Switch browser side audio capture path to use base time primitives.

media/audio/alsa/alsa_input.cc

Index: media/audio/alsa/alsa_input.cc
diff --git a/media/audio/alsa/alsa_input.cc b/media/audio/alsa/alsa_input.cc
index be414868ad91a7ac44eda71ef9b8f1577b471b0c..305d38950dd1936d42ba43c98fb2414ac961e102 100644
--- a/media/audio/alsa/alsa_input.cc
+++ b/media/audio/alsa/alsa_input.cc
@@ -200,8 +200,8 @@ void AlsaPcmInputStream::ReadAudio() {
   }
 
   int num_buffers = frames / params_.frames_per_buffer();
-  uint32_t hardware_delay_bytes =
-      static_cast<uint32_t>(GetCurrentDelay() * params_.GetBytesPerFrame());
+  base::TimeDelta hardware_delay = base::TimeDelta::FromSecondsD(
+      GetCurrentDelay() / static_cast<double>(params_.sample_rate()));
   double normalized_volume = 0.0;
 
   // Update the AGC volume level once every second. Note that, |volume| is
@@ -213,11 +213,10 @@ void AlsaPcmInputStream::ReadAudio() {
     int frames_read = wrapper_->PcmReadi(device_handle_, audio_buffer_.get(),
                                          params_.frames_per_buffer());
     if (frames_read == params_.frames_per_buffer()) {
-      audio_bus_->FromInterleaved(audio_buffer_.get(),
-                                  audio_bus_->frames(),
+      audio_bus_->FromInterleaved(audio_buffer_.get(), audio_bus_->frames(),
                                   params_.bits_per_sample() / 8);
-      callback_->OnData(
-          this, audio_bus_.get(), hardware_delay_bytes, normalized_volume);
+      callback_->OnData(this, audio_bus_.get(), hardware_delay,
+                        base::TimeTicks::Now(), normalized_volume);
     } else {
       LOG(WARNING) << "PcmReadi returning less than expected frames: "
                    << frames_read << " vs. " << params_.frames_per_buffer()