Fix Linux audio playback

Playing audio files shorter than about 11000 samples on Linux was either not working at all, or re-playing (starting and stopping on the same audio object) was working only intermittently.  It was possible to stop audio playback before the first buffer was played, resulting in no sound at all.  Also, if replaying, the buffer may have been dirty from the previous playback on the same device.

So on Linux at least, these issues appear fixed.

BUG=73045,47761,59367,59369,59370,65618
TEST=Manual, quickly playing short audio files on Linux should work

Review URL: http://codereview.chromium.org/6776024

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@81669 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 26 insertions, 2 deletions

diff --git a/media/audio/linux/alsa_output.cc b/media/audio/linux/alsa_output.cc
index d8e502c..8c5762b 100644
--- a/media/audio/linux/alsa_output.cc
+++ b/media/audio/linux/alsa_output.cc
@@ -395,6 +395,14 @@ void AlsaPcmOutputStream::StartTask() {
     return;
   }
 
+  if (shared_data_.state() != kIsPlaying) {
+    return;
+  }
+
+  // Before starting, the buffer might have audio from previous user of this
+  // device.
+  buffer_->Clear();
+
   // When starting again, drop all packets in the device and prepare it again
   // incase we are restarting from a pause state and need to flush old data.
   int error = wrapper_->PcmDrop(playback_handle_);
@@ -538,6 +546,10 @@ void AlsaPcmOutputStream::WritePacket() {
     return;
   }
 
+  if (shared_data_.state() == kIsStopped) {
+    return;
+  }
+
   CHECK_EQ(buffer_->forward_bytes() % bytes_per_output_frame_, 0u);
 
   const uint8* buffer_data;
@@ -579,6 +591,14 @@ void AlsaPcmOutputStream::WritePacket() {
       // Seek forward in the buffer after we've written some data to ALSA.
       buffer_->Seek(frames_written * bytes_per_output_frame_);
     }
+  } else {
+    // If nothing left to write and playback hasn't started yet, start it now.
+    // This ensures that shorter sounds will still play.
+    if (playback_handle_ &&
+        (wrapper_->PcmState(playback_handle_) == SND_PCM_STATE_PREPARED) &&
+        GetCurrentDelay() > 0) {
+      wrapper_->PcmStart(playback_handle_);
+    }
   }
 }
 
@@ -589,6 +609,10 @@ void AlsaPcmOutputStream::WriteTask() {
     return;
   }
 
+  if (shared_data_.state() == kIsStopped) {
+    return;
+  }
+
   bool source_exhausted;
   BufferPacket(&source_exhausted);
   WritePacket();
diff --git a/media/audio/linux/alsa_output_unittest.cc b/media/audio/linux/alsa_output_unittest.cc
index 832aa5f..911a8c8 100644
--- a/media/audio/linux/alsa_output_unittest.cc
+++ b/media/audio/linux/alsa_output_unittest.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
@@ -437,7 +437,7 @@ TEST_F(AlsaPcmOutputStreamTest, StartStop) {
   // Expect the pre-roll.
   MockAudioSourceCallback mock_callback;
   EXPECT_CALL(mock_alsa_wrapper_, PcmState(kFakeHandle))
-      .Times(2)
+      .Times(3)
       .WillRepeatedly(Return(SND_PCM_STATE_RUNNING));
   EXPECT_CALL(mock_alsa_wrapper_, PcmDelay(kFakeHandle, _))
       .Times(2)