moonlight-stream/moonlight-embedded
1
0
Fork 0
mirror of https://github.com/moonlight-stream/moonlight-embedded.git synced 2025-07-01 15:25:35 +00:00
Code Issues Packages Projects Releases Wiki Activity

Less audio stutter

Browse Source
This commit is contained in:
Iwan Timmer 2015-03-29 22:54:25 +02:00
parent ba65e6252e
commit fe829b32fe
6 changed files with 68 additions and 88 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

51
jni/nv_alsa/nv_alsa.c
View File

@ -2,29 +2,60 @@
* in this header */
#include <alsa/asoundlib.h>
#define CHECK_RETURN(f) if ((rc = f) != 0) return rc;
snd_pcm_t *handle;
unsigned int channels;
int nv_alsa_init(unsigned int channelCount, unsigned int sampleRate, unsigned char* device) {
int rc;
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
snd_pcm_uframes_t period_size = 240 * channelCount * 2;
snd_pcm_uframes_t buffer_size = 12 * period_size;
channels = channelCount;
/* Open PCM device for playback. */
if ((rc = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) != 0)
return rc;
if ((rc = snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, channelCount, sampleRate, 1, 50000)) != 0) //50ms latency
return rc;
CHECK_RETURN(snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))
/* Set hardware parameters */
CHECK_RETURN(snd_pcm_hw_params_malloc(&hw_params));
CHECK_RETURN(snd_pcm_hw_params_any(handle, hw_params));
CHECK_RETURN(snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED));
CHECK_RETURN(snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16_LE));
CHECK_RETURN(snd_pcm_hw_params_set_rate_near(handle, hw_params, &sampleRate, NULL));
CHECK_RETURN(snd_pcm_hw_params_set_channels(handle, hw_params, channelCount));
CHECK_RETURN(snd_pcm_hw_params_set_buffer_size_near(handle, hw_params, &buffer_size));
CHECK_RETURN(snd_pcm_hw_params_set_period_size_near(handle, hw_params, &period_size, NULL));
CHECK_RETURN(snd_pcm_hw_params(handle, hw_params));
snd_pcm_hw_params_free(hw_params);
/* Set software parameters */
CHECK_RETURN(snd_pcm_sw_params_malloc(&sw_params));
CHECK_RETURN(snd_pcm_sw_params_current(handle, sw_params));
CHECK_RETURN(snd_pcm_sw_params_set_start_threshold(handle, sw_params, buffer_size - period_size));
CHECK_RETURN(snd_pcm_sw_params_set_avail_min(handle, sw_params, period_size));
CHECK_RETURN(snd_pcm_sw_params(handle, sw_params));
snd_pcm_sw_params_free(sw_params);
CHECK_RETURN(snd_pcm_prepare(handle));
return 0;
}
int nv_alsa_play(const unsigned char* indata, int data_len) {
int frames = data_len/4; /* 2 bytes/sample, 2 channels */
int frames = data_len / (2 * channels); /* 2 bytes/sample */
int rc = snd_pcm_writei(handle, indata, frames);
if (rc == -EPIPE)
snd_pcm_prepare(handle);
snd_pcm_recover(handle, rc, 1);
return rc;
}
int nv_alsa_close(void) {
snd_pcm_drain(handle);
snd_pcm_close(handle);
if (handle != NULL) {
snd_pcm_drain(handle);
snd_pcm_close(handle);
}
}

35
jni/nv_opus_dec/nv_opus_dec.c
View File

@ -6,22 +6,11 @@
OpusDecoder* decoder;
#ifdef _WIN32
#pragma comment(lib, "opus.lib")
#pragma comment(lib, "celt.lib")
#pragma comment(lib, "silk_common.lib")
#pragma comment(lib, "silk_float.lib")
#endif
// This function must be called before
// any other decoding functions
int nv_opus_init(void) {
int nv_opus_init(unsigned int channelcount, unsigned int samplerate) {
int err;
decoder = opus_decoder_create(
nv_opus_get_sample_rate(),
nv_opus_get_channel_count(),
&err);
decoder = opus_decoder_create(samplerate, channelcount, &err);
return err;
}
@ -33,32 +22,16 @@ void nv_opus_destroy(void) {
}
}
// The Opus stream is stereo
int nv_opus_get_channel_count(void) {
return 2;
}
// This number assumes 2 channels with 16-bit samples at 48 KHz with 2.5 ms frames
int nv_opus_get_max_out_shorts(void) {
return 240*nv_opus_get_channel_count();
}
// The Opus stream is 48 KHz
int nv_opus_get_sample_rate(void) {
return 48000;
}
// outpcmdata must be 5760*2 shorts in length
// packets must be decoded in order
// a packet loss must call this function with NULL indata and 0 inlen
// returns the number of decoded samples
int nv_opus_decode(unsigned char* indata, int inlen, short* outpcmdata) {
int nv_opus_decode(unsigned char* indata, int inlen, int framesize, short* outpcmdata) {
int err;
// Decoding to 16-bit PCM with FEC off
// Maximum length assuming 48KHz sample rate
err = opus_decode(decoder, indata, inlen,
outpcmdata, 512, 0);
err = opus_decode(decoder, indata, inlen, outpcmdata, framesize, 0);
return err;
}

7
jni/nv_opus_dec/nv_opus_dec.h
View File

@ -1,6 +1,3 @@
int nv_opus_init(void);
int nv_opus_init(unsigned int channelcount, unsigned int samplerate);
void nv_opus_destroy(void);
int nv_opus_get_channel_count(void);
int nv_opus_get_max_out_shorts(void);
int nv_opus_get_sample_rate(void);
int nv_opus_decode(unsigned char* indata, int inlen, short* outpcmdata);
int nv_opus_decode(unsigned char* indata, int inlen, int framelen, short* outpcmdata);

31
jni/nv_opus_dec/nv_opus_dec_jni.c
View File

@ -8,8 +8,8 @@
// This function must be called before
// any other decoding functions
JNIEXPORT jint JNICALL
Java_com_limelight_binding_audio_OpusDecoder_init(JNIEnv *env, jobject this) {
return nv_opus_init();
Java_com_limelight_binding_audio_OpusDecoder_init(JNIEnv *env, jobject this, jint channelcount, jint samplerate) {
return nv_opus_init(channelcount, samplerate);
}
// This function must be called after
@ -19,24 +19,6 @@ Java_com_limelight_binding_audio_OpusDecoder_destroy(JNIEnv *env, jobject this)
nv_opus_destroy();
}
// The Opus stream is stereo
JNIEXPORT jint JNICALL
Java_com_limelight_binding_audio_OpusDecoder_getChannelCount(JNIEnv *env, jobject this) {
return nv_opus_get_channel_count();
}
// This number assumes 2 channels at 48 KHz
JNIEXPORT jint JNICALL
Java_com_limelight_binding_audio_OpusDecoder_getMaxOutputShorts(JNIEnv *env, jobject this) {
return nv_opus_get_max_out_shorts();
}
// The Opus stream is 48 KHz
JNIEXPORT jint JNICALL
Java_com_limelight_binding_audio_OpusDecoder_getSampleRate(JNIEnv *env, jobject this) {
return nv_opus_get_sample_rate();
}
// outpcmdata must be 5760*2 shorts in length
// packets must be decoded in order
// a packet loss must call this function with NULL indata and 0 inlen
@ -44,7 +26,7 @@ Java_com_limelight_binding_audio_OpusDecoder_getSampleRate(JNIEnv *env, jobject
JNIEXPORT jint JNICALL
Java_com_limelight_binding_audio_OpusDecoder_decode(
JNIEnv *env, jobject this, // JNI parameters
jbyteArray indata, jint inoff, jint inlen, // Input parameters
jbyteArray indata, jint inoff, jint inlen, jint framesize, // Input parameters
jbyteArray outpcmdata) // Output parameter
{
jint ret;
@ -55,13 +37,12 @@ Java_com_limelight_binding_audio_OpusDecoder_decode(
if (indata != NULL) {
jni_input_data = (*env)->GetByteArrayElements(env, indata, 0);
ret = nv_opus_decode(&jni_input_data[inoff], inlen, (jshort*) jni_pcm_data);
ret = nv_opus_decode(&jni_input_data[inoff], inlen, framesize, (jshort*) jni_pcm_data);
// The input data isn't changed so it can be safely aborted
(*env)->ReleaseByteArrayElements(env, indata, jni_input_data, JNI_ABORT);
}
else {
ret = nv_opus_decode(NULL, 0, (jshort*) jni_pcm_data);
} else {
ret = nv_opus_decode(NULL, 0, framesize, (jshort*) jni_pcm_data);
}
(*env)->ReleaseByteArrayElements(env, outpcmdata, jni_pcm_data, 0);

25
src/com/limelight/binding/audio/AlsaAudioDecoderRenderer.java
View File

@ -9,39 +9,40 @@ import com.limelight.nvstream.av.audio.AudioDecoderRenderer;
*/
public class AlsaAudioDecoderRenderer implements AudioDecoderRenderer {
private final static int CHANNEL_COUNT = 2;
private final static int SAMPLE_RATE = 48000;
/* Number of 16 bits frames */
private final static int FRAME_SIZE = 240;
private String device;
private byte[] decodedData;
public AlsaAudioDecoderRenderer(String device) {
this.device = device;
this.decodedData = new byte[OpusDecoder.getMaxOutputShorts()*2];
int err;
err = OpusDecoder.init();
if (err != 0) {
throw new IllegalStateException("Opus decoder failed to initialize");
}
this.decodedData = new byte[FRAME_SIZE * CHANNEL_COUNT * 2];
}
@Override
public boolean streamInitialize() {
return AlsaAudio.init(OpusDecoder.getChannelCount(), OpusDecoder.getSampleRate(), device) == 0;
return OpusDecoder.init(CHANNEL_COUNT, SAMPLE_RATE) == 0 && AlsaAudio.init(CHANNEL_COUNT, SAMPLE_RATE, device) == 0;
}
@Override
public void playAudio(byte[] bytes, int offset, int length) {
int decodeLen = OpusDecoder.decode(bytes, offset, length, decodedData);
int decodeLen = OpusDecoder.decode(bytes, offset, length, FRAME_SIZE, decodedData);
if (decodeLen > 0) {
//Value of decode is frames (shorts) decoded per channel
decodeLen *= 2*OpusDecoder.getChannelCount();
decodeLen *= CHANNEL_COUNT * 2;
int rc = AlsaAudio.play(decodedData, 0, decodeLen);
if (rc<0)
LimeLog.warning("Alsa error from writei: "+rc);
else if (rc!=length/4)
else if (rc!=decodeLen/4)
LimeLog.warning("Alsa short write, write "+rc+" frames");
} else {
LimeLog.warning("Opus error from decode: "+decodeLen);
}
}

7
src/com/limelight/binding/audio/OpusDecoder.java
View File

@ -5,10 +5,7 @@ public class OpusDecoder {
System.loadLibrary("nv_opus_dec");
}
public static native int init();
public static native int init(int channelCoumt, int sampleRate);
public static native void destroy();
public static native int getChannelCount();
public static native int getMaxOutputShorts();
public static native int getSampleRate();
public static native int decode(byte[] indata, int inoff, int inlen, byte[] outpcmdata);
public static native int decode(byte[] indata, int inoff, int inlen, int frameSize, byte[] outpcmdata);
}