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UxPlay 1.39 add support for Apple Lossless (ALAC) audio-only AirPlay

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fduncanh
2021-11-06 06:22:58 -04:00
parent ab110a7ce8
commit 7bf22cf30c
5 changed files with 70 additions and 47 deletions
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13
README.md
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@@ -1,4 +1,5 @@
# UxPlay 1.38
# UxPlay 1.39
This project is a GPLv3 unix AirPlay server which now also works on macOS.
Its main use is to act like an AppleTV for screen-mirroring (with audio) of iOS/macOS clients
@@ -15,14 +16,18 @@ On Linux and BSD Unix servers, this is usually provided by [Avahi](https://www.a
through the avahi-daemon service, and is included in most Linux distributions (this
service can also be provided by macOS, iOS or Windows servers).
_Note: UxPlay currently only works using the AirPlay screen-mirroring protocol (which streams audio in **AAC** format)
_New: UxPlay 1.39 now also supports the Airplay audio-only protocol as well as AirPlay Mirror protocal, and can play Apple Lossless (ALAC)
44100/16/2 audio streamed from Apple Music on the client in 2-channel stereo without video (the accompanying coverart and metadata is received by the server,
but not displayed)._
currently only works using the AirPlay screen-mirroring protocol (which streams audio in **AAC** format)
(not the AirPlay audio-only streaming protocol which uses lossless **ALAC** format)
but the **uxplay -vs 0** option streams AAC-ELD audio using screen-mirroring without displaying the client's screen.
If the client streams audio using AirPlay as opposed to AirPlay screen-mirroring, non-AAC data can be seen to be received and processed by UxPlay, and then
input into the GStreamer audio-rendering pipeline, but does not get rendered into audio output. If someone can adapt the GStreamer audio
pipeline to also render these Airplay audio streams, such an enhancement of UxPlay would be welcome as a Pull Request!_
UxPlay 1.38 is based on https://github.com/FD-/RPiPlay, with GStreamer integration from
UxPlay 1.39 is based on https://github.com/FD-/RPiPlay, with GStreamer integration from
https://github.com/antimof/UxPlay.
(UxPlay only uses GStreamer, and does not contain the alternative Raspberry-Pi-specific
audio and video renderers also found in RPiPlay.)
@@ -318,6 +323,8 @@ Also: image transforms that had been added to RPiPlay have been ported to UxPlay
# ChangeLog
1.39 2021-11-06 Added support for Apple Lossless (ALAC) audio streams.
1.38 2021-10-8 Add -as _audiosink_ option to allow user to choose the GStreamer audiosink.
1.37 2021-09-29 Append "@hostname" to AirPlay Server name, where "hostname" is the name of the

2
lib/raop.h
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@@ -36,13 +36,13 @@ struct raop_callbacks_s {
void (*audio_process)(void *cls, raop_ntp_t *ntp, aac_decode_struct *data);
void (*video_process)(void *cls, raop_ntp_t *ntp, h264_decode_struct *data);
void (*audio_setup)(void *cls, unsigned char *compression_type);
/* Optional but recommended callback functions */
void (*conn_init)(void *cls);
void (*conn_destroy)(void *cls);
void (*audio_flush)(void *cls);
void (*video_flush)(void *cls);
void (*audio_compression_type)(void *cls, unsigned char *compression_type);
void (*audio_set_volume)(void *cls, float volume);
void (*audio_set_metadata)(void *cls, const void *buffer, int buflen);
void (*audio_set_coverart)(void *cls, const void *buffer, int buflen);

14
lib/raop_handlers.h
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@@ -433,12 +433,10 @@ raop_handler_setup(raop_conn_t *conn,
// Audio
unsigned short cport = conn->raop->control_lport, dport = conn->raop->data_lport;
uint64_t ct;
if (conn->raop->callbacks.audio_compression_type) {
/* get audio compression type */
plist_t req_stream_ct_node = plist_dict_get_item(req_stream_node, "ct");
plist_get_uint_val(req_stream_ct_node, &ct);
conn->raop->callbacks.audio_compression_type(conn->raop->callbacks.cls, (unsigned char*) &ct);
}
/* get audio compression type */
plist_t req_stream_ct_node = plist_dict_get_item(req_stream_node, "ct");
plist_get_uint_val(req_stream_ct_node, &ct);
conn->raop->callbacks.audio_setup(conn->raop->callbacks.cls, (unsigned char*) &ct);
if (conn->raop_rtp) {
raop_rtp_start_audio(conn->raop_rtp, use_udp, remote_cport, &cport, &dport);
@@ -550,14 +548,14 @@ raop_handler_set_parameter(raop_conn_t *conn,
}
free(datastr);
} else if (!strcmp(content_type, "image/jpeg") || !strcmp(content_type, "image/png")) {
logger_log(conn->raop->logger, LOGGER_INFO, "Got image data of %d bytes", datalen);
logger_log(conn->raop->logger, LOGGER_DEBUG, "Got image data of %d bytes", datalen);
if (conn->raop_rtp) {
raop_rtp_set_coverart(conn->raop_rtp, data, datalen);
} else {
logger_log(conn->raop->logger, LOGGER_WARNING, "RAOP not initialized at SET_PARAMETER coverart");
}
} else if (!strcmp(content_type, "application/x-dmap-tagged")) {
logger_log(conn->raop->logger, LOGGER_INFO, "Got metadata of %d bytes", datalen);
logger_log(conn->raop->logger, LOGGER_DEBUG, "Got metadata of %d bytes", datalen);
if (conn->raop_rtp) {
raop_rtp_set_metadata(conn->raop_rtp, data, datalen);
} else {

33
renderers/audio_renderer_gstreamer.c
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@@ -34,7 +34,7 @@ static const char alac[] = "audio/x-alac,mpegversion=(int)4,channnels=(int)2,rat
/* ct = 4; codec_data from MPEG v4 ISO 14996-3 Section 1.6.2.1: AAC-LC 44100/2 spf = 1024 */
static const char aac_lc[] ="audio/mpeg,mpegversion=(int)4,channnels=(int)2,rate=(int)44100,stream-format=raw,codec_data=(buffer)1210";
/* ct = 8; codec_data from MPEG v4 ISO 14996-3 Section 1.6.2.1: AAC_ELD 44100/2 spf = 460 */
/* ct = 8; codec_data from MPEG v4 ISO 14996-3 Section 1.6.2.1: AAC_ELD 44100/2 spf = 480 */
static const char aac_eld[] ="audio/mpeg,mpegversion=(int)4,channnels=(int)2,rate=(int)44100,stream-format=raw,codec_data=(buffer)f8e85000";
struct audio_renderer_s {
@@ -71,9 +71,6 @@ audio_renderer_t *audio_renderer_init(logger_t *logger, unsigned char *compressi
GError *error = NULL;
GstCaps *caps = NULL;
switch (*compression_type) {
case 1: /* uncompressed PCM */
case 2: /* Apple lossless ALAC */
@@ -127,7 +124,7 @@ audio_renderer_t *audio_renderer_init(logger_t *logger, unsigned char *compressi
g_object_set(renderer->appsrc, "caps", caps, NULL);
gst_caps_unref(caps);
return renderer;
}
@@ -142,6 +139,12 @@ void audio_renderer_render_buffer(audio_renderer_t *renderer, raop_ntp_t *ntp, u
if (data_len == 0) return;
/* all audio received seems to be either ct = 8 (AAC_ELD 44100/2 spf 460 ) AirPlay Mirror protocol */
/* or ct = 2 (ALAC 44100/16/2 spf 352) AirPlay protocol */
/* first byte data[0] of ALAC frame is 0x20, first byte of AAC_ELD is 0x8d or 0x8e, AAC_LC is 0xff (ADTS) */
/* GStreamer caps_filter could be used here to switch the appsrc caps between aac_eld and alac */
/* depending on the initial byte of the buffer, with a pipeline using decodebin */
buffer = gst_buffer_new_and_alloc(data_len);
assert(buffer != NULL);
GST_BUFFER_DTS(buffer) = (GstClockTime)pts;
@@ -162,14 +165,18 @@ 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_object_unref (renderer->appsrc);
gst_element_set_state (renderer->pipeline, GST_STATE_NULL);
gst_object_unref (renderer->pipeline);
gst_object_unref (renderer->volume);
if (renderer) {
if(renderer) {
if(renderer->appsrc) {
gst_app_src_end_of_stream (GST_APP_SRC(renderer->appsrc));
gst_object_unref (renderer->appsrc);
}
if (renderer->pipeline) {
gst_element_set_state (renderer->pipeline, GST_STATE_NULL);
gst_object_unref (renderer->pipeline);
}
if(renderer->volume) {
gst_object_unref (renderer->volume);
}
free(renderer);
}
}

55
uxplay.cpp
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@@ -44,7 +44,7 @@
static int start_server (std::vector<char> hw_addr, std::string name, unsigned short display[5],
unsigned short tcp[3], unsigned short udp[3], videoflip_t videoflip[2],
bool use_audio, bool debug_log, std::string videosink, std::string audiosink);
bool debug_log, std::string videosink);
static int stop_server ();
@@ -60,7 +60,9 @@ static bool connections_stopped = false;
static unsigned int server_timeout = 0;
static unsigned int counter;
static bool use_video = true;
static unsigned char compression_type = 8;
static unsigned char compression_type = 0;
static std::string audiosink = "autoaudiosink";
static bool use_audio = true;
gboolean connection_callback (gpointer loop){
@@ -294,13 +296,11 @@ static void append_hostname(std::string &server_name) {
int main (int argc, char *argv[]) {
std::string server_name = DEFAULT_NAME;
std::vector<char> server_hw_addr;
bool use_audio = true;
bool use_random_hw_addr = false;
bool debug_log = DEFAULT_DEBUG_LOG;
unsigned short display[5] = {0}, tcp[3] = {0}, udp[3] = {0};
videoflip_t videoflip[2] = { NONE , NONE };
std::string videosink = "autovideosink";
std::string audiosink = "autoaudiosink";
#ifdef SUPPRESS_AVAHI_COMPAT_WARNING
// suppress avahi_compat nag message. avahi emits a "nag" warning (once)
@@ -392,6 +392,12 @@ int main (int argc, char *argv[]) {
}
}
if(audiosink == "0") {
use_audio = false;
}
if(!use_audio) LOGI("audio_disabled");
if (udp[0]) LOGI("using network ports UDP %d %d %d TCP %d %d %d\n",
udp[0],udp[1], udp[2], tcp[0], tcp[1], tcp[2]);
@@ -408,9 +414,10 @@ int main (int argc, char *argv[]) {
append_hostname(server_name);
relaunch:
compression_type = 0;
connections_stopped = false;
if (start_server(server_hw_addr, server_name, display, tcp, udp,
videoflip,use_audio, debug_log, videosink, audiosink)) {
videoflip, debug_log, videosink)) {
return 1;
}
@@ -466,7 +473,23 @@ extern "C" void audio_set_volume (void *cls, float volume) {
}
}
extern "C" void audio_compression_type (void *cls, unsigned char *ct) {
extern "C" void audio_setup (void *cls, unsigned char *ct) {
if(use_audio) {
LOGI("new audio compression type %d (was %d)",*ct, compression_type);
if (*ct != compression_type) {
if (compression_type && audio_renderer) {
audio_renderer_destroy(audio_renderer);
LOGD("previous audio_renderer destroyed");
}
compression_type = *ct;
audio_renderer = audio_renderer_init(render_logger, &compression_type, audiosink.c_str());
if (audio_renderer) {
audio_renderer_start(audio_renderer);
} else {
LOGW("could not init audio_renderer");
}
}
}
}
extern "C" void log_callback (void *cls, int level, const char *msg) {
@@ -494,8 +517,8 @@ extern "C" void log_callback (void *cls, int level, const char *msg) {
}
int start_server (std::vector<char> hw_addr, std::string name, unsigned short display[5],
unsigned short tcp[3], unsigned short udp[3], videoflip_t videoflip[2],
bool use_audio, bool debug_log, std::string videosink, std::string audiosink) {
unsigned short tcp[3], unsigned short udp[3], videoflip_t videoflip[2],
bool debug_log, std::string videosink) {
raop_callbacks_t raop_cbs;
memset(&raop_cbs, 0, sizeof(raop_cbs));
raop_cbs.conn_init = conn_init;
@@ -505,7 +528,7 @@ int start_server (std::vector<char> hw_addr, std::string name, unsigned short di
raop_cbs.audio_flush = audio_flush;
raop_cbs.video_flush = video_flush;
raop_cbs.audio_set_volume = audio_set_volume;
raop_cbs.audio_compression_type = audio_compression_type;
raop_cbs.audio_setup = audio_setup;
raop = raop_init(10, &raop_cbs);
if (raop == NULL) {
@@ -520,9 +543,6 @@ int start_server (std::vector<char> hw_addr, std::string name, unsigned short di
use_video = false;
display[3] = 1; /* set fps to 1 frame per sec when no video will be shown */
}
if(audiosink == "0") {
use_audio = false;
}
raop_set_display(raop, display[0], display[1], display[2], display[3], display[4]);
@@ -548,17 +568,7 @@ int start_server (std::vector<char> hw_addr, std::string name, unsigned short di
return -1;
}
if (! use_audio) {
LOGI("Audio disabled");
} else if ((audio_renderer = audio_renderer_init(render_logger, &compression_type, audiosink.c_str())) ==
NULL) {
LOGE("Could not init audio renderer");
stop_server();
return -1;
}
if (use_video && video_renderer) video_renderer_start(video_renderer);
if (audio_renderer) audio_renderer_start(audio_renderer);
unsigned short port = raop_get_port(raop);
raop_start(raop, &port);
@@ -594,6 +604,7 @@ int stop_server () {
}
if (audio_renderer) audio_renderer_destroy(audio_renderer);
if (video_renderer) video_renderer_destroy(video_renderer);
compression_type = 0;
if (render_logger) logger_destroy(render_logger);
return 0;
}