moonlight-stream/moonlight-common-c
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Add a workaround to get low bandwidth stereo working again and a cap flag to never request HQ audio

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This commit is contained in:
Cameron Gutman 2019-04-25 01:03:27 -07:00
parent d6971983aa
commit c87d4a755e
4 changed files with 23 additions and 4 deletions
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4
src/Limelight-internal.h
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@ -37,6 +37,10 @@ extern int OriginalVideoBitrate;
#define UDP_RECV_POLL_TIMEOUT_MS 100
// At this value, we will request high quality audio unless CAPABILITY_SLOW_OPUS_DECODER
// is set on the audio renderer.
#define HIGH_AUDIO_BITRATE_THRESHOLD 15000
int serviceEnetHost(ENetHost* client, ENetEvent* event, enet_uint32 timeoutMs);
int extractVersionQuadFromString(const char* string, int* quad);
int isReferenceFrameInvalidationEnabled(void);

5
src/Limelight.h
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@ -166,6 +166,11 @@ typedef struct _DECODE_UNIT {
// supports reference frame invalidation for HEVC/H.265 streams. This flag is only valid on video renderers.
#define CAPABILITY_REFERENCE_FRAME_INVALIDATION_HEVC 0x4
// If set in the audio renderer capabilities field, this flag will cause the RTSP negotiation
// to never request the "high quality" audio preset. If unset, high quality audio will be
// used with video streams above 15 Mbps.
#define CAPABILITY_SLOW_OPUS_DECODER 0x8
// If set in the video renderer capabilities field, this macro specifies that the renderer
// supports slicing to increase decoding performance. The parameter specifies the desired
// number of slices per frame. This capability is only valid on video renderers.

13
src/RtspConnection.c
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@ -430,9 +430,20 @@ static int sendVideoAnnounce(PRTSP_MESSAGE response, int* error) {
int performRtspHandshake(void) {
int ret;
// HACK: In order to get GFE to respect our request for a lower audio bitrate, we must
// fake our target address so it doesn't match any of the PC's local interfaces. It seems
// that the only way to get it to give you "low quality" stereo audio nowadays is if it
// thinks you are remote (target address != any local address).
if (OriginalVideoBitrate >= HIGH_AUDIO_BITRATE_THRESHOLD &&
(AudioCallbacks.capabilities & CAPABILITY_SLOW_OPUS_DECODER) == 0) {
addrToUrlSafeString(&RemoteAddr, urlAddr);
}
else {
strcpy(urlAddr, "0.0.0.0");
}
// Initialize global state
useEnet = (AppVersionQuad[0] >= 5) && (AppVersionQuad[0] <= 7) && (AppVersionQuad[2] < 404);
addrToUrlSafeString(&RemoteAddr, urlAddr);
sprintf(rtspTargetUrl, "rtsp%s://%s:48010", useEnet ? "ru" : "", urlAddr);
currentSeqNumber = 1;
hasSessionId = 0;

5
src/SdpGenerator.c
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@ -11,8 +11,6 @@
#define CHANNEL_MASK_STEREO 0x3
#define CHANNEL_MASK_51_SURROUND 0xFC
#define HIGH_BITRATE_THRESHOLD 15000
typedef struct _SDP_OPTION {
char name[MAX_OPTION_NAME_LEN + 1];
void* payload;
@ -362,7 +360,8 @@ static PSDP_OPTION getAttributesList(char*urlSafeAddr) {
if (AppVersionQuad[0] >= 7) {
// Decide to use HQ audio based on the original video bitrate, not the HEVC-adjusted value
if (OriginalVideoBitrate >= HIGH_BITRATE_THRESHOLD && audioChannelCount > 2) {
if (OriginalVideoBitrate >= HIGH_AUDIO_BITRATE_THRESHOLD && audioChannelCount > 2 &&
(AudioCallbacks.capabilities & CAPABILITY_SLOW_OPUS_DECODER) == 0) {
// Enable high quality mode for surround sound
err |= addAttributeString(&optionHead, "x-nv-audio.surround.AudioQuality", "1");