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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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1448 | 1448 |
1449 void RecreateAudioSendStream() { | 1449 void RecreateAudioSendStream() { |
1450 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 1450 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
1451 if (stream_) { | 1451 if (stream_) { |
1452 call_->DestroyAudioSendStream(stream_); | 1452 call_->DestroyAudioSendStream(stream_); |
1453 stream_ = nullptr; | 1453 stream_ = nullptr; |
1454 } | 1454 } |
1455 RTC_DCHECK(!stream_); | 1455 RTC_DCHECK(!stream_); |
1456 if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == | 1456 if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == |
1457 "Enabled") { | 1457 "Enabled") { |
1458 config_.min_bitrate_bps = kOpusMinBitrateBps; | |
1459 config_.max_bitrate_bps = kOpusBitrateFbBps; | |
1458 // TODO(mflodman): Keep testing this and set proper values. | 1460 // TODO(mflodman): Keep testing this and set proper values. |
1459 // Note: This is an early experiment currently only supported by Opus. | 1461 // Note: This is an early experiment currently only supported by Opus. |
1460 if (webrtc::field_trial::FindFullName( | 1462 if (webrtc::field_trial::FindFullName( |
1461 "WebRTC-SendSideBwe-WithOverhead") == "Enabled") { | 1463 "WebRTC-SendSideBwe-WithOverhead") == "Enabled") { |
1462 auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs( | 1464 auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs( |
1463 config_.send_codec_spec.codec_inst); | 1465 config_.send_codec_spec.codec_inst); |
1464 if (!packet_sizes_ms.empty()) { | 1466 if (!packet_sizes_ms.empty()) { |
1465 int max_packet_size_ms = | 1467 int max_packet_size_ms = |
1466 *std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end()); | 1468 *std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end()); |
1467 int min_packet_size_ms = | 1469 int min_packet_size_ms = |
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1480 | 1482 |
1481 int min_overhead_bps = | 1483 int min_overhead_bps = |
1482 kOverheadPerPacket * 8 * 1000 / max_packet_size_ms; | 1484 kOverheadPerPacket * 8 * 1000 / max_packet_size_ms; |
1483 | 1485 |
1484 int max_overhead_bps = | 1486 int max_overhead_bps = |
1485 kOverheadPerPacket * 8 * 1000 / min_packet_size_ms; | 1487 kOverheadPerPacket * 8 * 1000 / min_packet_size_ms; |
1486 | 1488 |
1487 config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps; | 1489 config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps; |
1488 config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps; | 1490 config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps; |
1489 } | 1491 } |
1490 } else { | |
1491 config_.min_bitrate_bps = kOpusMinBitrateBps; | |
minyue-webrtc
2017/01/24 14:22:00
Then it should be outside if(webrtc::field_trial::
stefan-webrtc
2017/01/24 14:48:03
I don't follow, I moved it from the else branch to
| |
1492 config_.max_bitrate_bps = kOpusBitrateFbBps; | |
1493 } | 1492 } |
1494 } | 1493 } |
1495 stream_ = call_->CreateAudioSendStream(config_); | 1494 stream_ = call_->CreateAudioSendStream(config_); |
1496 RTC_CHECK(stream_); | 1495 RTC_CHECK(stream_); |
1497 UpdateSendState(); | 1496 UpdateSendState(); |
1498 } | 1497 } |
1499 | 1498 |
1500 rtc::ThreadChecker worker_thread_checker_; | 1499 rtc::ThreadChecker worker_thread_checker_; |
1501 rtc::RaceChecker audio_capture_race_checker_; | 1500 rtc::RaceChecker audio_capture_race_checker_; |
1502 webrtc::AudioTransport* const voe_audio_transport_ = nullptr; | 1501 webrtc::AudioTransport* const voe_audio_transport_ = nullptr; |
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2705 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 2704 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
2706 const auto it = send_streams_.find(ssrc); | 2705 const auto it = send_streams_.find(ssrc); |
2707 if (it != send_streams_.end()) { | 2706 if (it != send_streams_.end()) { |
2708 return it->second->channel(); | 2707 return it->second->channel(); |
2709 } | 2708 } |
2710 return -1; | 2709 return -1; |
2711 } | 2710 } |
2712 } // namespace cricket | 2711 } // namespace cricket |
2713 | 2712 |
2714 #endif // HAVE_WEBRTC_VOICE | 2713 #endif // HAVE_WEBRTC_VOICE |
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