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/*
* Copyright (C) 2018 Konsulko Group
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#include <stdio.h>
#include <stdlib.h>
#include <stdbool.h>
#include <string.h>
#include <errno.h>
#include <gst/gst.h>
#include "radio_output.h"
#include "rtl_fm.h"
// Output buffer
static unsigned int extra;
static int16_t extra_buf[1];
static unsigned char *output_buf;
// GStreamer state
static GstElement *pipeline, *appsrc;
static bool running;
int radio_output_open()
{
unsigned int rate = 24000;
GstElement *queue, *convert, *sink, *resample;
char *p;
if(pipeline)
return 0;
// Initialize GStreamer
#ifdef DEBUG
unsigned int argc = 2;
char **argv = malloc(2 * sizeof(char*));
argv[0] = strdup("test");
argv[1] = strdup("--gst-debug-level=5");
gst_init(&argc, &argv);
#else
gst_init(NULL, NULL);
#endif
// Setup pipeline
// NOTE: With our use of the simple buffer pushing mode, there seems to
// be no need for a mainloop, so currently not instantiating one.
pipeline = gst_pipeline_new("pipeline");
appsrc = gst_element_factory_make("appsrc", "source");
queue = gst_element_factory_make("queue", "queue");
convert = gst_element_factory_make("audioconvert", "convert");
resample = gst_element_factory_make("audioresample", "resample");
sink = gst_element_factory_make("pwaudiosink", "sink");
if(!(pipeline && appsrc && queue && convert && resample && sink)) {
fprintf(stderr, "pipeline element construction failed!\n");
}
g_object_set(G_OBJECT(appsrc), "caps",
gst_caps_new_simple("audio/x-raw",
"format", G_TYPE_STRING, "S16LE",
"rate", G_TYPE_INT, rate,
"channels", G_TYPE_INT, 2,
"layout", G_TYPE_STRING, "interleaved",
"channel-mask", G_TYPE_UINT64, 3,
NULL), NULL);
gst_util_set_object_arg(sink, "stream-properties", "p,media.role=Multimedia");
if((p = getenv("RADIO_OUTPUT"))) {
fprintf(stderr, "Using output device %s\n", p);
g_object_set(sink, "device", p, NULL);
}
gst_bin_add_many(GST_BIN(pipeline), appsrc, queue, convert, resample, sink, NULL);
gst_element_link_many(appsrc, queue, convert, resample, sink, NULL);
//gst_bin_add_many(GST_BIN(pipeline), appsrc, convert, resample, sink, NULL);
//gst_element_link_many(appsrc, convert, resample, sink, NULL);
// Set up appsrc
// NOTE: Radio seems like it matches the use case the "is-live" property
// is for, but setting it seems to require a lot more work with
// respect to latency settings to make the pipeline work smoothly.
// For now, leave it unset since the result seems to work
// reasonably well.
g_object_set(G_OBJECT(appsrc),
"stream-type", 0,
"format", GST_FORMAT_TIME,
NULL);
// Start pipeline in paused state
gst_element_set_state(pipeline, GST_STATE_PAUSED);
// Set up output buffer
extra = 0;
output_buf = malloc(sizeof(unsigned char) * RTL_FM_MAXIMUM_BUF_LENGTH);
return 0;
}
int radio_output_start(void)
{
int rc = 0;
if(!pipeline) {
rc = radio_output_open();
if(rc)
return rc;
}
// Start pipeline
running = true;
gst_element_set_state(pipeline, GST_STATE_PLAYING);
return rc;
}
void radio_output_stop(void)
{
GstEvent *event;
if(pipeline && running) {
// Stop pipeline
running = false;
gst_element_set_state(pipeline, GST_STATE_PAUSED);
// Flush pipeline
// This seems required to avoid stutters on starts after a stop
event = gst_event_new_flush_start();
gst_element_send_event(GST_ELEMENT(pipeline), event);
event = gst_event_new_flush_stop(TRUE);
gst_element_send_event(GST_ELEMENT(pipeline), event);
}
}
void radio_output_suspend(int state)
{
// Placeholder
}
void radio_output_close(void)
{
radio_output_stop();
if(pipeline) {
// Tear down pipeline
gst_element_set_state(pipeline, GST_STATE_NULL);
gst_object_unref(GST_OBJECT(pipeline));
pipeline = NULL;
running = false;
}
free(output_buf);
output_buf = NULL;
}
ssize_t radio_output_write(void *buf, int len)
{
ssize_t rc = -EINVAL;
size_t n = len;
int samples = len / 2;
unsigned char *p;
GstBuffer *buffer;
GstFlowReturn ret;
if(!(pipeline && buf)) {
return rc;
}
// Don't bother pushing samples if output hasn't started
if(!running)
return 0;
/*
* Handle the rtl_fm code giving us an odd number of samples.
* This extra buffer copying approach is not particularly efficient,
* but works for now. It looks feasible to hack in something in the
* demod and output thread routines in rtl_fm.c to handle it there
* if more performance is required.
*/
p = output_buf;
if(extra) {
memcpy(output_buf, extra_buf, sizeof(int16_t));
if((extra + samples) % 2) {
// We still have an extra sample, n remains the same, store the extra
memcpy(output_buf + sizeof(int16_t), buf, n - 2);
memcpy(extra_buf, ((unsigned char*) buf) + n - 2, sizeof(int16_t));
} else {
// We have an even number of samples, no extra
memcpy(output_buf + sizeof(int16_t), buf, n);
n += 2;
extra = 0;
}
} else if(samples % 2) {
// We have an extra sample, store it, and decrease n
n -= 2;
memcpy(output_buf + sizeof(int16_t), buf, n);
memcpy(extra_buf, ((unsigned char*) buf) + n, sizeof(int16_t));
extra = 1;
} else {
p = buf;
}
// Push buffer into pipeline
buffer = gst_buffer_new_allocate(NULL, n, NULL);
gst_buffer_fill(buffer, 0, p, n);
g_signal_emit_by_name(appsrc, "push-buffer", buffer, &ret);
gst_buffer_unref(buffer);
rc = n;
return rc;
}
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