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/*
* Copyright (C) 2017 Igalia Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are 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.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND 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 THE COPYRIGHT
* OWNER OR 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"
#if USE(LIBWEBRTC) && USE(GSTREAMER)
#include "RealtimeIncomingAudioSourceLibWebRTC.h"
#include "LibWebRTCAudioFormat.h"
#include "gstreamer/GStreamerAudioData.h"
#include "gstreamer/GStreamerAudioStreamDescription.h"
namespace WebCore {
Ref<RealtimeIncomingAudioSource> RealtimeIncomingAudioSource::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
{
auto source = RealtimeIncomingAudioSourceLibWebRTC::create(WTFMove(audioTrack), WTFMove(audioTrackId));
source->start();
return source;
}
Ref<RealtimeIncomingAudioSourceLibWebRTC> RealtimeIncomingAudioSourceLibWebRTC::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
{
return adoptRef(*new RealtimeIncomingAudioSourceLibWebRTC(WTFMove(audioTrack), WTFMove(audioTrackId)));
}
RealtimeIncomingAudioSourceLibWebRTC::RealtimeIncomingAudioSourceLibWebRTC(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
: RealtimeIncomingAudioSource(WTFMove(audioTrack), WTFMove(audioTrackId))
{
}
void RealtimeIncomingAudioSourceLibWebRTC::OnData(const void* audioData, int, int sampleRate, size_t numberOfChannels, size_t numberOfFrames)
{
GstAudioInfo info;
GstAudioFormat format = gst_audio_format_build_integer(
LibWebRTCAudioFormat::isSigned,
LibWebRTCAudioFormat::isBigEndian ? G_BIG_ENDIAN : G_LITTLE_ENDIAN,
LibWebRTCAudioFormat::sampleSize,
LibWebRTCAudioFormat::sampleSize);
gst_audio_info_set_format(&info, format, sampleRate, numberOfChannels, NULL);
auto bufferSize = GST_AUDIO_INFO_BPF(&info) * numberOfFrames;
auto buffer = adoptGRef(gst_buffer_new_memdup(const_cast<gpointer>(audioData), bufferSize));
gst_buffer_add_audio_meta(buffer.get(), &info, numberOfFrames, nullptr);
auto caps = adoptGRef(gst_audio_info_to_caps(&info));
if (m_baseTime == MediaTime::invalidTime())
m_baseTime = MediaTime::createWithSeconds(MonotonicTime::now().secondsSinceEpoch());
MediaTime mediaTime = m_baseTime + MediaTime((m_numberOfFrames * G_USEC_PER_SEC) / sampleRate, G_USEC_PER_SEC);
GST_BUFFER_PTS(buffer.get()) = toGstUnsigned64Time(mediaTime);
auto sample = adoptGRef(gst_sample_new(buffer.get(), caps.get(), nullptr, nullptr));
GStreamerAudioData data(WTFMove(sample), info);
audioSamplesAvailable(mediaTime, data, GStreamerAudioStreamDescription(info), numberOfFrames);
m_numberOfFrames += numberOfFrames;
}
}
#endif // USE(LIBWEBRTC)
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