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/*
* Copyright (C) 2018 Metrological Group B.V.
* Copyright (C) 2020 Igalia S.L.
* Author: Thibault Saunier <tsaunier@igalia.com>
* Author: Alejandro G. Castro <alex@igalia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public License
* aint with this library; see the file COPYING.LIB. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include "config.h"
#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
#include "GStreamerAudioCapturer.h"
#if USE(LIBWEBRTC)
#include "LibWebRTCAudioFormat.h"
#endif
#include <gst/app/gstappsink.h>
namespace WebCore {
#if USE(LIBWEBRTC)
static constexpr size_t s_audioCaptureSampleRate = LibWebRTCAudioFormat::sampleRate;
#else
static constexpr size_t s_audioCaptureSampleRate = 48000;
#endif
GST_DEBUG_CATEGORY(webkit_audio_capturer_debug);
#define GST_CAT_DEFAULT webkit_audio_capturer_debug
static void initializeAudioCapturerDebugCategory()
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_audio_capturer_debug, "webkitaudiocapturer", 0, "WebKit Audio Capturer");
});
}
GStreamerAudioCapturer::GStreamerAudioCapturer(GStreamerCaptureDevice&& device)
: GStreamerCapturer(WTFMove(device), adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, s_audioCaptureSampleRate, nullptr)))
{
initializeAudioCapturerDebugCategory();
}
GStreamerAudioCapturer::GStreamerAudioCapturer()
: GStreamerCapturer("appsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, s_audioCaptureSampleRate, nullptr)), CaptureDevice::DeviceType::Microphone)
{
initializeAudioCapturerDebugCategory();
}
void GStreamerAudioCapturer::setSinkAudioCallback(SinkAudioDataCallback&& callback)
{
if (m_sinkAudioDataCallback.first)
g_signal_handler_disconnect(sink(), m_sinkAudioDataCallback.first);
m_sinkAudioDataCallback.second = WTFMove(callback);
m_sinkAudioDataCallback.first = g_signal_connect_swapped(sink(), "new-sample", G_CALLBACK(+[](GStreamerAudioCapturer* capturer, GstElement* sink) -> GstFlowReturn {
auto gstSample = adoptGRef(gst_app_sink_pull_sample(GST_APP_SINK(sink)));
auto presentationTime = fromGstClockTime(GST_BUFFER_PTS(gst_sample_get_buffer(gstSample.get())));
capturer->m_sinkAudioDataCallback.second(WTFMove(gstSample), WTFMove(presentationTime));
return GST_FLOW_OK;
}), this);
}
GstElement* GStreamerAudioCapturer::createConverter()
{
auto* bin = gst_bin_new(nullptr);
auto* audioconvert = makeGStreamerElement("audioconvert", nullptr);
auto* audioresample = makeGStreamerElement("audioresample", nullptr);
gst_bin_add_many(GST_BIN_CAST(bin), audioconvert, audioresample, nullptr);
gst_element_link(audioconvert, audioresample);
#if USE(GSTREAMER_WEBRTC)
if (auto* webrtcdsp = makeGStreamerElement("webrtcdsp", nullptr)) {
g_object_set(webrtcdsp, "echo-cancel", FALSE, "voice-detection", TRUE, nullptr);
auto* audioconvert2 = makeGStreamerElement("audioconvert", nullptr);
auto* audioresample2 = makeGStreamerElement("audioresample", nullptr);
gst_bin_add_many(GST_BIN_CAST(bin), audioconvert2, audioresample2, webrtcdsp, nullptr);
gst_element_link_many(audioconvert2, audioresample2, webrtcdsp, audioconvert, nullptr);
}
#endif
if (auto pad = adoptGRef(gst_bin_find_unlinked_pad(GST_BIN_CAST(bin), GST_PAD_SRC)))
gst_element_add_pad(GST_ELEMENT_CAST(bin), gst_ghost_pad_new("src", pad.get()));
if (auto pad = adoptGRef(gst_bin_find_unlinked_pad(GST_BIN_CAST(bin), GST_PAD_SINK)))
gst_element_add_pad(GST_ELEMENT_CAST(bin), gst_ghost_pad_new("sink", pad.get()));
return bin;
}
bool GStreamerAudioCapturer::setSampleRate(int sampleRate)
{
if (sampleRate <= 0) {
GST_INFO_OBJECT(m_pipeline.get(), "Not forcing sample rate");
return false;
}
GST_INFO_OBJECT(m_pipeline.get(), "Setting SampleRate %d", sampleRate);
m_caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, sampleRate, nullptr));
if (!m_capsfilter)
return false;
g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);
return true;
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
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