zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Wrap WebRTC OBJC API types with RTC_OBJC_TYPE.

Browse Source 
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:

- RTC_OBJC_TYPE_PREFIX:
  Macro used to prepend a prefix to the API types that are exported with
  RTC_OBJC_EXPORT.

  Clients can patch the definition of this macro locally and build
  WebRTC.framework with their own prefix in case symbol clashing is a
  problem.

  This macro must only be defined by changing the value in
  sdk/objc/base/RTCMacros.h  and not on via compiler flag to ensure
  it has a unique value.

- RCT_OBJC_TYPE:
  Macro used internally to reference API types. Declaring an API type
  without using this macro will not include the declared type in the
  set of types that will be affected by the configurable
  RTC_OBJC_TYPE_PREFIX.

Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10

The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.

Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
This commit is contained in:
Mirko Bonadei
2020-05-04 16:14:32 +02:00
committed by Commit Bot
parent ce1320cc4d
commit a81e9c82fc
303 changed files with 2534 additions and 2189 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
sdk/objc/native/src/audio/audio_device_ios.mm
View File

@ -152,7 +152,8 @@ AudioDeviceGeneric::InitStatus AudioDeviceIOS::Init() {
// here. They have not been set and confirmed yet since configureForWebRTC
// is not called until audio is about to start. However, it makes sense to
// store the parameters now and then verify at a later stage.
RTCAudioSessionConfiguration* config = [RTCAudioSessionConfiguration webRTCConfiguration];
RTC_OBJC_TYPE(RTCAudioSessionConfiguration)* config =
[RTC_OBJC_TYPE(RTCAudioSessionConfiguration) webRTCConfiguration];
playout_parameters_.reset(config.sampleRate, config.outputNumberOfChannels);
record_parameters_.reset(config.sampleRate, config.inputNumberOfChannels);
// Ensure that the audio device buffer (ADB) knows about the internal audio
@ -532,12 +533,12 @@ void AudioDeviceIOS::HandleInterruptionEnd() {
// Allocate new buffers given the potentially new stream format.
SetupAudioBuffersForActiveAudioSession();
}
UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord);
UpdateAudioUnit([RTC_OBJC_TYPE(RTCAudioSession) sharedInstance].canPlayOrRecord);
}
void AudioDeviceIOS::HandleValidRouteChange() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
RTCLog(@"%@", session);
HandleSampleRateChange(session.sampleRate);
}
@ -565,7 +566,7 @@ void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) {
// The audio unit is already initialized or started.
// Check to see if the sample rate or buffer size has changed.
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
const double session_sample_rate = session.sampleRate;
const NSTimeInterval session_buffer_duration = session.IOBufferDuration;
const size_t session_frames_per_buffer =
@ -646,7 +647,7 @@ void AudioDeviceIOS::HandlePlayoutGlitchDetected() {
int64_t glitch_count = num_detected_playout_glitches_;
dispatch_async(dispatch_get_main_queue(), ^{
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
[session notifyDidDetectPlayoutGlitch:glitch_count];
});
}
@ -678,7 +679,7 @@ void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
LOGI() << "SetupAudioBuffersForActiveAudioSession";
// Verify the current values once the audio session has been activated.
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
double sample_rate = session.sampleRate;
NSTimeInterval io_buffer_duration = session.IOBufferDuration;
RTCLog(@"%@", session);
@ -687,7 +688,8 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
// hardware sample rate but continue and use the non-ideal sample rate after
// reinitializing the audio parameters. Most BT headsets only support 8kHz or
// 16kHz.
RTCAudioSessionConfiguration* webRTCConfig = [RTCAudioSessionConfiguration webRTCConfiguration];
RTC_OBJC_TYPE(RTCAudioSessionConfiguration)* webRTCConfig =
[RTC_OBJC_TYPE(RTCAudioSessionConfiguration) webRTCConfiguration];
if (sample_rate != webRTCConfig.sampleRate) {
RTC_LOG(LS_WARNING) << "Unable to set the preferred sample rate";
}
@ -797,7 +799,7 @@ void AudioDeviceIOS::UpdateAudioUnit(bool can_play_or_record) {
if (should_start_audio_unit) {
RTCLog(@"Starting audio unit for UpdateAudioUnit");
// Log session settings before trying to start audio streaming.
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
RTCLog(@"%@", session);
if (!audio_unit_->Start()) {
RTCLogError(@"Failed to start audio unit.");
@ -827,7 +829,7 @@ bool AudioDeviceIOS::ConfigureAudioSession() {
RTCLogWarning(@"Audio session already configured.");
return false;
}
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
[session lockForConfiguration];
bool success = [session configureWebRTCSession:nil];
[session unlockForConfiguration];
@ -847,7 +849,7 @@ void AudioDeviceIOS::UnconfigureAudioSession() {
RTCLogWarning(@"Audio session already unconfigured.");
return;
}
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
[session lockForConfiguration];
[session unconfigureWebRTCSession:nil];
[session endWebRTCSession:nil];
@ -865,7 +867,7 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
return false;
}
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
// Subscribe to audio session events.
[session pushDelegate:audio_session_observer_];
is_interrupted_ = session.isInterrupted ? true : false;
@ -915,7 +917,7 @@ void AudioDeviceIOS::ShutdownPlayOrRecord() {
io_thread_checker_.Detach();
// Remove audio session notification observers.
RTCAudioSession* session = [RTCAudioSession sharedInstance];
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
[session removeDelegate:audio_session_observer_];
// All I/O should be stopped or paused prior to deactivating the audio