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36d4155112
Bug: webrtc:15094 Change-Id: I9b9dd8d7cba3ccfb1e8acdb6e1df42f9efe1cea6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303780 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39984}
112 lines
4.7 KiB
Objective-C
112 lines
4.7 KiB
Objective-C
/*
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* Copyright 2016 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import "RTCAudioSessionConfiguration.h"
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#import "RTCAudioSession.h"
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#import "helpers/RTCDispatcher.h"
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#import "helpers/UIDevice+RTCDevice.h"
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// Try to use mono to save resources. Also avoids channel format conversion
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// in the I/O audio unit. Initial tests have shown that it is possible to use
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// mono natively for built-in microphones and for BT headsets but not for
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// wired headsets. Wired headsets only support stereo as native channel format
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// but it is a low cost operation to do a format conversion to mono in the
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// audio unit. Hence, we will not hit a RTC_CHECK in
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// VerifyAudioParametersForActiveAudioSession() for a mismatch between the
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// preferred number of channels and the actual number of channels.
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const int kRTCAudioSessionPreferredNumberOfChannels = 1;
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// Preferred hardware sample rate (unit is in Hertz). The client sample rate
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// will be set to this value as well to avoid resampling the the audio unit's
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// format converter. Note that, some devices, e.g. BT headsets, only supports
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// 8000Hz as native sample rate.
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const double kRTCAudioSessionHighPerformanceSampleRate = 48000.0;
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// Use a hardware I/O buffer size (unit is in seconds) that matches the 10ms
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// size used by WebRTC. The exact actual size will differ between devices.
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// Example: using 48kHz on iPhone 6 results in a native buffer size of
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// ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
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// take care of any buffering required to convert between native buffers and
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// buffers used by WebRTC. It is beneficial for the performance if the native
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// size is as an even multiple of 10ms as possible since it results in "clean"
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// callback sequence without bursts of callbacks back to back.
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const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.02;
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static RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *gWebRTCConfiguration = nil;
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@implementation RTC_OBJC_TYPE (RTCAudioSessionConfiguration)
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@synthesize category = _category;
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@synthesize categoryOptions = _categoryOptions;
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@synthesize mode = _mode;
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@synthesize sampleRate = _sampleRate;
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@synthesize ioBufferDuration = _ioBufferDuration;
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@synthesize inputNumberOfChannels = _inputNumberOfChannels;
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@synthesize outputNumberOfChannels = _outputNumberOfChannels;
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- (instancetype)init {
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if (self = [super init]) {
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// Use a category which supports simultaneous recording and playback.
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// By default, using this category implies that our app’s audio is
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// nonmixable, hence activating the session will interrupt any other
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// audio sessions which are also nonmixable.
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_category = AVAudioSessionCategoryPlayAndRecord;
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_categoryOptions = AVAudioSessionCategoryOptionAllowBluetooth;
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// Specify mode for two-way voice communication (e.g. VoIP).
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_mode = AVAudioSessionModeVoiceChat;
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// Use best sample rate and buffer duration if the CPU has more than one
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// core.
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_sampleRate = kRTCAudioSessionHighPerformanceSampleRate;
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_ioBufferDuration = kRTCAudioSessionHighPerformanceIOBufferDuration;
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// We try to use mono in both directions to save resources and format
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// conversions in the audio unit. Some devices does only support stereo;
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// e.g. wired headset on iPhone 6.
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// TODO(henrika): add support for stereo if needed.
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_inputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
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_outputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
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}
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return self;
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}
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+ (void)initialize {
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gWebRTCConfiguration = [[self alloc] init];
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}
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+ (instancetype)currentConfiguration {
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RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
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RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *config =
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[[RTC_OBJC_TYPE(RTCAudioSessionConfiguration) alloc] init];
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config.category = session.category;
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config.categoryOptions = session.categoryOptions;
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config.mode = session.mode;
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config.sampleRate = session.sampleRate;
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config.ioBufferDuration = session.IOBufferDuration;
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config.inputNumberOfChannels = session.inputNumberOfChannels;
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config.outputNumberOfChannels = session.outputNumberOfChannels;
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return config;
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}
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+ (instancetype)webRTCConfiguration {
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@synchronized(self) {
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return (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)gWebRTCConfiguration;
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}
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}
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+ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration {
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@synchronized(self) {
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gWebRTCConfiguration = configuration;
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}
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}
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@end
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