blob: 47196c41fce8efcbadddcf2c9832eb5c59f30a21 [file] [log] [blame]
/*
* Copyright (C) 2017-2019 Apple Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted, provided that the following conditions
* are required to be met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. Neither the name of Apple Inc. nor the names of
* its contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
* ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
* CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#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_sinks.isEmpty());
stop();
}
void RealtimeOutgoingAudioSource::observeSource()
{
m_audioSource->addObserver(*this);
initializeConverter();
}
void RealtimeOutgoingAudioSource::unobserveSource()
{
m_audioSource->removeObserver(*this);
}
bool RealtimeOutgoingAudioSource::setSource(Ref<MediaStreamTrackPrivate>&& newSource)
{
ALWAYS_LOG("Changing source to ", newSource->logIdentifier());
auto locker = holdLock(m_sinksLock);
bool hasSinks = !m_sinks.isEmpty();
if (hasSinks)
unobserveSource();
m_audioSource = WTFMove(newSource);
if (hasSinks)
observeSource();
sourceUpdated();
return true;
}
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);
if (!m_sinks.add(sink) || m_sinks.size() != 1)
return;
}
callOnMainThread([protectedThis = makeRef(*this)]() {
protectedThis->observeSource();
});
}
void RealtimeOutgoingAudioSource::RemoveSink(webrtc::AudioTrackSinkInterface* sink)
{
{
auto locker = holdLock(m_sinksLock);
if (!m_sinks.remove(sink) || !m_sinks.isEmpty())
return;
}
unobserveSource();
}
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
} // namespace WebCore
#endif // USE(LIBWEBRTC)