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

Issue 2719733002: Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. (Closed)
Patch Set: Fixing test. Created 3 years, 10 months 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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269 config.send_codec_spec.cng_payload_type = 42; 269 config.send_codec_spec.cng_payload_type = 42;
270 config.send_codec_spec.cng_plfreq = 56; 270 config.send_codec_spec.cng_plfreq = 56;
271 config.send_codec_spec.min_ptime_ms = 20; 271 config.send_codec_spec.min_ptime_ms = 20;
272 config.send_codec_spec.max_ptime_ms = 60; 272 config.send_codec_spec.max_ptime_ms = 60;
273 config.send_codec_spec.codec_inst = kIsacCodec; 273 config.send_codec_spec.codec_inst = kIsacCodec;
274 config.rtp.extensions.push_back( 274 config.rtp.extensions.push_back(
275 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); 275 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
276 EXPECT_EQ( 276 EXPECT_EQ(
277 "{rtp: {ssrc: 1234, extensions: [{uri: " 277 "{rtp: {ssrc: 1234, extensions: [{uri: "
278 "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: " 278 "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: "
279 "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, " 279 "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: null, "
280 "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, " 280 "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
281 "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " 281 "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
282 "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: " 282 "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
283 "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: " 283 "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: "
284 "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: " 284 "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: "
285 "320, channels: 1, rate: 32000}}}", 285 "320, channels: 1, rate: 32000}}}",
286 config.ToString()); 286 config.ToString());
287 } 287 }
288 288
289 TEST(AudioSendStreamTest, ConstructDestruct) { 289 TEST(AudioSendStreamTest, ConstructDestruct) {
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459 internal::AudioSendStream send_stream( 459 internal::AudioSendStream send_stream(
460 helper.config(), helper.audio_state(), helper.worker_queue(), 460 helper.config(), helper.audio_state(), helper.worker_queue(),
461 helper.packet_router(), helper.congestion_controller(), 461 helper.packet_router(), helper.congestion_controller(),
462 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); 462 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
463 EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000)); 463 EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
464 send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000); 464 send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
465 } 465 }
466 466
467 } // namespace test 467 } // namespace test
468 } // namespace webrtc 468 } // namespace webrtc
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