Files
AirplayServer/renderers/audio_renderer_gstreamer.c

121 lines
4.0 KiB
C

/**
* 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 <stdlib.h>
#include <assert.h>
#include <stdio.h>
#include <string.h>
#include <stdbool.h>
#include <unistd.h>
#include <math.h>
#include <gst/app/gstappsrc.h>
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);
}
}