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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 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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1129 } | 1129 } |
1130 | 1130 |
1131 void RecreateAudioSendStream( | 1131 void RecreateAudioSendStream( |
1132 const std::vector<webrtc::RtpExtension>& extensions) { | 1132 const std::vector<webrtc::RtpExtension>& extensions) { |
1133 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 1133 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
1134 if (stream_) { | 1134 if (stream_) { |
1135 call_->DestroyAudioSendStream(stream_); | 1135 call_->DestroyAudioSendStream(stream_); |
1136 stream_ = nullptr; | 1136 stream_ = nullptr; |
1137 } | 1137 } |
1138 config_.rtp.extensions = extensions; | 1138 config_.rtp.extensions = extensions; |
1139 if (webrtc::field_trial::FindFullName("WebRTC-AdaptAudioBitrate") == | |
1140 "Enabled") { | |
1141 // TODO(mflodman):Set this properly. | |
1142 config_.min_bitrate_kbps = kOpusMinBitrate; | |
1143 config_.max_bitrate_kbps = kOpusBitrateFb; | |
stefan-webrtc
2016/07/20 10:15:00
kOpusBitrateFb seems like a bad variable name. Cou
minyue-webrtc
2016/07/21 10:59:19
is it possible to use kOpusMaxBitrate (although we
mflodman
2016/07/22 13:50:29
@minyue:
I didn't want to increase the bitrate mor
mflodman
2016/07/22 13:50:29
@stefan:
There are three different Opus bitrate c
stefan-webrtc
2016/07/25 15:06:20
I see. Probably good enough then, although I would
mflodman
2016/07/26 09:50:52
I agree and I'll bring that up for discussion.
| |
1144 } | |
1145 | |
1139 RTC_DCHECK(!stream_); | 1146 RTC_DCHECK(!stream_); |
1140 stream_ = call_->CreateAudioSendStream(config_); | 1147 stream_ = call_->CreateAudioSendStream(config_); |
1141 RTC_CHECK(stream_); | 1148 RTC_CHECK(stream_); |
1142 UpdateSendState(); | 1149 UpdateSendState(); |
1143 } | 1150 } |
1144 | 1151 |
1145 bool SendTelephoneEvent(int payload_type, int event, int duration_ms) { | 1152 bool SendTelephoneEvent(int payload_type, int event, int duration_ms) { |
1146 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 1153 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
1147 RTC_DCHECK(stream_); | 1154 RTC_DCHECK(stream_); |
1148 return stream_->SendTelephoneEvent(payload_type, event, duration_ms); | 1155 return stream_->SendTelephoneEvent(payload_type, event, duration_ms); |
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2616 } | 2623 } |
2617 } else { | 2624 } else { |
2618 LOG(LS_INFO) << "Stopping playout for channel #" << channel; | 2625 LOG(LS_INFO) << "Stopping playout for channel #" << channel; |
2619 engine()->voe()->base()->StopPlayout(channel); | 2626 engine()->voe()->base()->StopPlayout(channel); |
2620 } | 2627 } |
2621 return true; | 2628 return true; |
2622 } | 2629 } |
2623 } // namespace cricket | 2630 } // namespace cricket |
2624 | 2631 |
2625 #endif // HAVE_WEBRTC_VOICE | 2632 #endif // HAVE_WEBRTC_VOICE |
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