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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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34 AudioSendStream::Config::Config(Transport* send_transport) | 34 AudioSendStream::Config::Config(Transport* send_transport) |
35 : send_transport(send_transport) {} | 35 : send_transport(send_transport) {} |
36 | 36 |
37 AudioSendStream::Config::~Config() = default; | 37 AudioSendStream::Config::~Config() = default; |
38 | 38 |
39 std::string AudioSendStream::Config::ToString() const { | 39 std::string AudioSendStream::Config::ToString() const { |
40 std::stringstream ss; | 40 std::stringstream ss; |
41 ss << "{rtp: " << rtp.ToString(); | 41 ss << "{rtp: " << rtp.ToString(); |
42 ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr"); | 42 ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr"); |
43 ss << ", voe_channel_id: " << voe_channel_id; | 43 ss << ", voe_channel_id: " << voe_channel_id; |
44 ss << ", min_bitrate_kbps: " << min_bitrate_kbps; | 44 ss << ", min_bitrate_bps: " << min_bitrate_bps; |
45 ss << ", max_bitrate_kbps: " << max_bitrate_kbps; | 45 ss << ", max_bitrate_bps: " << max_bitrate_bps; |
46 ss << ", send_codec_spec: " << send_codec_spec.ToString(); | 46 ss << ", send_codec_spec: " << send_codec_spec.ToString(); |
47 ss << '}'; | 47 ss << '}'; |
48 return ss.str(); | 48 return ss.str(); |
49 } | 49 } |
50 | 50 |
51 AudioSendStream::Config::Rtp::Rtp() = default; | 51 AudioSendStream::Config::Rtp::Rtp() = default; |
52 | 52 |
53 AudioSendStream::Config::Rtp::~Rtp() = default; | 53 AudioSendStream::Config::Rtp::~Rtp() = default; |
54 | 54 |
55 std::string AudioSendStream::Config::Rtp::ToString() const { | 55 std::string AudioSendStream::Config::Rtp::ToString() const { |
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99 enable_opus_dtx == rhs.enable_opus_dtx && | 99 enable_opus_dtx == rhs.enable_opus_dtx && |
100 opus_max_playback_rate == rhs.opus_max_playback_rate && | 100 opus_max_playback_rate == rhs.opus_max_playback_rate && |
101 cng_payload_type == rhs.cng_payload_type && | 101 cng_payload_type == rhs.cng_payload_type && |
102 cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms && | 102 cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms && |
103 min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) { | 103 min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) { |
104 return true; | 104 return true; |
105 } | 105 } |
106 return false; | 106 return false; |
107 } | 107 } |
108 } // namespace webrtc | 108 } // namespace webrtc |
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