mirror of
https://github.com/oxen-io/session-android.git
synced 2023-12-14 02:53:01 +01:00
parent
b5b564cfe1
commit
56a3c99289
8 changed files with 27 additions and 33 deletions
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@ -50,9 +50,15 @@ CallAudioManager::~CallAudioManager() {
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__android_log_print(ANDROID_LOG_WARN, TAG, "Shutting down...");
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microphoneReader.stop();
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__android_log_print(ANDROID_LOG_WARN, TAG, "Stopping audio player...");
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audioPlayer.stop();
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__android_log_print(ANDROID_LOG_WARN, TAG, "Stopping jitter buffer...");
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webRtcJitterBuffer.stop();
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__android_log_print(ANDROID_LOG_WARN, TAG, "Freeing resources...");
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if (sockAddr != NULL) {
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free(sockAddr);
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}
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@ -15,7 +15,7 @@ class CallAudioManager {
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private:
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volatile int running;
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volatile int finished;
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int finished;
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SLObjectItf engineObject;
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SLEngineItf engineEngine;
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AudioCodec audioCodec;
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@ -1,5 +1,5 @@
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#include "WebRtcJitterBuffer.h"
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#include <time.h>
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//#include <time.h>
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#define TAG "WebRtcJitterBuffer"
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@ -15,9 +15,7 @@ int WebRtcJitterBuffer::init() {
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webrtc::NetEq::Config config;
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config.sample_rate_hz = 8000;
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pthread_mutex_lock(&lock);
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neteq = webrtc::NetEq::Create(config);
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pthread_mutex_unlock(&lock);
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if (neteq == NULL) {
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__android_log_print(ANDROID_LOG_WARN, TAG, "Failed to construct NetEq!");
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@ -29,7 +27,8 @@ int WebRtcJitterBuffer::init() {
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return -1;
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}
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// pthread_create(&stats, NULL, &WebRtcJitterBuffer::collectStats, this);
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// pthread_t thread;
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// pthread_create(&thread, NULL, &WebRtcJitterBuffer::collectStats, this);
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return 0;
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}
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@ -67,21 +66,13 @@ int WebRtcJitterBuffer::getAudio(short *rawData, int maxRawData) {
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}
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void WebRtcJitterBuffer::stop() {
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// pthread_mutex_lock(&lock);
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running = 0;
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// pthread_cond_signal(&condition);
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// pthread_mutex_unlock(&lock);
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// pthread_join(stats, NULL);
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}
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void WebRtcJitterBuffer::collectStats() {
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while (running) {
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webrtc::NetEqNetworkStatistics stats;
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pthread_mutex_lock(&lock);
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neteq->NetworkStatistics(&stats);
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pthread_mutex_unlock(&lock);
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__android_log_print(ANDROID_LOG_WARN, "WebRtcJitterBuffer",
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"Jitter Stats:\n{\n" \
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@ -107,21 +98,22 @@ void WebRtcJitterBuffer::collectStats() {
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stats.clockdrift_ppm,
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stats.added_zero_samples);
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struct timespec timeToWait;
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struct timeval now;
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gettimeofday(&now, NULL);
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timeToWait.tv_sec = now.tv_sec;
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timeToWait.tv_nsec = now.tv_usec * 1000;
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timeToWait.tv_sec += 30;
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pthread_mutex_lock(&lock);
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if (running) {
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pthread_cond_timedwait(&condition, &lock, &timeToWait);
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}
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pthread_mutex_unlock(&lock);
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// struct timespec timeToWait;
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// struct timeval now;
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// gettimeofday(&now, NULL);
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//
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// timeToWait.tv_sec = now.tv_sec;
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// timeToWait.tv_nsec = now.tv_usec * 1000;
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// timeToWait.tv_sec += 30;
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//
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// pthread_mutex_lock(&lock);
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//
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// if (running) {
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// pthread_cond_timedwait(&condition, &lock, &timeToWait);
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// }
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//
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// pthread_mutex_unlock(&lock);
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sleep(30);
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}
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}
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@ -18,10 +18,6 @@ private:
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webrtc::NetEq *neteq;
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WebRtcCodec webRtcCodec;
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pthread_t stats;
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pthread_mutex_t lock;
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pthread_cond_t condition;
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public:
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WebRtcJitterBuffer(AudioCodec &codec);
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~WebRtcJitterBuffer();
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@ -47,7 +47,7 @@ public class CallAudioManager {
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if (Build.VERSION.SDK_INT >= 11) {
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ServiceUtil.getAudioManager(context).setMode(AudioManager.MODE_IN_COMMUNICATION);
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} else {
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ServiceUtil.getAudioManager(context).setMode(AudioManager.MODE_IN_CALL);
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// ServiceUtil.getAudioManager(context).setMode(AudioManager.MODE_IN_CALL);
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}
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try {
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