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

Refactor the PlatformThread API.

Browse Source 
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.

Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
  a) Handles for detached threads are simply closed where applicable.
  b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.

Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
This commit is contained in:
Markus Handell
2021-05-05 10:42:04 +02:00
committed by WebRTC LUCI CQ
parent 1c5c2178fe
commit c89fdd716c
38 changed files with 586 additions and 864 deletions
Download Patch File Download Diff File Expand all files Collapse all files

55
modules/audio_device/linux/audio_device_pulse_linux.cc
View File

@ -15,6 +15,7 @@
#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/platform_thread.h"
WebRTCPulseSymbolTable* GetPulseSymbolTable() {
static WebRTCPulseSymbolTable* pulse_symbol_table =
@ -158,18 +159,22 @@ AudioDeviceGeneric::InitStatus AudioDeviceLinuxPulse::Init() {
#endif
// RECORDING
_ptrThreadRec.reset(new rtc::PlatformThread(
RecThreadFunc, this, "webrtc_audio_module_rec_thread",
rtc::ThreadAttributes().SetPriority(rtc::kRealtimePriority)));
_ptrThreadRec->Start();
const auto attributes =
rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
_ptrThreadRec = rtc::PlatformThread::SpawnJoinable(
[this] {
while (RecThreadProcess()) {
}
},
"webrtc_audio_module_rec_thread", attributes);
// PLAYOUT
_ptrThreadPlay.reset(new rtc::PlatformThread(
PlayThreadFunc, this, "webrtc_audio_module_play_thread",
rtc::ThreadAttributes().SetPriority(rtc::kRealtimePriority)));
_ptrThreadPlay->Start();
_ptrThreadPlay = rtc::PlatformThread::SpawnJoinable(
[this] {
while (PlayThreadProcess()) {
}
},
"webrtc_audio_module_play_thread", attributes);
_initialized = true;
return InitStatus::OK;
@ -187,22 +192,12 @@ int32_t AudioDeviceLinuxPulse::Terminate() {
_mixerManager.Close();
// RECORDING
if (_ptrThreadRec) {
rtc::PlatformThread* tmpThread = _ptrThreadRec.release();
_timeEventRec.Set();
tmpThread->Stop();
delete tmpThread;
}
_timeEventRec.Set();
_ptrThreadRec.Finalize();
// PLAYOUT
if (_ptrThreadPlay) {
rtc::PlatformThread* tmpThread = _ptrThreadPlay.release();
_timeEventPlay.Set();
tmpThread->Stop();
delete tmpThread;
}
_timeEventPlay.Set();
_ptrThreadPlay.Finalize();
// Terminate PulseAudio
if (TerminatePulseAudio() < 0) {
@ -1981,18 +1976,6 @@ int32_t AudioDeviceLinuxPulse::ProcessRecordedData(int8_t* bufferData,
return 0;
}
void AudioDeviceLinuxPulse::PlayThreadFunc(void* pThis) {
AudioDeviceLinuxPulse* device = static_cast<AudioDeviceLinuxPulse*>(pThis);
while (device->PlayThreadProcess()) {
}
}
void AudioDeviceLinuxPulse::RecThreadFunc(void* pThis) {
AudioDeviceLinuxPulse* device = static_cast<AudioDeviceLinuxPulse*>(pThis);
while (device->RecThreadProcess()) {
}
}
bool AudioDeviceLinuxPulse::PlayThreadProcess() {
if (!_timeEventPlay.Wait(1000)) {
return true;