Variable audio bitrate.

webrtc/call/rampup_tests.cc

Index: webrtc/call/rampup_tests.cc
diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc
index 150fc770ff0b75fb215375a11982ff64ad33dd5f..320d22de64daaea18ca0f9b77409f463826eace7 100644
--- a/webrtc/call/rampup_tests.cc
+++ b/webrtc/call/rampup_tests.cc
@@ -41,6 +41,7 @@ RampUpTester::RampUpTester(size_t num_video_streams,
       num_audio_streams_(num_audio_streams),
       rtx_(rtx),
       red_(red),
+      sender_call_(nullptr),
       send_stream_(nullptr),
       start_bitrate_bps_(start_bitrate_bps),
       start_bitrate_verified_(false),
@@ -53,8 +54,7 @@ RampUpTester::RampUpTester(size_t num_video_streams,
       audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)),
       poller_thread_(&BitrateStatsPollingThread,
                      this,
-                     "BitrateStatsPollingThread"),
-      sender_call_(nullptr) {
+                     "BitrateStatsPollingThread") {
   EXPECT_LE(num_audio_streams_, 1u);
   if (rtx_) {
     for (size_t i = 0; i < video_ssrcs_.size(); ++i)
@@ -188,6 +188,9 @@ void RampUpTester::ModifyAudioConfigs(
   send_config->rtp.ssrc = audio_ssrcs_[0];
   send_config->rtp.extensions.clear();
 
+  send_config->min_bitrate_kbps = 6;
+  send_config->max_bitrate_kbps = 60;
+
   bool transport_cc = false;
   if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
     transport_cc = false;
@@ -267,6 +270,9 @@ void RampUpTester::TriggerTestDone() {
   RTC_DCHECK_GE(test_start_ms_, 0);
 
   // TODO(holmer): Add audio send stats here too when those APIs are available.
+  if (!send_stream_)
+    return;
+
   VideoSendStream::Stats send_stats = send_stream_->GetStats();
 
   size_t total_packets_sent = 0;
@@ -341,6 +347,11 @@ bool RampUpDownUpTester::PollStats() {
       transmit_bitrate_bps += it.second.total_bitrate_bps;
     }
     EvolveTestState(transmit_bitrate_bps, stats.suspended);
+  } else if (num_audio_streams_ > 0 && sender_call_ != nullptr) {
+    // An audio send stream doesn't have bitrate stats, so the call send BW is
+    // currently used instead.
+    int transmit_bitrate_bps = sender_call_->GetStats().send_bandwidth_bps;
+    EvolveTestState(transmit_bitrate_bps, false);
   }
 
   return !event_.Wait(kPollIntervalMs);
@@ -380,7 +391,7 @@ void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
   switch (test_state_) {
     case kFirstRampup: {
       EXPECT_FALSE(suspended);
-      if (bitrate_bps > kExpectedHighBitrateBps) {
+      if (bitrate_bps >= kExpectedHighBitrateBps) {
         // The first ramp-up has reached the target bitrate. Change the
         // channel limit, and move to the next test state.
         forward_transport_config_.link_capacity_kbps =
@@ -397,7 +408,10 @@ void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
       break;
     }
     case kLowRate: {
-      if (bitrate_bps < kExpectedLowBitrateBps && suspended) {
+      // Audio streams are never suspended.
+      bool check_suspend_state = num_video_streams_ > 0;
+      if (bitrate_bps < kExpectedLowBitrateBps &&
+          suspended == check_suspend_state) {
         // The ramp-down was successful. Change the channel limit back to a
         // high value, and move to the next test state.
         forward_transport_config_.link_capacity_kbps =
@@ -414,7 +428,7 @@ void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
       break;
     }
     case kSecondRampup: {
-      if (bitrate_bps > kExpectedHighBitrateBps && !suspended) {
+      if (bitrate_bps >= kExpectedHighBitrateBps && !suspended) {
         webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
                                   "second_rampup", now - state_start_ms_, "ms",
                                   false);
@@ -520,6 +534,13 @@ TEST_F(RampUpTest, DISABLED_SendSideAudioVideoUpDownUpRtx) {
   RunBaseTest(&test);
 }
 
+TEST_F(RampUpTest, SendSideAudioOnlyUpDownUpRtx) {
+  RampUpDownUpTester test(0, 1, kStartBitrateBps,
+                          RtpExtension::kTransportSequenceNumberUri, true,
+                          false);
+  RunBaseTest(&test);
+}
+
 TEST_F(RampUpTest, AbsSendTimeSingleStream) {
   RampUpTester test(1, 0, 0, RtpExtension::kAbsSendTimeUri, false, false);
   RunBaseTest(&test);