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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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243 int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); | 243 int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); |
244 | 244 |
245 // Set audio packet size, used to determine when it's time to send a DTMF | 245 // Set audio packet size, used to determine when it's time to send a DTMF |
246 // packet in silence (CNG). | 246 // packet in silence (CNG). |
247 int32_t SetAudioPacketSize(uint16_t packet_size_samples); | 247 int32_t SetAudioPacketSize(uint16_t packet_size_samples); |
248 | 248 |
249 // Store the audio level in d_bov for | 249 // Store the audio level in d_bov for |
250 // header-extension-for-audio-level-indication. | 250 // header-extension-for-audio-level-indication. |
251 int32_t SetAudioLevel(uint8_t level_d_bov); | 251 int32_t SetAudioLevel(uint8_t level_d_bov); |
252 | 252 |
253 // Set payload type for Redundant Audio Data RFC 2198. | |
254 int32_t SetRED(int8_t payload_type); | |
255 | |
256 // Get payload type for Redundant Audio Data RFC 2198. | |
257 int32_t RED(int8_t *payload_type) const; | |
258 | |
259 RtpVideoCodecTypes VideoCodecType() const; | 253 RtpVideoCodecTypes VideoCodecType() const; |
260 | 254 |
261 uint32_t MaxConfiguredBitrateVideo() const; | 255 uint32_t MaxConfiguredBitrateVideo() const; |
262 | 256 |
263 // FEC. | 257 // FEC. |
264 void SetGenericFECStatus(bool enable, | 258 void SetGenericFECStatus(bool enable, |
265 uint8_t payload_type_red, | 259 uint8_t payload_type_red, |
266 uint8_t payload_type_fec); | 260 uint8_t payload_type_fec); |
267 | 261 |
268 void GenericFECStatus(bool* enable, | 262 void GenericFECStatus(bool* enable, |
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429 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_); | 423 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_); |
430 | 424 |
431 RateLimiter* const retransmission_rate_limiter_; | 425 RateLimiter* const retransmission_rate_limiter_; |
432 | 426 |
433 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender); | 427 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender); |
434 }; | 428 }; |
435 | 429 |
436 } // namespace webrtc | 430 } // namespace webrtc |
437 | 431 |
438 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ | 432 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ |
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