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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include <stdint.h> | 5 #include <stdint.h> |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/bind_helpers.h" | 8 #include "base/bind_helpers.h" |
9 #include "base/memory/scoped_ptr.h" | 9 #include "base/memory/scoped_ptr.h" |
10 #include "base/test/simple_test_tick_clock.h" | 10 #include "base/test/simple_test_tick_clock.h" |
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62 task_runner_, | 62 task_runner_, |
63 task_runner_, | 63 task_runner_, |
64 task_runner_); | 64 task_runner_); |
65 audio_config_.codec = transport::kOpus; | 65 audio_config_.codec = transport::kOpus; |
66 audio_config_.use_external_encoder = false; | 66 audio_config_.use_external_encoder = false; |
67 audio_config_.frequency = kDefaultAudioSamplingRate; | 67 audio_config_.frequency = kDefaultAudioSamplingRate; |
68 audio_config_.channels = 2; | 68 audio_config_.channels = 2; |
69 audio_config_.bitrate = kDefaultAudioEncoderBitrate; | 69 audio_config_.bitrate = kDefaultAudioEncoderBitrate; |
70 audio_config_.rtp_config.payload_type = 127; | 70 audio_config_.rtp_config.payload_type = 127; |
71 | 71 |
| 72 transport::CastTransportAudioConfig transport_config; |
| 73 transport_config.base.rtp_config.payload_type = 127; |
| 74 transport_config.channels = 2; |
72 net::IPEndPoint dummy_endpoint; | 75 net::IPEndPoint dummy_endpoint; |
73 | 76 |
74 transport_sender_.reset(new transport::CastTransportSenderImpl( | 77 transport_sender_.reset(new transport::CastTransportSenderImpl( |
75 NULL, | 78 NULL, |
76 testing_clock_, | 79 testing_clock_, |
77 dummy_endpoint, | 80 dummy_endpoint, |
78 base::Bind(&UpdateCastTransportStatus), | 81 base::Bind(&UpdateCastTransportStatus), |
79 transport::BulkRawEventsCallback(), | 82 transport::BulkRawEventsCallback(), |
80 base::TimeDelta(), | 83 base::TimeDelta(), |
81 task_runner_, | 84 task_runner_, |
82 &transport_)); | 85 &transport_)); |
| 86 transport_sender_->InitializeAudio(transport_config); |
83 audio_sender_.reset(new AudioSender( | 87 audio_sender_.reset(new AudioSender( |
84 cast_environment_, audio_config_, transport_sender_.get())); | 88 cast_environment_, audio_config_, transport_sender_.get())); |
85 task_runner_->RunTasks(); | 89 task_runner_->RunTasks(); |
86 } | 90 } |
87 | 91 |
88 virtual ~AudioSenderTest() {} | 92 virtual ~AudioSenderTest() {} |
89 | 93 |
90 static void UpdateCastTransportStatus(transport::CastTransportStatus status) { | 94 static void UpdateCastTransportStatus(transport::CastTransportStatus status) { |
91 EXPECT_EQ(status, transport::TRANSPORT_AUDIO_INITIALIZED); | 95 EXPECT_EQ(status, transport::TRANSPORT_AUDIO_INITIALIZED); |
92 } | 96 } |
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132 base::TimeDelta max_rtcp_timeout = | 136 base::TimeDelta max_rtcp_timeout = |
133 base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2); | 137 base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2); |
134 testing_clock_->Advance(max_rtcp_timeout); | 138 testing_clock_->Advance(max_rtcp_timeout); |
135 task_runner_->RunTasks(); | 139 task_runner_->RunTasks(); |
136 EXPECT_GE(transport_.number_of_rtp_packets(), 1); | 140 EXPECT_GE(transport_.number_of_rtp_packets(), 1); |
137 EXPECT_EQ(transport_.number_of_rtcp_packets(), 1); | 141 EXPECT_EQ(transport_.number_of_rtcp_packets(), 1); |
138 } | 142 } |
139 | 143 |
140 } // namespace cast | 144 } // namespace cast |
141 } // namespace media | 145 } // namespace media |
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