Modified: branches/safari-609-branch/Source/WebCore/ChangeLog (261941 => 261942)
--- branches/safari-609-branch/Source/WebCore/ChangeLog 2020-05-20 19:59:53 UTC (rev 261941)
+++ branches/safari-609-branch/Source/WebCore/ChangeLog 2020-05-20 20:12:28 UTC (rev 261942)
@@ -1,7 +1,3 @@
-2020-05-19 Russell Epstein <[email protected]>
-
- Revert r261582. rdar://problem/63156090
-
2020-05-12 Alan Coon <[email protected]>
Cherry-pick r259853. rdar://problem/63156090
Modified: branches/safari-609-branch/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp (261941 => 261942)
--- branches/safari-609-branch/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp 2020-05-20 19:59:53 UTC (rev 261941)
+++ branches/safari-609-branch/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp 2020-05-20 20:12:28 UTC (rev 261942)
@@ -28,15 +28,14 @@
#if USE(LIBWEBRTC)
-#include <wtf/MonotonicTime.h>
+#include "LibWebRTCAudioFormat.h"
+#include "Logging.h"
namespace WebCore {
LibWebRTCAudioModule::LibWebRTCAudioModule()
- : m_audioTaskRunner(rtc::Thread::Create())
+ : m_queue(WorkQueue::create("WebKitWebRTCAudioModule", WorkQueue::Type::Serial, WorkQueue::QOS::UserInteractive))
{
- m_audioTaskRunner->SetName("WebKitWebRTCAudioModule", nullptr);
- m_audioTaskRunner->Start();
}
int32_t LibWebRTCAudioModule::RegisterAudioCallback(webrtc::AudioTransport* audioTransport)
@@ -45,18 +44,16 @@
return 0;
}
-void LibWebRTCAudioModule::OnMessage(rtc::Message* message)
+int32_t LibWebRTCAudioModule::StartPlayout()
{
- ASSERT_UNUSED(message, message->message_id == 1);
- StartPlayoutOnAudioThread();
-}
+ if (m_isPlaying)
+ return 0;
-int32_t LibWebRTCAudioModule::StartPlayout()
-{
- if (!m_isPlaying && m_audioTaskRunner) {
- m_audioTaskRunner->Post(RTC_FROM_HERE, this, 1);
- m_isPlaying = true;
- }
+ m_isPlaying = true;
+ m_queue->dispatch([this] {
+ m_pollingTime = MonotonicTime::now();
+ pollAudioData();
+ });
return 0;
}
@@ -68,29 +65,33 @@
}
// libwebrtc uses 10ms frames.
-const unsigned samplingRate = 48000;
-const unsigned frameLengthMs = 10;
-const unsigned samplesPerFrame = samplingRate * frameLengthMs / 1000;
+const unsigned frameLengthMs = 1000 * LibWebRTCAudioFormat::chunkSampleCount / LibWebRTCAudioFormat::sampleRate;
const unsigned pollSamples = 5;
const unsigned pollInterval = 5 * frameLengthMs;
const unsigned channels = 2;
-const unsigned bytesPerSample = 2;
-void LibWebRTCAudioModule::StartPlayoutOnAudioThread()
+void LibWebRTCAudioModule::pollAudioData()
{
- MonotonicTime startTime = MonotonicTime::now();
- while (true) {
- PollFromSource();
+ if (!m_isPlaying)
+ return;
- MonotonicTime now = MonotonicTime::now();
- double sleepFor = pollInterval - remainder((now - startTime).milliseconds(), pollInterval);
- m_audioTaskRunner->SleepMs(sleepFor);
- if (!m_isPlaying)
- return;
+ pollFromSource();
+
+ auto now = MonotonicTime::now();
+ auto delayUntilNextPolling = m_pollingTime + Seconds::fromMilliseconds(pollInterval) - now;
+ if (delayUntilNextPolling.milliseconds() < 0) {
+ callOnMainThread([timeSpent = (now - m_pollingTime).milliseconds()] {
+ RELEASE_LOG(WebRTC, "LibWebRTCAudioModule::pollAudioData, polling took too much time: %d ms", (int)timeSpent);
+ });
+ delayUntilNextPolling = 0_s;
}
+ m_pollingTime = now + delayUntilNextPolling;
+ m_queue->dispatchAfter(delayUntilNextPolling, [this] {
+ pollAudioData();
+ });
}
-void LibWebRTCAudioModule::PollFromSource()
+void LibWebRTCAudioModule::pollFromSource()
{
if (!m_audioTransport)
return;
@@ -98,8 +99,8 @@
for (unsigned i = 0; i < pollSamples; i++) {
int64_t elapsedTime = -1;
int64_t ntpTime = -1;
- char data[(bytesPerSample * channels * samplesPerFrame)];
- m_audioTransport->PullRenderData(bytesPerSample * 8, samplingRate, channels, samplesPerFrame, data, &elapsedTime, &ntpTime);
+ char data[LibWebRTCAudioFormat::sampleByteSize * channels * LibWebRTCAudioFormat::chunkSampleCount];
+ m_audioTransport->PullRenderData(LibWebRTCAudioFormat::sampleByteSize * 8, LibWebRTCAudioFormat::sampleRate, channels, LibWebRTCAudioFormat::chunkSampleCount, data, &elapsedTime, &ntpTime);
}
}
Modified: branches/safari-609-branch/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h (261941 => 261942)
--- branches/safari-609-branch/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h 2020-05-20 19:59:53 UTC (rev 261941)
+++ branches/safari-609-branch/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h 2020-05-20 20:12:28 UTC (rev 261942)
@@ -32,8 +32,8 @@
ALLOW_UNUSED_PARAMETERS_BEGIN
#include <webrtc/modules/audio_device/include/audio_device.h>
-#include <webrtc/rtc_base/message_handler.h>
-#include <webrtc/rtc_base/thread.h>
+#include <wtf/MonotonicTime.h>
+#include <wtf/WorkQueue.h>
ALLOW_UNUSED_PARAMETERS_END
@@ -40,7 +40,7 @@
namespace WebCore {
// LibWebRTCAudioModule is pulling streamed data to ensure audio data is passed to the audio track.
-class LibWebRTCAudioModule final : public webrtc::AudioDeviceModule, private rtc::MessageHandler {
+class LibWebRTCAudioModule final : public webrtc::AudioDeviceModule {
WTF_MAKE_FAST_ALLOCATED;
public:
LibWebRTCAudioModule();
@@ -54,7 +54,6 @@
void AddRef() const final { }
rtc::RefCountReleaseStatus Release() const final { return rtc::RefCountReleaseStatus::kOtherRefsRemained; }
- void OnMessage(rtc::Message*);
// webrtc::AudioDeviceModule API
int32_t StartPlayout() final;
@@ -123,14 +122,13 @@
#endif
private:
- void StartPlayoutOnAudioThread();
+ void pollAudioData();
+ void pollFromSource();
- void PollFromSource();
-
- std::unique_ptr<rtc::Thread> m_audioTaskRunner;
-
- bool m_isPlaying = false;
- webrtc::AudioTransport* m_audioTransport = nullptr;
+ Ref<WorkQueue> m_queue;
+ bool m_isPlaying { false };
+ webrtc::AudioTransport* m_audioTransport { nullptr };
+ MonotonicTime m_pollingTime;
};
} // namespace WebCore
