Support for Signal calls.

Merge in RedPhone // FREEBIE
2025-12-24 04:58:45 +00:00 · 2015-09-09 13:54:29 -07:00
parent 3d4ae60d81
commit d83a3d71bc
2585 changed files with 803492 additions and 45 deletions
--- a/jni/redphone/WebRtcJitterBuffer.cpp
+++ b/jni/redphone/WebRtcJitterBuffer.cpp
@@ -0,0 +1,109 @@
+#include "WebRtcJitterBuffer.h"
+
+#define TAG "WebRtcJitterBuffer"
+
+static volatile int running = 0;
+
+WebRtcJitterBuffer::WebRtcJitterBuffer(AudioCodec &codec) :
+  neteq(NULL), webRtcCodec(codec)
+{
+  running = 1;
+}
+
+int WebRtcJitterBuffer::init() {
+  webrtc::NetEq::Config config;
+  config.sample_rate_hz = 8000;
+
+  neteq = webrtc::NetEq::Create(config);
+
+  if (neteq == NULL) {
+    __android_log_print(ANDROID_LOG_WARN, TAG, "Failed to construct NetEq!");
+    return -1;
+  }
+
+  if (neteq->RegisterExternalDecoder(&webRtcCodec, webrtc::kDecoderPCMu, 0) != 0) {
+    __android_log_print(ANDROID_LOG_WARN, TAG, "Failed to register external codec!");
+    return -1;
+  }
+
+  pthread_t thread;
+  pthread_create(&thread, NULL, &WebRtcJitterBuffer::collectStats, this);
+
+  return 0;
+}
+
+WebRtcJitterBuffer::~WebRtcJitterBuffer() {
+  if (neteq != NULL) {
+    delete neteq;
+  }
+}
+
+void WebRtcJitterBuffer::addAudio(RtpPacket *packet, uint32_t tick) {
+  webrtc::WebRtcRTPHeader header;
+  header.header.payloadType    = packet->getPayloadType();
+  header.header.sequenceNumber = packet->getSequenceNumber();
+  header.header.timestamp      = packet->getTimestamp();
+  header.header.ssrc           = packet->getSsrc();
+
+  uint8_t *payload = (uint8_t*)malloc(packet->getPayloadLen());
+  memcpy(payload, packet->getPayload(), packet->getPayloadLen());
+
+  if (neteq->InsertPacket(header, payload, packet->getPayloadLen(), tick) != 0) {
+    __android_log_print(ANDROID_LOG_WARN, TAG, "neteq->InsertPacket() failed!");
+  }
+}
+
+int WebRtcJitterBuffer::getAudio(short *rawData, int maxRawData) {
+  int samplesPerChannel = 0;
+  int numChannels       = 0;
+
+  if (neteq->GetAudio(maxRawData, rawData, &samplesPerChannel, &numChannels, NULL) != 0) {
+    __android_log_print(ANDROID_LOG_WARN, TAG, "neteq->GetAudio() failed!");
+  }
+
+  return samplesPerChannel;
+}
+
+void WebRtcJitterBuffer::stop() {
+  running = 0;
+}
+
+void WebRtcJitterBuffer::collectStats() {
+  while (running) {
+    webrtc::NetEqNetworkStatistics stats;
+    neteq->NetworkStatistics(&stats);
+
+    __android_log_print(ANDROID_LOG_WARN, "WebRtcJitterBuffer",
+                        "Jitter Stats:\n{\n" \
+                        "  current_buffer_size_ms:   %d,\n" \
+                        "  preferred_buffer_size_ms: %d\n" \
+                        "  jitter_peaks_found:       %d\n" \
+                        "  packet_loss_rate:         %d\n" \
+                        "  packet_discard_rate:      %d\n" \
+                        "  expand_rate:              %d\n" \
+                        "  preemptive_rate:          %d\n" \
+                        "  accelerate_rate:          %d\n" \
+                        "  clockdrift_ppm:           %d\n" \
+                        "  added_zero_samples:       %d\n" \
+                        "}",
+                        stats.current_buffer_size_ms,
+                        stats.preferred_buffer_size_ms,
+                        stats.jitter_peaks_found,
+                        stats.packet_loss_rate,
+                        stats.packet_discard_rate,
+                        stats.expand_rate,
+                        stats.preemptive_rate,
+                        stats.accelerate_rate,
+                        stats.clockdrift_ppm,
+                        stats.added_zero_samples);
+    sleep(30);
+  }
+}
+
+void* WebRtcJitterBuffer::collectStats(void *context) {
+  WebRtcJitterBuffer* jitterBuffer = static_cast<WebRtcJitterBuffer*>(context);
+  jitterBuffer->collectStats();
+
+  return 0;
+}
+