/** * RPiPlay - An open-source AirPlay mirroring server for Raspberry Pi * Copyright (C) 2019 Florian Draschbacher * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 3 of the License, or * (at your option) any later version. * * This program 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software Foundation, * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "audio_renderer.h" #include #include #include #include #include #include #include #include struct audio_renderer_s { logger_t *logger; GstElement *appsrc; GstElement *pipeline; GstElement *volume; }; audio_renderer_t *audio_renderer_init(logger_t *logger, video_renderer_t *video_renderer, audio_device_t device, bool low_latency) { audio_renderer_t *renderer; GError *error = NULL; renderer = calloc(1, sizeof(audio_renderer_t)); if (!renderer) { return NULL; } renderer->logger = logger; renderer->pipeline = gst_parse_launch("appsrc name=audio_source stream-type=0 format=GST_FORMAT_TIME is-live=true ! queue ! decodebin !" "audioconvert ! volume name=volume ! level ! autoaudiosink sync=false", &error); g_assert (renderer->pipeline); renderer->appsrc = gst_bin_get_by_name (GST_BIN (renderer->pipeline), "audio_source"); renderer->volume = gst_bin_get_by_name (GST_BIN (renderer->pipeline), "volume"); gchar eld_conf[] = { 0xF8, 0xE8, 0x50, 0x00 }; GstBuffer *codec_data = gst_buffer_new_and_alloc(sizeof(eld_conf)); GstMapInfo map; gst_buffer_map (codec_data, &map, GST_MAP_WRITE); memset (map.data, eld_conf[0], map.size); memset (map.data+1, eld_conf[1], map.size); memset (map.data+2, eld_conf[2], map.size); memset (map.data+3, eld_conf[3], map.size); GstCaps *caps = gst_caps_new_simple ("audio/mpeg", "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, "mpegversion", G_TYPE_INT, 4, "stream-format", G_TYPE_STRING, "raw", "codec_data", GST_TYPE_BUFFER, codec_data, NULL); g_object_set(renderer->appsrc, "caps", caps, NULL); gst_caps_unref(caps); gst_buffer_unmap (codec_data, &map); gst_buffer_unref (codec_data); return renderer; } void audio_renderer_start(audio_renderer_t *renderer) { //g_signal_connect( renderer->pipeline, "deep-notify", G_CALLBACK(gst_object_default_deep_notify ), NULL ); gst_element_set_state (renderer->pipeline, GST_STATE_PLAYING); } void audio_renderer_render_buffer(audio_renderer_t *renderer, raop_ntp_t *ntp, unsigned char* data, int data_len, uint64_t pts) { GstBuffer *buffer; if (data_len == 0) return; buffer = gst_buffer_new_and_alloc(data_len); assert(buffer != NULL); GST_BUFFER_DTS(buffer) = (GstClockTime)pts; gst_buffer_fill(buffer, 0, data, data_len); gst_app_src_push_buffer(GST_APP_SRC(renderer->appsrc), buffer); } void audio_renderer_set_volume(audio_renderer_t *renderer, float volume) { float avol; if (fabs(volume) < 28) { avol=floorf(((28-fabs(volume))/28)*10)/10; g_object_set(renderer->volume, "volume", avol, NULL); } } void audio_renderer_flush(audio_renderer_t *renderer) { } void audio_renderer_destroy(audio_renderer_t *renderer) { gst_app_src_end_of_stream (GST_APP_SRC(renderer->appsrc)); gst_element_set_state (renderer->pipeline, GST_STATE_NULL); gst_object_unref (renderer->pipeline); gst_object_unref (renderer->appsrc); gst_object_unref (renderer->volume); if (renderer) { free(renderer); } }