| /* |
| * 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; |
| gpointer bufferData = g_malloc(bufferSize); |
| if (muted()) |
| gst_audio_format_fill_silence(info.finfo, bufferData, bufferSize); |
| else |
| memcpy(bufferData, audioData, bufferSize); |
| |
| auto buffer = adoptGRef(gst_buffer_new_wrapped(bufferData, bufferSize)); |
| GRefPtr<GstCaps> caps = adoptGRef(gst_audio_info_to_caps(&info)); |
| auto sample = adoptGRef(gst_sample_new(buffer.get(), caps.get(), nullptr, nullptr)); |
| auto data(std::unique_ptr<GStreamerAudioData>(new GStreamerAudioData(WTFMove(sample), info))); |
| |
| auto mediaTime = MediaTime((m_numberOfFrames * G_USEC_PER_SEC) / sampleRate, G_USEC_PER_SEC); |
| audioSamplesAvailable(mediaTime, *data.get(), GStreamerAudioStreamDescription(info), numberOfFrames); |
| |
| m_numberOfFrames += numberOfFrames; |
| } |
| } |
| |
| #endif // USE(LIBWEBRTC) |