ffbbcac4c6
We support multiple payload types, and one which matches the audio codec the closest, is picked (or the one with lowest clock rate, if no perfect match is found). The exact clock rate is then ignored and DTMF packets are time stamped with the rate of the current audio codec. This is exactly the way the code has worked up to this point, but until now we have been under the impression that we were in fact sending 8k DTMF. In other words, this is an improvement over the current situation, since we will most likely find a payload type which matches the codec clock rate. This CL also does a little cleaning of the DTMFQueue and RTPSenderAudio classes. BUG=webrtc:2795 Review-Url: https://codereview.webrtc.org/2392883002 Cr-Commit-Position: refs/heads/master@{#15129}
42 lines
1.0 KiB
C++
42 lines
1.0 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
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#include <list>
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#include "webrtc/base/criticalsection.h"
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namespace webrtc {
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class DtmfQueue {
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public:
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struct Event {
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uint16_t duration_ms = 0;
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uint8_t payload_type = 0;
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uint8_t key = 0;
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uint8_t level = 0;
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};
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DtmfQueue();
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~DtmfQueue();
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bool AddDtmf(const Event& event);
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bool NextDtmf(Event* event);
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bool PendingDtmf() const;
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private:
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rtc::CriticalSection dtmf_critsect_;
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std::list<Event> queue_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
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