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Rework audio bitrate and quality handling (#642)

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Cameron Gutman 2022-12-28 08:30:51 -06:00 committed by GitHub
parent c7fe8f65bd
commit f4a48f44e4
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GPG Key ID: 4AEE18F83AFDEB23
3 changed files with 35 additions and 21 deletions
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41
src/audio.cpp
View File

@ -39,6 +39,15 @@ opus_stream_config_t stream_configs[MAX_STREAM_CONFIG] {
1,
1,
platf::speaker::map_stereo,
96000,
},
{
SAMPLE_RATE,
2,
1,
1,
platf::speaker::map_stereo,
512000,
},
{
SAMPLE_RATE,
@ -46,6 +55,7 @@ opus_stream_config_t stream_configs[MAX_STREAM_CONFIG] {
4,
2,
platf::speaker::map_surround51,
256000,
},
{
SAMPLE_RATE,
@ -53,6 +63,7 @@ opus_stream_config_t stream_configs[MAX_STREAM_CONFIG] {
6,
0,
platf::speaker::map_surround51,
1536000,
},
{
SAMPLE_RATE,
@ -60,6 +71,7 @@ opus_stream_config_t stream_configs[MAX_STREAM_CONFIG] {
5,
3,
platf::speaker::map_surround71,
450000,
},
{
SAMPLE_RATE,
@ -67,6 +79,7 @@ opus_stream_config_t stream_configs[MAX_STREAM_CONFIG] {
8,
0,
platf::speaker::map_surround71,
2048000,
},
};
@ -74,9 +87,7 @@ auto control_shared = safe::make_shared<audio_ctx_t>(start_audio_control, stop_a
void encodeThread(sample_queue_t samples, config_t config, void *channel_data) {
auto packets = mail::man->queue<packet_t>(mail::audio_packets);
// FIXME: Pick correct opus_stream_config_t based on config.channels
auto stream = &stream_configs[map_stream(config.channels, config.flags[config_t::HIGH_QUALITY])];
auto stream = &stream_configs[map_stream(config.channels, config.flags[config_t::HIGH_QUALITY])];
opus_t opus { opus_multistream_encoder_create(
stream->sampleRate,
@ -84,17 +95,15 @@ void encodeThread(sample_queue_t samples, config_t config, void *channel_data) {
stream->streams,
stream->coupledStreams,
stream->mapping,
OPUS_APPLICATION_AUDIO,
OPUS_APPLICATION_RESTRICTED_LOWDELAY,
nullptr) };
// For some reason, audio is crackling when the encoder is set to constant bitstream.
// We simulate a constant bitstream with OPUS_SET_BITRATE(OPUS_BITRATE_MAX) -->
// which tries to occupy as much space as possible in the packet
opus_multistream_encoder_ctl(opus.get(), OPUS_SET_BITRATE(OPUS_BITRATE_MAX));
opus_multistream_encoder_ctl(opus.get(), OPUS_SET_BITRATE(stream->bitrate));
opus_multistream_encoder_ctl(opus.get(), OPUS_SET_VBR(0));
auto frame_size = config.packetDuration * stream->sampleRate / 1000;
while(auto sample = samples->pop()) {
buffer_t packet { 1400 }; // 1KB
buffer_t packet { 1400 };
int bytes = opus_multistream_encode(opus.get(), sample->data(), frame_size, std::begin(packet), packet.size());
if(bytes < 0) {
@ -104,14 +113,6 @@ void encodeThread(sample_queue_t samples, config_t config, void *channel_data) {
return;
}
// Even with OPUS_SET_BITRATE(OPUS_BITRATE_MAX), silent packets are smaller than the rest
// Drop silent packets to ensure Moonlight won't complain
// A packet size of 128 seems a reasonable enough threshold
if(bytes < 128) {
BOOST_LOG(verbose) << "Dropped silent packet"sv;
continue;
}
packet.fake_resize(bytes);
packets->raise(channel_data, std::move(packet));
}
@ -119,9 +120,7 @@ void encodeThread(sample_queue_t samples, config_t config, void *channel_data) {
void capture(safe::mail_t mail, config_t config, void *channel_data) {
auto shutdown_event = mail->event<bool>(mail::shutdown);
// FIXME: Pick correct opus_stream_config_t based on config.channels
auto stream = &stream_configs[map_stream(config.channels, config.flags[config_t::HIGH_QUALITY])];
auto stream = &stream_configs[map_stream(config.channels, config.flags[config_t::HIGH_QUALITY])];
auto ref = control_shared.ref();
if(!ref) {
@ -225,7 +224,7 @@ int map_stream(int channels, bool quality) {
int shift = quality ? 1 : 0;
switch(channels) {
case 2:
return STEREO;
return STEREO + shift;
case 6:
return SURROUND51 + shift;
case 8:

2
src/audio.h
View File

@ -6,6 +6,7 @@
namespace audio {
enum stream_config_e : int {
STEREO,
HIGH_STEREO,
SURROUND51,
HIGH_SURROUND51,
SURROUND71,
@ -19,6 +20,7 @@ struct opus_stream_config_t {
int streams;
int coupledStreams;
const std::uint8_t *mapping;
int bitrate;
};
extern opus_stream_config_t stream_configs[MAX_STREAM_CONFIG];

13
src/rtsp.cpp
View File

@ -646,6 +646,19 @@ void cmd_announce(rtsp_server_t *server, tcp::socket &sock, msg_t &&req) {
return;
}
// When using stereo audio, the audio quality is (strangely) indicated by whether the Host field
// in the RTSP message matches a local interface's IP address. Fortunately, Moonlight always sends
// 0.0.0.0 when it wants low quality, so it is easy to check without enumerating interfaces.
if(config.audio.channels == 2) {
for(auto option = req->options; option != nullptr; option = option->next) {
if("Host"sv == option->option) {
std::string_view content { option->content };
BOOST_LOG(debug) << "Found Host: "sv << content;
config.audio.flags[audio::config_t::HIGH_QUALITY] = (content.find("0.0.0.0"sv) == std::string::npos);
}
}
}
if(config.monitor.videoFormat != 0 && config::video.hevc_mode == 1) {
BOOST_LOG(warning) << "HEVC is disabled, yet the client requested HEVC"sv;