352 .WillOnce(Return(0)); |
367 internal::AudioSendStream send_stream( | 353 internal::AudioSendStream send_stream( |
368 stream_config, helper.audio_state(), helper.worker_queue(), | 354 stream_config, helper.audio_state(), helper.worker_queue(), |
369 helper.congestion_controller(), helper.bitrate_allocator(), | 355 helper.congestion_controller(), helper.bitrate_allocator(), |
370 helper.event_log()); | 356 helper.event_log()); |
371 } | 357 } |
372 | 358 |
373 } // namespace test | 359 } // namespace test |
374 } // namespace webrtc | 360 } // namespace webrtc |
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