/*
 * 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

namespace WebCore {

#if USE(LIBWEBRTC)
static constexpr size_t s_audioCaptureSampleRate = LibWebRTCAudioFormat::sampleRate;
#else
static constexpr size_t s_audioCaptureSampleRate = 48000;
#endif

GStreamerAudioCapturer::GStreamerAudioCapturer(GStreamerCaptureDevice device)
    : GStreamerCapturer(device, adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, s_audioCaptureSampleRate, nullptr)))
{
}

GStreamerAudioCapturer::GStreamerAudioCapturer()
    : GStreamerCapturer("appsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, s_audioCaptureSampleRate, nullptr)), CaptureDevice::DeviceType::Microphone)
{
}

GstElement* GStreamerAudioCapturer::createConverter()
{
    auto* bin = gst_bin_new(nullptr);
    auto* audioconvert = gst_element_factory_make("audioconvert", nullptr);
    auto* audioresample = gst_element_factory_make("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);
        gst_bin_add(GST_BIN_CAST(bin), webrtcdsp);
        gst_element_link(webrtcdsp, audioconvert);
    }
#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.get())
        return false;

    g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);

    return true;
}

} // namespace WebCore

#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)