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

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This commit is contained in:
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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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);
}
}