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Issue 545203002: Improved native WebRTC logging for failing initialization of MS source (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 3 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 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/webrtc/peer_connection_dependency_factory.h" 5 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
6 6
7 #include <vector> 7 #include <vector>
8 8
9 #include "base/command_line.h" 9 #include "base/command_line.h"
10 #include "base/strings/stringprintf.h"
10 #include "base/strings/utf_string_conversions.h" 11 #include "base/strings/utf_string_conversions.h"
11 #include "base/synchronization/waitable_event.h" 12 #include "base/synchronization/waitable_event.h"
12 #include "content/common/media/media_stream_messages.h" 13 #include "content/common/media/media_stream_messages.h"
13 #include "content/public/common/content_switches.h" 14 #include "content/public/common/content_switches.h"
14 #include "content/renderer/media/media_stream.h" 15 #include "content/renderer/media/media_stream.h"
15 #include "content/renderer/media/media_stream_audio_processor.h" 16 #include "content/renderer/media/media_stream_audio_processor.h"
16 #include "content/renderer/media/media_stream_audio_processor_options.h" 17 #include "content/renderer/media/media_stream_audio_processor_options.h"
17 #include "content/renderer/media/media_stream_audio_source.h" 18 #include "content/renderer/media/media_stream_audio_source.h"
18 #include "content/renderer/media/media_stream_video_source.h" 19 #include "content/renderer/media/media_stream_video_source.h"
19 #include "content/renderer/media/media_stream_video_track.h" 20 #include "content/renderer/media/media_stream_video_track.h"
20 #include "content/renderer/media/peer_connection_identity_service.h" 21 #include "content/renderer/media/peer_connection_identity_service.h"
21 #include "content/renderer/media/rtc_media_constraints.h" 22 #include "content/renderer/media/rtc_media_constraints.h"
22 #include "content/renderer/media/rtc_peer_connection_handler.h" 23 #include "content/renderer/media/rtc_peer_connection_handler.h"
23 #include "content/renderer/media/rtc_video_decoder_factory.h" 24 #include "content/renderer/media/rtc_video_decoder_factory.h"
24 #include "content/renderer/media/rtc_video_encoder_factory.h" 25 #include "content/renderer/media/rtc_video_encoder_factory.h"
25 #include "content/renderer/media/webaudio_capturer_source.h" 26 #include "content/renderer/media/webaudio_capturer_source.h"
26 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 27 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
27 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" 28 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
28 #include "content/renderer/media/webrtc_audio_device_impl.h" 29 #include "content/renderer/media/webrtc_audio_device_impl.h"
29 #include "content/renderer/media/webrtc_local_audio_track.h" 30 #include "content/renderer/media/webrtc_local_audio_track.h"
31 #include "content/renderer/media/webrtc_logging.h"
30 #include "content/renderer/media/webrtc_uma_histograms.h" 32 #include "content/renderer/media/webrtc_uma_histograms.h"
31 #include "content/renderer/p2p/ipc_network_manager.h" 33 #include "content/renderer/p2p/ipc_network_manager.h"
32 #include "content/renderer/p2p/ipc_socket_factory.h" 34 #include "content/renderer/p2p/ipc_socket_factory.h"
33 #include "content/renderer/p2p/port_allocator.h" 35 #include "content/renderer/p2p/port_allocator.h"
34 #include "content/renderer/render_thread_impl.h" 36 #include "content/renderer/render_thread_impl.h"
35 #include "jingle/glue/thread_wrapper.h" 37 #include "jingle/glue/thread_wrapper.h"
36 #include "media/filters/gpu_video_accelerator_factories.h" 38 #include "media/filters/gpu_video_accelerator_factories.h"
37 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" 39 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
38 #include "third_party/WebKit/public/platform/WebMediaStream.h" 40 #include "third_party/WebKit/public/platform/WebMediaStream.h"
39 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" 41 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
(...skipping 169 matching lines...) Expand 10 before | Expand all | Expand 10 after
209 StreamDeviceInfo device_info = source_data->device_info(); 211 StreamDeviceInfo device_info = source_data->device_info();
210 RTCMediaConstraints constraints = native_audio_constraints; 212 RTCMediaConstraints constraints = native_audio_constraints;
211 // May modify both |constraints| and |effects|. 213 // May modify both |constraints| and |effects|.
212 HarmonizeConstraintsAndEffects(&constraints, 214 HarmonizeConstraintsAndEffects(&constraints,
213 &device_info.device.input.effects); 215 &device_info.device.input.effects);
214 216
215 scoped_refptr<WebRtcAudioCapturer> capturer( 217 scoped_refptr<WebRtcAudioCapturer> capturer(
216 CreateAudioCapturer(render_view_id, device_info, audio_constraints, 218 CreateAudioCapturer(render_view_id, device_info, audio_constraints,
217 source_data)); 219 source_data));
218 if (!capturer.get()) { 220 if (!capturer.get()) {
219 DLOG(WARNING) << "Failed to create the capturer for device " 221 std::string log_string = base::StringPrintf(
no longer working on chromium 2014/09/05 12:57:23 nit, const std::string
henrika (OOO until Aug 14) 2014/09/05 13:02:21 Done.
220 << device_info.device.id; 222 "PCDF::InitializeMediaStreamAudioSource: "
223 "fails to create capturer for device %s",
224 device_info.device.id.c_str());
no longer working on chromium 2014/09/05 12:57:23 You can't log the device id.
henrika (OOO until Aug 14) 2014/09/05 13:02:21 Yes I can ;-)
225 WebRtcLogMessage(log_string);
226 DVLOG(1) << log_string;
221 // TODO(xians): Don't we need to check if source_observer is observing 227 // TODO(xians): Don't we need to check if source_observer is observing
222 // something? If not, then it looks like we have a leak here. 228 // something? If not, then it looks like we have a leak here.
223 // OTOH, if it _is_ observing something, then the callback might 229 // OTOH, if it _is_ observing something, then the callback might
224 // be called multiple times which is likely also a bug. 230 // be called multiple times which is likely also a bug.
225 return false; 231 return false;
226 } 232 }
227 source_data->SetAudioCapturer(capturer.get()); 233 source_data->SetAudioCapturer(capturer.get());
228 234
229 // Creates a LocalAudioSource object which holds audio options. 235 // Creates a LocalAudioSource object which holds audio options.
230 // TODO(xians): The option should apply to the track instead of the source. 236 // TODO(xians): The option should apply to the track instead of the source.
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654 } 660 }
655 661
656 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { 662 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
657 if (audio_device_.get()) 663 if (audio_device_.get())
658 return; 664 return;
659 665
660 audio_device_ = new WebRtcAudioDeviceImpl(); 666 audio_device_ = new WebRtcAudioDeviceImpl();
661 } 667 }
662 668
663 } // namespace content 669 } // namespace content
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