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refactor audio

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This commit is contained in:
loki
2019-12-30 11:49:45 +01:00
parent 0289662017
commit 94e1dc903e
3 changed files with 37 additions and 51 deletions
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sunshine/audio.cpp
+6 -5
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@@ -12,6 +12,7 @@
namespace audio {
using namespace std::literals;
using opus_t = util::safe_ptr<OpusMSEncoder, opus_multistream_encoder_destroy>;
using sample_queue_t = std::shared_ptr<safe::queue_t<std::vector<std::int16_t>>>;
struct opus_stream_config_t {
std::int32_t sampleRate;
@@ -49,7 +50,7 @@ static opus_stream_config_t HighSurround51 = {
map_high_surround51
};
void encodeThread(std::shared_ptr<safe::queue_t<packet_t>> packets, std::shared_ptr<safe::queue_t<platf::audio_t>> samples, config_t config) {
void encodeThread(std::shared_ptr<safe::queue_t<packet_t>> packets, sample_queue_t samples, config_t config) {
//FIXME: Pick correct opus_stream_config_t based on config.channels
auto stream = &stereo;
opus_t opus { opus_multistream_encoder_create(
@@ -66,7 +67,7 @@ void encodeThread(std::shared_ptr<safe::queue_t<packet_t>> packets, std::shared_
while(auto sample = samples->pop()) {
packet_t packet { 16*1024 }; // 16KB
int bytes = opus_multistream_encode(opus.get(), platf::audio_data(sample), frame_size, std::begin(packet), packet.size());
int bytes = opus_multistream_encode(opus.get(), sample->data(), frame_size, std::begin(packet), packet.size());
if(bytes < 0) {
std::cout << "Error: "sv << opus_strerror(bytes) << std::endl;
exit(7);
@@ -78,7 +79,7 @@ void encodeThread(std::shared_ptr<safe::queue_t<packet_t>> packets, std::shared_
}
void capture(std::shared_ptr<safe::queue_t<packet_t>> packets, config_t config) {
auto samples = std::make_shared<safe::queue_t<platf::audio_t>>();
auto samples = std::make_shared<sample_queue_t::element_type>();
auto mic = platf::microphone();
if(!mic) {
@@ -89,11 +90,11 @@ void capture(std::shared_ptr<safe::queue_t<packet_t>> packets, config_t config)
auto stream = &stereo;
auto frame_size = config.packetDuration * stream->sampleRate / 1000;
int bytes_per_frame = frame_size * sizeof(std::int16_t) * stream->channelCount;
int samples_per_frame = frame_size * stream->channelCount;
std::thread thread { encodeThread, packets, samples, config };
while(packets->running()) {
auto sample = platf::audio(mic, bytes_per_frame);
auto sample = mic->sample(samples_per_frame);
samples->raise(std::move(sample));
}