Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(159)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

Issue 1812453002: Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@voe_dtmf_7
Patch Set: Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
11 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h" 11 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
12 12
13 #include <assert.h> // assert 13 #include <assert.h> // assert
14 #include <math.h> // pow() 14 #include <math.h> // pow()
15 #include <string.h> // memcpy() 15 #include <string.h> // memcpy()
16 16
17 #include "webrtc/base/logging.h" 17 #include "webrtc/base/logging.h"
18 #include "webrtc/base/trace_event.h" 18 #include "webrtc/base/trace_event.h"
19 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 19 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
20 20
21 namespace webrtc { 21 namespace webrtc {
22 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy( 22 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
23 RtpData* data_callback) { 23 RtpData* data_callback,
24 return new RTPReceiverAudio(data_callback); 24 RtpAudioFeedback* incoming_messages_callback) {
25 return new RTPReceiverAudio(data_callback, incoming_messages_callback);
25 } 26 }
26 27
27 RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback) 28 RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
29 RtpAudioFeedback* incoming_messages_callback)
28 : RTPReceiverStrategy(data_callback), 30 : RTPReceiverStrategy(data_callback),
29 TelephoneEventHandler(), 31 TelephoneEventHandler(),
30 last_received_frequency_(8000), 32 last_received_frequency_(8000),
31 telephone_event_forward_to_decoder_(false), 33 telephone_event_forward_to_decoder_(false),
32 telephone_event_payload_type_(-1), 34 telephone_event_payload_type_(-1),
33 cng_nb_payload_type_(-1), 35 cng_nb_payload_type_(-1),
34 cng_wb_payload_type_(-1), 36 cng_wb_payload_type_(-1),
35 cng_swb_payload_type_(-1), 37 cng_swb_payload_type_(-1),
36 cng_fb_payload_type_(-1), 38 cng_fb_payload_type_(-1),
37 cng_payload_type_(-1), 39 cng_payload_type_(-1),
38 g722_payload_type_(-1), 40 g722_payload_type_(-1),
39 last_received_g722_(false), 41 last_received_g722_(false),
40 num_energy_(0), 42 num_energy_(0),
41 current_remote_energy_() { 43 current_remote_energy_(),
44 cb_audio_feedback_(incoming_messages_callback) {
42 last_payload_.Audio.channels = 1; 45 last_payload_.Audio.channels = 1;
43 memset(current_remote_energy_, 0, sizeof(current_remote_energy_)); 46 memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
44 } 47 }
45 48
46 // Outband TelephoneEvent(DTMF) detection 49 // Outband TelephoneEvent(DTMF) detection
47 void RTPReceiverAudio::SetTelephoneEventForwardToDecoder( 50 void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
48 bool forward_to_decoder) { 51 bool forward_to_decoder) {
49 CriticalSectionScoped lock(crit_sect_.get()); 52 CriticalSectionScoped lock(crit_sect_.get());
50 telephone_event_forward_to_decoder_ = forward_to_decoder; 53 telephone_event_forward_to_decoder_ = forward_to_decoder;
51 } 54 }
(...skipping 321 matching lines...) Expand 10 before | Expand all | Expand 10 after
373 // only one frame in the RED strip the one byte to help NetEq 376 // only one frame in the RED strip the one byte to help NetEq
374 return data_callback_->OnReceivedPayloadData( 377 return data_callback_->OnReceivedPayloadData(
375 payload_data + 1, payload_length - 1, rtp_header); 378 payload_data + 1, payload_length - 1, rtp_header);
376 } 379 }
377 380
378 rtp_header->type.Audio.channel = audio_specific.channels; 381 rtp_header->type.Audio.channel = audio_specific.channels;
379 return data_callback_->OnReceivedPayloadData( 382 return data_callback_->OnReceivedPayloadData(
380 payload_data, payload_length, rtp_header); 383 payload_data, payload_length, rtp_header);
381 } 384 }
382 } // namespace webrtc 385 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h ('k') | webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698