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Issue 225143006: Limit the max volume to 255 before pass it to APM. (Closed) Base URL: http://git.chromium.org/chromium/src.git@master
Patch Set: fixed the comment Created 6 years, 8 months ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 1 // Copyright 2013 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/media_stream_audio_processor.h" 5 #include "content/renderer/media/media_stream_audio_processor.h"
6 6
7 #include "base/command_line.h" 7 #include "base/command_line.h"
8 #include "base/debug/trace_event.h" 8 #include "base/debug/trace_event.h"
9 #include "base/metrics/field_trial.h" 9 #include "base/metrics/field_trial.h"
10 #include "base/metrics/histogram.h" 10 #include "base/metrics/histogram.h"
11 #include "content/public/common/content_switches.h" 11 #include "content/public/common/content_switches.h"
12 #include "content/renderer/media/media_stream_audio_processor_options.h" 12 #include "content/renderer/media/media_stream_audio_processor_options.h"
13 #include "content/renderer/media/rtc_media_constraints.h" 13 #include "content/renderer/media/rtc_media_constraints.h"
14 #include "content/renderer/media/webrtc_audio_device_impl.h"
14 #include "media/audio/audio_parameters.h" 15 #include "media/audio/audio_parameters.h"
15 #include "media/base/audio_converter.h" 16 #include "media/base/audio_converter.h"
16 #include "media/base/audio_fifo.h" 17 #include "media/base/audio_fifo.h"
17 #include "media/base/channel_layout.h" 18 #include "media/base/channel_layout.h"
18 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" 19 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
19 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" 20 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h"
20 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" 21 #include "third_party/webrtc/modules/audio_processing/typing_detection.h"
21 22
22 namespace content { 23 namespace content {
23 24
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467 DCHECK_LT(capture_delay_ms, 468 DCHECK_LT(capture_delay_ms,
468 std::numeric_limits<base::subtle::Atomic32>::max()); 469 std::numeric_limits<base::subtle::Atomic32>::max());
469 int total_delay_ms = capture_delay_ms + render_delay_ms; 470 int total_delay_ms = capture_delay_ms + render_delay_ms;
470 if (total_delay_ms > 300) { 471 if (total_delay_ms > 300) {
471 LOG(WARNING) << "Large audio delay, capture delay: " << capture_delay_ms 472 LOG(WARNING) << "Large audio delay, capture delay: " << capture_delay_ms
472 << "ms; render delay: " << render_delay_ms << "ms"; 473 << "ms; render delay: " << render_delay_ms << "ms";
473 } 474 }
474 475
475 audio_processing_->set_stream_delay_ms(total_delay_ms); 476 audio_processing_->set_stream_delay_ms(total_delay_ms);
476 477
478 DCHECK_LE(volume, WebRtcAudioDeviceImpl::kMaxVolumeLevel);
477 webrtc::GainControl* agc = audio_processing_->gain_control(); 479 webrtc::GainControl* agc = audio_processing_->gain_control();
478 int err = agc->set_stream_analog_level(volume); 480 int err = agc->set_stream_analog_level(volume);
479 DCHECK_EQ(err, 0) << "set_stream_analog_level() error: " << err; 481 DCHECK_EQ(err, 0) << "set_stream_analog_level() error: " << err;
480 482
481 audio_processing_->set_stream_key_pressed(key_pressed); 483 audio_processing_->set_stream_key_pressed(key_pressed);
482 484
483 err = audio_processing_->ProcessStream(audio_frame); 485 err = audio_processing_->ProcessStream(audio_frame);
484 DCHECK_EQ(err, 0) << "ProcessStream() error: " << err; 486 DCHECK_EQ(err, 0) << "ProcessStream() error: " << err;
485 487
486 if (typing_detector_ && 488 if (typing_detector_ &&
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510 } 512 }
511 513
512 bool MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() const { 514 bool MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() const {
513 const std::string group_name = 515 const std::string group_name =
514 base::FieldTrialList::FindFullName("MediaStreamAudioTrackProcessing"); 516 base::FieldTrialList::FindFullName("MediaStreamAudioTrackProcessing");
515 return group_name == "Enabled" || CommandLine::ForCurrentProcess()->HasSwitch( 517 return group_name == "Enabled" || CommandLine::ForCurrentProcess()->HasSwitch(
516 switches::kEnableAudioTrackProcessing); 518 switches::kEnableAudioTrackProcessing);
517 } 519 }
518 520
519 } // namespace content 521 } // namespace content
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