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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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47 int32_t SetAudioPacketSize(uint16_t packet_size_samples); | 47 int32_t SetAudioPacketSize(uint16_t packet_size_samples); |
48 | 48 |
49 // Store the audio level in dBov for | 49 // Store the audio level in dBov for |
50 // header-extension-for-audio-level-indication. | 50 // header-extension-for-audio-level-indication. |
51 // Valid range is [0,100]. Actual value is negative. | 51 // Valid range is [0,100]. Actual value is negative. |
52 int32_t SetAudioLevel(uint8_t level_dbov); | 52 int32_t SetAudioLevel(uint8_t level_dbov); |
53 | 53 |
54 // Send a DTMF tone using RFC 2833 (4733) | 54 // Send a DTMF tone using RFC 2833 (4733) |
55 int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); | 55 int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); |
56 | 56 |
57 int AudioFrequency() const; | |
58 | |
59 protected: | 57 protected: |
60 bool SendTelephoneEventPacket( | 58 bool SendTelephoneEventPacket( |
61 bool ended, | 59 bool ended, |
62 int8_t dtmf_payload_type, | 60 int8_t dtmf_payload_type, |
63 uint32_t dtmf_timestamp, | 61 uint32_t dtmf_timestamp, |
64 uint16_t duration, | 62 uint16_t duration, |
65 bool marker_bit); // set on first packet in talk burst | 63 bool marker_bit); // set on first packet in talk burst |
66 | 64 |
67 bool MarkerBit(FrameType frame_type, int8_t payload_type); | 65 bool MarkerBit(FrameType frame_type, int8_t payload_type); |
68 | 66 |
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98 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) | 96 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) |
99 uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_); | 97 uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_); |
100 OneTimeEvent first_packet_sent_; | 98 OneTimeEvent first_packet_sent_; |
101 | 99 |
102 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio); | 100 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio); |
103 }; | 101 }; |
104 | 102 |
105 } // namespace webrtc | 103 } // namespace webrtc |
106 | 104 |
107 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ | 105 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ |
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