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Side by Side Diff: webrtc/voice_engine/channel.h

Issue 1803923003: Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 9 months ago
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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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149 rtc::CriticalSection lock_; 149 rtc::CriticalSection lock_;
150 State state_; 150 State state_;
151 }; 151 };
152 152
153 class Channel 153 class Channel
154 : public RtpData, 154 : public RtpData,
155 public RtpFeedback, 155 public RtpFeedback,
156 public FileCallback, // receiving notification from file player & 156 public FileCallback, // receiving notification from file player &
157 // recorder 157 // recorder
158 public Transport, 158 public Transport,
159 public RtpAudioFeedback,
160 public AudioPacketizationCallback, // receive encoded packets from the 159 public AudioPacketizationCallback, // receive encoded packets from the
161 // ACM 160 // ACM
162 public ACMVADCallback, // receive voice activity from the ACM 161 public ACMVADCallback, // receive voice activity from the ACM
163 public MixerParticipant // supplies output mixer with audio frames 162 public MixerParticipant // supplies output mixer with audio frames
164 { 163 {
165 public: 164 public:
166 friend class VoERtcpObserver; 165 friend class VoERtcpObserver;
167 166
168 enum { KNumSocketThreads = 1 }; 167 enum { KNumSocketThreads = 1 };
169 enum { KNumberOfSocketBuffers = 8 }; 168 enum { KNumberOfSocketBuffers = 8 };
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378 377
379 // From RtpFeedback in the RTP/RTCP module 378 // From RtpFeedback in the RTP/RTCP module
380 int32_t OnInitializeDecoder(int8_t payloadType, 379 int32_t OnInitializeDecoder(int8_t payloadType,
381 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 380 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
382 int frequency, 381 int frequency,
383 size_t channels, 382 size_t channels,
384 uint32_t rate) override; 383 uint32_t rate) override;
385 void OnIncomingSSRCChanged(uint32_t ssrc) override; 384 void OnIncomingSSRCChanged(uint32_t ssrc) override;
386 void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override; 385 void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
387 386
388 // From RtpAudioFeedback in the RTP/RTCP module
389 void OnPlayTelephoneEvent(uint8_t event,
390 uint16_t lengthMs,
391 uint8_t volume) override;
392
393 // From Transport (called by the RTP/RTCP module) 387 // From Transport (called by the RTP/RTCP module)
394 bool SendRtp(const uint8_t* data, 388 bool SendRtp(const uint8_t* data,
395 size_t len, 389 size_t len,
396 const PacketOptions& packet_options) override; 390 const PacketOptions& packet_options) override;
397 bool SendRtcp(const uint8_t* data, size_t len) override; 391 bool SendRtcp(const uint8_t* data, size_t len) override;
398 392
399 // From MixerParticipant 393 // From MixerParticipant
400 int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override; 394 int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override;
401 int32_t NeededFrequency(int32_t id) const override; 395 int32_t NeededFrequency(int32_t id) const override;
402 396
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573 PacketRouter* packet_router_ = nullptr; 567 PacketRouter* packet_router_ = nullptr;
574 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; 568 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
575 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; 569 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
576 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; 570 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
577 }; 571 };
578 572
579 } // namespace voe 573 } // namespace voe
580 } // namespace webrtc 574 } // namespace webrtc
581 575
582 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 576 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
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