RealtimeOutgoingAudioSource.cpp [plain text]
#include "config.h"
#include "RealtimeOutgoingAudioSource.h"
#if USE(LIBWEBRTC)
#include "LibWebRTCAudioFormat.h"
#include "LibWebRTCProvider.h"
#include "Logging.h"
#include <wtf/CryptographicallyRandomNumber.h>
namespace WebCore {
RealtimeOutgoingAudioSource::RealtimeOutgoingAudioSource(Ref<MediaStreamTrackPrivate>&& source)
: m_audioSource(WTFMove(source))
{
}
RealtimeOutgoingAudioSource::~RealtimeOutgoingAudioSource()
{
ASSERT(!m_audioSource->hasObserver(*this));
#if !ASSERT_DISABLED
auto locker = holdLock(m_sinksLock);
#endif
ASSERT(m_sinks.isEmpty());
stop();
}
void RealtimeOutgoingAudioSource::observeSource()
{
ASSERT(!m_audioSource->hasObserver(*this));
m_audioSource->addObserver(*this);
initializeConverter();
}
void RealtimeOutgoingAudioSource::unobserveSource()
{
m_audioSource->removeObserver(*this);
}
void RealtimeOutgoingAudioSource::setSource(Ref<MediaStreamTrackPrivate>&& newSource)
{
ALWAYS_LOG("Changing source to ", newSource->logIdentifier());
ASSERT(!m_audioSource->hasObserver(*this));
m_audioSource = WTFMove(newSource);
sourceUpdated();
}
void RealtimeOutgoingAudioSource::initializeConverter()
{
m_muted = m_audioSource->muted();
m_enabled = m_audioSource->enabled();
}
void RealtimeOutgoingAudioSource::sourceMutedChanged()
{
m_muted = m_audioSource->muted();
}
void RealtimeOutgoingAudioSource::sourceEnabledChanged()
{
m_enabled = m_audioSource->enabled();
}
void RealtimeOutgoingAudioSource::AddSink(webrtc::AudioTrackSinkInterface* sink)
{
auto locker = holdLock(m_sinksLock);
m_sinks.add(sink);
}
void RealtimeOutgoingAudioSource::RemoveSink(webrtc::AudioTrackSinkInterface* sink)
{
auto locker = holdLock(m_sinksLock);
m_sinks.remove(sink);
}
void RealtimeOutgoingAudioSource::sendAudioFrames(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames)
{
#if !RELEASE_LOG_DISABLED
if (!(++m_chunksSent % 200))
ALWAYS_LOG(LOGIDENTIFIER, "chunk ", m_chunksSent);
#endif
auto locker = holdLock(m_sinksLock);
for (auto sink : m_sinks)
sink->OnData(audioData, bitsPerSample, sampleRate, numberOfChannels, numberOfFrames);
}
#if !RELEASE_LOG_DISABLED
WTFLogChannel& RealtimeOutgoingAudioSource::logChannel() const
{
return LogWebRTC;
}
#endif
}
#endif // USE(LIBWEBRTC)