Fix AudioPipeReader to use default pipe buffer size.

remoting/host/linux/audio_pipe_reader.cc

Index: remoting/host/linux/audio_pipe_reader.cc
diff --git a/remoting/host/linux/audio_pipe_reader.cc b/remoting/host/linux/audio_pipe_reader.cc
index 76be36c2ae4d361bfa7d0aaff63e770be73ad11f..a6a70c83da6ecf743cfd08aeb86ac1b890abd946 100644
--- a/remoting/host/linux/audio_pipe_reader.cc
+++ b/remoting/host/linux/audio_pipe_reader.cc
@@ -21,16 +21,6 @@ const int kSampleBytesPerSecond = AudioPipeReader::kSamplingRate *
                                   AudioPipeReader::kChannels *
                                   AudioPipeReader::kBytesPerSample;
 
-// Read data from the pipe every 40ms.
-const int kCapturingPeriodMs = 40;
-
-// Size of the pipe buffer in milliseconds.
-const int kPipeBufferSizeMs = kCapturingPeriodMs * 2;
-
-// Size of the pipe buffer in bytes.
-const int kPipeBufferSizeBytes = kPipeBufferSizeMs * kSampleBytesPerSecond /
-    base::Time::kMillisecondsPerSecond;
-
 #if !defined(F_SETPIPE_SZ)
 // F_SETPIPE_SZ is supported only starting linux 2.6.35, but we want to be able
 // to compile this code on machines with older kernel.
@@ -137,11 +127,9 @@ void AudioPipeReader::TryOpenPipe() {
       return;
     }
 
-    // Set buffer size for the pipe.
-    if (HANDLE_EINTR(fcntl(
-            pipe_.GetPlatformFile(), F_SETPIPE_SZ, kPipeBufferSizeBytes)) < 0) {
-      PLOG(ERROR) << "fcntl";
-    }
+    pipe_buffer_size_ = fpathconf(pipe_.GetPlatformFile(), _PC_PIPE_BUF);

[Lambros, 2014/12/23 23:07:18]

I don't think this call can fail, can it?

[Sergey Ulanov, 2014/12/24 00:01:28]

it shouldn't, but it's still better to check if it does. Done.

+    capture_period_ = base::TimeDelta::FromSeconds(1) * pipe_buffer_size_ /
+                      kSampleBytesPerSecond / 2;

[Lambros, 2014/12/23 23:07:18]

I'm curious to know why we divide by 2? Is it simply so that we capture twice as
often as we really need?

[Sergey Ulanov, 2014/12/24 00:01:26]

Yes. We need to capture more often than the buffer length. E.g. if the buffer
can fit 20ms of audio samples then we need to capture a more often than every
20ms. If we try scheduling the timer every 20ms, then, because the timer is not
precise, sometimes more than 20ms will pass between between DoCapture() calls
and as result it won't be reading enough audio samples, starving the stream on
the client.

 
     WaitForPipeReadable();
   }
@@ -151,8 +139,7 @@ void AudioPipeReader::StartTimer() {
   DCHECK(task_runner_->BelongsToCurrentThread());
   started_time_ = base::TimeTicks::Now();
   last_capture_position_ = 0;
-  timer_.Start(FROM_HERE, base::TimeDelta::FromMilliseconds(kCapturingPeriodMs),
-               this, &AudioPipeReader::DoCapture);
+  timer_.Start(FROM_HERE, capture_period_, this, &AudioPipeReader::DoCapture);
 }
 
 void AudioPipeReader::DoCapture() {
@@ -201,8 +188,8 @@ void AudioPipeReader::DoCapture() {
   // to read |bytes_to_read| bytes, but in case it's misbehaving we need to make
   // sure that |stream_position_bytes| doesn't go out of sync with the current
   // stream position.
-  if (stream_position_bytes - last_capture_position_ > kPipeBufferSizeBytes)
-    last_capture_position_ = stream_position_bytes - kPipeBufferSizeBytes;
+  if (stream_position_bytes - last_capture_position_ > pipe_buffer_size_)
+    last_capture_position_ = stream_position_bytes - pipe_buffer_size_;
   DCHECK_LE(last_capture_position_, stream_position_bytes);
 
   // Dispatch asynchronous notification to the stream observers.