GStreamerAudioCapturer.cpp [plain text]
#include "config.h"
#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
#include "GStreamerAudioCapturer.h"
#include "LibWebRTCAudioFormat.h"
#include <gst/app/gstappsink.h>
namespace WebCore {
GStreamerAudioCapturer::GStreamerAudioCapturer(GStreamerCaptureDevice device)
: GStreamerCapturer(device, adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
{
}
GStreamerAudioCapturer::GStreamerAudioCapturer()
: GStreamerCapturer("appsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
{
}
GstElement* GStreamerAudioCapturer::createConverter()
{
auto converter = gst_parse_bin_from_description("audioconvert ! audioresample", TRUE, nullptr);
ASSERT(converter);
return converter;
}
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.get())
return false;
g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);
return true;
}
}
#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)