Automatically fall back to non-low latency on open() failure.

media/audio/audio_output_resampler.h

 // Copyright (c) 2012 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.
 
 #ifndef MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_
 #define MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_
 
 #include "base/basictypes.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
@@ skipping 14 matching lines @@
 //
 // The FIFO and resampler are only used when necessary.  To be clear:
 //   - The resampler is only used if the input and output sample rates differ.
 //   - The FIFO is only used if the input and output frame sizes differ or if
 //     the resampler is used.
 //
 // AOR works by intercepting the AudioSourceCallback provided to StartStream()
 // and redirecting to the appropriate resampling or FIFO callback which passes
 // through to the original callback only when necessary.
 //
+// AOR will automatically fall back from AUDIO_PCM_LOW_LATENCY to
+// AUDIO_PCM_LINEAR if the output device fails to open at the requested output
+// parameters.
+// TODO(dalecurtis): Ideally the low latency path will be as reliable as the
+// high latency path once we have channel mixing and support querying for the
+// hardware's configured bit depth.  Monitor the UMA stats for fallback and
+// remove fallback support once it's stable.  http://crbug.com/148418
+//
 // Currently channel downmixing and upmixing is not supported.
 // TODO(dalecurtis): Add channel remixing.  http://crbug.com/138762
 class MEDIA_EXPORT AudioOutputResampler
     : public AudioOutputDispatcher,
       public AudioOutputStream::AudioSourceCallback {
  public:
   AudioOutputResampler(AudioManager* audio_manager,
                        const AudioParameters& input_params,
                        const AudioParameters& output_params,
                        const base::TimeDelta& close_delay);
@@ skipping 14 matching lines @@
   virtual int OnMoreIOData(AudioBus* source,
                            AudioBus* dest,
                            AudioBuffersState buffers_state) OVERRIDE;
   virtual void OnError(AudioOutputStream* stream, int code) OVERRIDE;
   virtual void WaitTillDataReady() OVERRIDE;
 
  private:
   friend class base::RefCountedThreadSafe<AudioOutputResampler>;
   virtual ~AudioOutputResampler();
 
+  // Used to initialize the FIFO and resamplers.
+  void Initialize();
+
   // Called by MultiChannelResampler when more data is necessary.
   void ProvideInput(AudioBus* audio_bus);
 
   // Called by AudioPullFifo when more data is necessary.  Requires
   // |source_lock_| to have been acquired.
   void SourceCallback_Locked(AudioBus* audio_bus);
 
   // Used by StopStream()/CloseStream()/Shutdown() to clear internal state.
   // TODO(dalecurtis): Probably only one of these methods needs to call this,
   // the rest should DCHECK()/CHECK() that the values were reset.
@@ skipping 12 matching lines @@
   // Used to buffer data between the client and the output device in cases where
   // the client buffer size is not the same as the output device buffer size.
   // Bound to SourceCallback_Locked() so must only be used when |source_lock_|
   // has already been acquired.
   scoped_ptr<AudioPullFifo> audio_fifo_;
 
   // Ratio of input bytes to output bytes used to correct playback delay with
   // regard to buffering and resampling.
   double io_ratio_;
 
-  // Helper values for determining playback delay adjustment.
-  int input_bytes_per_frame_;
-  int output_bytes_per_frame_;
+  // Used by AudioOutputDispatcherImpl; kept so we can reinitialize on the fly.
+  base::TimeDelta close_delay_;
 
   // Last AudioBuffersState object received via OnMoreData(), used to correct
   // playback delay by ProvideInput() and passed on to |source_callback_|.
   AudioBuffersState current_buffers_state_;
 
   // Total number of bytes (in terms of output parameters) stored in resampler
   // or FIFO buffers which have not been sent to the audio device.
   int outstanding_audio_bytes_;
 
+  // AudioParameters used to setup the output stream.
+  AudioParameters output_params_;
+
   DISALLOW_COPY_AND_ASSIGN(AudioOutputResampler);
 };
 
 }  // namespace media
 
 #endif  // MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_