Revert of Remove the last piece of deprecated synchronous IO code.

media/audio/alsa/audio_manager_alsa.cc

Index: media/audio/alsa/audio_manager_alsa.cc
diff --git a/media/audio/alsa/audio_manager_alsa.cc b/media/audio/alsa/audio_manager_alsa.cc
index 76248348c464fd88575437bd2ea582ba4f755398..beb60bad88b02af515dd1075fb54a43edd226cb6 100644
--- a/media/audio/alsa/audio_manager_alsa.cc
+++ b/media/audio/alsa/audio_manager_alsa.cc
@@ -311,6 +311,7 @@
   int sample_rate = kDefaultSampleRate;
   int buffer_size = kDefaultOutputBufferSize;
   int bits_per_sample = 16;
+  int input_channels = 0;
   if (input_params.IsValid()) {
     // Some clients, such as WebRTC, have a more limited use case and work
     // acceptably with a smaller buffer size.  The check below allows clients
@@ -320,6 +321,7 @@
     sample_rate = input_params.sample_rate();
     bits_per_sample = input_params.bits_per_sample();
     channel_layout = input_params.channel_layout();
+    input_channels = input_params.input_channels();
     buffer_size = std::min(input_params.frames_per_buffer(), buffer_size);
   }
 
@@ -328,7 +330,7 @@
     buffer_size = user_buffer_size;
 
   return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
       sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS);
 }