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Renamed methods.

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Renaming inputSampleRate, outputSampleRate, terminate to avoid triggering Apple's private API check.

Change-Id: I9857fb374bf30c4a6ef937fb183ef4858af7e0c1
Bug: webrtc:14193
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275641
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38094}
This commit is contained in:
Peter Hanspers
2022-09-15 13:52:07 +02:00
committed by WebRTC LUCI CQ
parent bcf24f5bcd
commit 1a59cb6108
2 changed files with 6 additions and 6 deletions
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6
sdk/objc/components/audio/RTCAudioDevice.h
View File

@ -171,7 +171,7 @@ RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate
* notifyAudioParametersChange]`.
*/
@property(readonly) double inputSampleRate;
@property(readonly) double deviceInputSampleRate;
/**
* Indicates current size of record buffer. Changes to this property
@ -194,7 +194,7 @@ RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
* Indicates current sample rate of audio playback. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) double outputSampleRate;
@property(readonly) double deviceOutputSampleRate;
/**
* Indicates current size of playback buffer. Changes to this property
@ -230,7 +230,7 @@ RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
* De-initializes RTCAudioDevice. Implementation should forget about `delegate` provided in
* `initializeWithDelegate`.
*/
- (BOOL)terminate;
- (BOOL)terminateDevice;
/**
* Property to indicate if `initializePlayout` call required before invocation of `startPlayout`.

6
sdk/objc/native/src/objc_audio_device.mm
View File

@ -22,7 +22,7 @@
namespace {
webrtc::AudioParameters RecordParameters(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) {
const double sample_rate = static_cast<int>([audio_device inputSampleRate]);
const double sample_rate = static_cast<int>([audio_device deviceInputSampleRate]);
const size_t channels = static_cast<size_t>([audio_device inputNumberOfChannels]);
const size_t frames_per_buffer =
static_cast<size_t>(sample_rate * [audio_device inputIOBufferDuration] + .5);
@ -30,7 +30,7 @@ webrtc::AudioParameters RecordParameters(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio
}
webrtc::AudioParameters PlayoutParameters(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) {
const double sample_rate = static_cast<int>([audio_device outputSampleRate]);
const double sample_rate = static_cast<int>([audio_device deviceOutputSampleRate]);
const size_t channels = static_cast<size_t>([audio_device outputNumberOfChannels]);
const size_t frames_per_buffer =
static_cast<size_t>(sample_rate * [audio_device outputIOBufferDuration] + .5);
@ -112,7 +112,7 @@ int32_t ObjCAudioDeviceModule::Terminate() {
}
if ([audio_device_ isInitialized]) {
if (![audio_device_ terminate]) {
if (![audio_device_ terminateDevice]) {
RTC_LOG_F(LS_ERROR) << "Failed to terminate audio device";
return -1;
}