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Side by Side Diff: content/renderer/media/rtc_peer_connection_handler.cc

Issue 1834323002: MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Reworked unit tests around structural changes, and added exhaustive media_stream_audio_unittest.cc. Created 4 years, 7 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "content/renderer/media/rtc_peer_connection_handler.h" 5 #include "content/renderer/media/rtc_peer_connection_handler.h"
6 6
7 #include <string.h> 7 #include <string.h>
8 8
9 #include <string> 9 #include <string>
10 #include <utility> 10 #include <utility>
11 #include <vector> 11 #include <vector>
12 12
13 #include "base/command_line.h" 13 #include "base/command_line.h"
14 #include "base/lazy_instance.h" 14 #include "base/lazy_instance.h"
15 #include "base/location.h" 15 #include "base/location.h"
16 #include "base/logging.h" 16 #include "base/logging.h"
17 #include "base/metrics/histogram.h" 17 #include "base/metrics/histogram.h"
18 #include "base/metrics/sparse_histogram.h" 18 #include "base/metrics/sparse_histogram.h"
19 #include "base/stl_util.h" 19 #include "base/stl_util.h"
20 #include "base/strings/utf_string_conversions.h" 20 #include "base/strings/utf_string_conversions.h"
21 #include "base/thread_task_runner_handle.h" 21 #include "base/thread_task_runner_handle.h"
22 #include "base/trace_event/trace_event.h" 22 #include "base/trace_event/trace_event.h"
23 #include "content/public/common/content_features.h" 23 #include "content/public/common/content_features.h"
24 #include "content/public/common/content_switches.h" 24 #include "content/public/common/content_switches.h"
25 #include "content/renderer/media/media_stream_audio_track.h"
26 #include "content/renderer/media/media_stream_constraints_util.h" 25 #include "content/renderer/media/media_stream_constraints_util.h"
27 #include "content/renderer/media/media_stream_track.h" 26 #include "content/renderer/media/media_stream_track.h"
28 #include "content/renderer/media/peer_connection_tracker.h" 27 #include "content/renderer/media/peer_connection_tracker.h"
29 #include "content/renderer/media/remote_media_stream_impl.h" 28 #include "content/renderer/media/remote_media_stream_impl.h"
30 #include "content/renderer/media/rtc_certificate.h" 29 #include "content/renderer/media/rtc_certificate.h"
31 #include "content/renderer/media/rtc_data_channel_handler.h" 30 #include "content/renderer/media/rtc_data_channel_handler.h"
32 #include "content/renderer/media/rtc_dtmf_sender_handler.h" 31 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
33 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" 32 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
34 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h" 33 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
35 #include "content/renderer/media/webrtc_audio_capturer.h"
36 #include "content/renderer/media/webrtc_audio_device_impl.h" 34 #include "content/renderer/media/webrtc_audio_device_impl.h"
37 #include "content/renderer/media/webrtc_uma_histograms.h" 35 #include "content/renderer/media/webrtc_uma_histograms.h"
38 #include "content/renderer/render_thread_impl.h" 36 #include "content/renderer/render_thread_impl.h"
39 #include "media/base/media_switches.h" 37 #include "media/base/media_switches.h"
40 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" 38 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
41 #include "third_party/WebKit/public/platform/WebRTCAnswerOptions.h" 39 #include "third_party/WebKit/public/platform/WebRTCAnswerOptions.h"
42 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h" 40 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
43 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h" 41 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
44 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h" 42 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
45 #include "third_party/WebKit/public/platform/WebRTCOfferOptions.h" 43 #include "third_party/WebKit/public/platform/WebRTCOfferOptions.h"
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1478 1476
1479 ++num_data_channels_created_; 1477 ++num_data_channels_created_;
1480 1478
1481 return new RtcDataChannelHandler(base::ThreadTaskRunnerHandle::Get(), 1479 return new RtcDataChannelHandler(base::ThreadTaskRunnerHandle::Get(),
1482 webrtc_channel); 1480 webrtc_channel);
1483 } 1481 }
1484 1482
1485 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender( 1483 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
1486 const blink::WebMediaStreamTrack& track) { 1484 const blink::WebMediaStreamTrack& track) {
1487 DCHECK(thread_checker_.CalledOnValidThread()); 1485 DCHECK(thread_checker_.CalledOnValidThread());
1486 DCHECK(!track.isNull());
1488 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender"); 1487 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender");
1489 DVLOG(1) << "createDTMFSender."; 1488 DVLOG(1) << "createDTMFSender.";
1490 1489
1491 MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::From(track); 1490 // Find the WebRtc track referenced by the blink track's ID.
1492 if (!native_track || !native_track->is_local_track() || 1491 webrtc::AudioTrackInterface* webrtc_track = nullptr;
1493 track.source().getType() != blink::WebMediaStreamSource::TypeAudio) { 1492 for (const WebRtcMediaStreamAdapter* s : local_streams_) {
1494 DLOG(ERROR) << "The DTMF sender requires a local audio track."; 1493 webrtc_track = s->webrtc_media_stream()->FindAudioTrack(track.id().utf8());
1494 if (webrtc_track)
1495 break;
1496 }
1497 if (!webrtc_track) {
1498 DLOG(ERROR) << "Audio track with ID '" << track.id().utf8()
1499 << "' has no known WebRtc sink.";
1495 return nullptr; 1500 return nullptr;
1496 } 1501 }
1497 1502
1498 scoped_refptr<webrtc::AudioTrackInterface> audio_track =
1499 native_track->GetAudioAdapter();
1500 rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender( 1503 rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender(
1501 native_peer_connection_->CreateDtmfSender(audio_track.get())); 1504 native_peer_connection_->CreateDtmfSender(webrtc_track));
1502 if (!sender) { 1505 if (!sender) {
1503 DLOG(ERROR) << "Could not create native DTMF sender."; 1506 DLOG(ERROR) << "Could not create native DTMF sender.";
1504 return nullptr; 1507 return nullptr;
1505 } 1508 }
1506 if (peer_connection_tracker_) 1509 if (peer_connection_tracker_)
1507 peer_connection_tracker_->TrackCreateDTMFSender(this, track); 1510 peer_connection_tracker_->TrackCreateDTMFSender(this, track);
1508 1511
1509 return new RtcDtmfSenderHandler(sender); 1512 return new RtcDtmfSenderHandler(sender);
1510 } 1513 }
1511 1514
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1795 } 1798 }
1796 1799
1797 void RTCPeerConnectionHandler::ResetUMAStats() { 1800 void RTCPeerConnectionHandler::ResetUMAStats() {
1798 DCHECK(thread_checker_.CalledOnValidThread()); 1801 DCHECK(thread_checker_.CalledOnValidThread());
1799 num_local_candidates_ipv6_ = 0; 1802 num_local_candidates_ipv6_ = 0;
1800 num_local_candidates_ipv4_ = 0; 1803 num_local_candidates_ipv4_ = 0;
1801 ice_connection_checking_start_ = base::TimeTicks(); 1804 ice_connection_checking_start_ = base::TimeTicks();
1802 memset(ice_state_seen_, 0, sizeof(ice_state_seen_)); 1805 memset(ice_state_seen_, 0, sizeof(ice_state_seen_));
1803 } 1806 }
1804 } // namespace content 1807 } // namespace content
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