Fixed ended event when playing audio on linux.

AudioRendererBase::FillBuffers() sends ended event only when playback_delay == 0, but with ALSA output the delay was never set to 0 because it includes hardware delay. Changed ALSA output to include only internal buffers in the delay value: this matches windows and mac versions.

BUG=45074
TEST=<audio> fires ended event when necessary

Review URL: http://codereview.chromium.org/2978005

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@52407 0039d316-1c4b-4281-b951-d872f2087c98

3 files changed, 3 insertions, 43 deletions

diff --git a/media/audio/linux/alsa_output.cc b/media/audio/linux/alsa_output.cc
index 0440ecf..1358dc3 100644
--- a/media/audio/linux/alsa_output.cc
+++ b/media/audio/linux/alsa_output.cc
@@ -486,7 +486,7 @@ void AlsaPcmOutputStream::BufferPacket(bool* source_exhausted) {
     // Before making a request to source for data. We need to determine the
     // delay (in bytes) for the requested data to be played.
     snd_pcm_sframes_t delay = buffer_->forward_bytes() * bytes_per_frame_ /
-        bytes_per_output_frame_ + GetCurrentDelay() * bytes_per_output_frame_;
+        bytes_per_output_frame_;
 
     scoped_refptr<media::DataBuffer> packet =
         new media::DataBuffer(packet_size_);
@@ -496,7 +496,7 @@ void AlsaPcmOutputStream::BufferPacket(bool* source_exhausted) {
     CHECK(packet_size <= packet->GetBufferSize()) <<
         "Data source overran buffer.";
 
-    // This should not happen, but incase it does, drop any trailing bytes
+    // This should not happen, but in case it does, drop any trailing bytes
     // that aren't large enough to make a frame.  Without this, packet writing
     // may stall because the last few bytes in the packet may never get used by
     // WritePacket.
@@ -801,29 +801,6 @@ snd_pcm_sframes_t AlsaPcmOutputStream::GetAvailableFrames() {
   return available_frames;
 }
 
-snd_pcm_sframes_t AlsaPcmOutputStream::GetCurrentDelay() {
-  snd_pcm_sframes_t delay = 0;
-
-  // Don't query ALSA's delay if we have underrun since it'll be jammed at
-  // some non-zero value and potentially even negative!
-  if (wrapper_->PcmState(playback_handle_) != SND_PCM_STATE_XRUN) {
-    int error = wrapper_->PcmDelay(playback_handle_, &delay);
-    if (error < 0) {
-      // Assume a delay of zero and attempt to recover the device.
-      delay = 0;
-      error = wrapper_->PcmRecover(playback_handle_,
-                                   error,
-                                   kPcmRecoverIsSilent);
-      if (error < 0) {
-        LOG(ERROR) << "Failed querying delay: " << wrapper_->StrError(error);
-      }
-    }
-    if (delay < 0)
-      delay = 0;
-  }
-  return delay;
-}
-
 snd_pcm_t* AlsaPcmOutputStream::AutoSelectDevice(unsigned int latency) {
   // For auto-selection:
   //   1) Attempt to open a device that best matches the number of channels
diff --git a/media/audio/linux/alsa_output.h b/media/audio/linux/alsa_output.h
index eaea5a2..6bc7de9 100644
--- a/media/audio/linux/alsa_output.h
+++ b/media/audio/linux/alsa_output.h
@@ -146,7 +146,6 @@ class AlsaPcmOutputStream :
                         uint32 latency);
   bool CloseDevice(snd_pcm_t* handle);
   snd_pcm_sframes_t GetAvailableFrames();
-  snd_pcm_sframes_t GetCurrentDelay();
 
   // Attempts to find the best matching linux audio device for the given number
   // of channels.  This function will set |device_name_| and |should_downmix_|.
diff --git a/media/audio/linux/alsa_output_unittest.cc b/media/audio/linux/alsa_output_unittest.cc
index 922ecdd..f0aefe6 100644
--- a/media/audio/linux/alsa_output_unittest.cc
+++ b/media/audio/linux/alsa_output_unittest.cc
@@ -439,12 +439,6 @@ TEST_F(AlsaPcmOutputStreamTest, StartStop) {
 
   // Expect the pre-roll.
   MockAudioSourceCallback mock_callback;
-  EXPECT_CALL(mock_alsa_wrapper_, PcmState(kFakeHandle))
-      .Times(2)
-      .WillRepeatedly(Return(SND_PCM_STATE_RUNNING));
-  EXPECT_CALL(mock_alsa_wrapper_, PcmDelay(kFakeHandle, _))
-      .Times(2)
-      .WillRepeatedly(DoAll(SetArgumentPointee<1>(0), Return(0)));
   EXPECT_CALL(mock_callback,
               OnMoreData(test_stream_.get(), _, kTestPacketSize, 0))
       .Times(2)
@@ -551,12 +545,8 @@ TEST_F(AlsaPcmOutputStreamTest, BufferPacket) {
 
   // Return a partially filled packet.
   MockAudioSourceCallback mock_callback;
-  EXPECT_CALL(mock_alsa_wrapper_, PcmState(_))
-      .WillOnce(Return(SND_PCM_STATE_RUNNING));
-  EXPECT_CALL(mock_alsa_wrapper_, PcmDelay(_, _))
-      .WillOnce(DoAll(SetArgumentPointee<1>(1), Return(0)));
   EXPECT_CALL(mock_callback,
-              OnMoreData(test_stream_.get(), _, _, kTestBytesPerFrame))
+              OnMoreData(test_stream_.get(), _, _, 0))
       .WillOnce(Return(10));
 
   bool source_exhausted;
@@ -574,10 +564,6 @@ TEST_F(AlsaPcmOutputStreamTest, BufferPacket_Negative) {
 
   // Simulate where the underrun has occurred right after checking the delay.
   MockAudioSourceCallback mock_callback;
-  EXPECT_CALL(mock_alsa_wrapper_, PcmState(_))
-      .WillOnce(Return(SND_PCM_STATE_RUNNING));
-  EXPECT_CALL(mock_alsa_wrapper_, PcmDelay(_, _))
-      .WillOnce(DoAll(SetArgumentPointee<1>(-1), Return(0)));
   EXPECT_CALL(mock_callback,
               OnMoreData(test_stream_.get(), _, _, 0))
       .WillOnce(Return(10));
@@ -597,8 +583,6 @@ TEST_F(AlsaPcmOutputStreamTest, BufferPacket_Underrun) {
 
   // If ALSA has underrun then we should assume a delay of zero.
   MockAudioSourceCallback mock_callback;
-  EXPECT_CALL(mock_alsa_wrapper_, PcmState(_))
-      .WillOnce(Return(SND_PCM_STATE_XRUN));
   EXPECT_CALL(mock_callback,
               OnMoreData(test_stream_.get(), _, _, 0))
       .WillOnce(Return(10));