Merge AudioRendererImpl and AudioRendererBase; add NullAudioSink

This CL removes AudioRendererImpl and replaces it with AudioRendererBase.
NullAudioRenderer is also removed and replaced with NullAudioSink.
Also, a subtle bug is fixed in AudioRendererBase to allow for smooth
video playback when running Chrome with the --disable-audio flag.

BUG=119549,116645
TEST=media_unittests, playing video on Chrome/content_shell with and without --disable-audio flag should look identical


Review URL: http://codereview.chromium.org/9826023

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@131089 0039d316-1c4b-4281-b951-d872f2087c98

4 files changed, 211 insertions, 0 deletions

diff --git a/media/audio/audio_util.cc b/media/audio/audio_util.cc
index c15e845..01d71ee 100644
--- a/media/audio/audio_util.cc
+++ b/media/audio/audio_util.cc
@@ -427,6 +427,38 @@ ChannelLayout GetAudioInputHardwareChannelLayout(const std::string& device_id) {
 #endif
 }
 
+// Computes a buffer size based on the given |sample_rate|. Must be used in
+// conjunction with AUDIO_PCM_LINEAR.
+size_t GetHighLatencyOutputBufferSize(int sample_rate) {
+  // TODO(vrk/crogers): The buffer sizes that this function computes is probably
+  // overly conservative. However, reducing the buffer size to 2048-8192 bytes
+  // caused crbug.com/108396. This computation should be revisited while making
+  // sure crbug.com/108396 doesn't happen again.
+
+  // The minimum number of samples in a hardware packet.
+  // This value is selected so that we can handle down to 5khz sample rate.
+  static const size_t kMinSamplesPerHardwarePacket = 1024;
+
+  // The maximum number of samples in a hardware packet.
+  // This value is selected so that we can handle up to 192khz sample rate.
+  static const size_t kMaxSamplesPerHardwarePacket = 64 * 1024;
+
+  // This constant governs the hardware audio buffer size, this value should be
+  // chosen carefully.
+  // This value is selected so that we have 8192 samples for 48khz streams.
+  static const size_t kMillisecondsPerHardwarePacket = 170;
+
+  // Select the number of samples that can provide at least
+  // |kMillisecondsPerHardwarePacket| worth of audio data.
+  size_t samples = kMinSamplesPerHardwarePacket;
+  while (samples <= kMaxSamplesPerHardwarePacket &&
+         samples * base::Time::kMillisecondsPerSecond <
+         sample_rate * kMillisecondsPerHardwarePacket) {
+    samples *= 2;
+  }
+  return samples;
+}
+
 // When transferring data in the shared memory, first word is size of data
 // in bytes. Actual data starts immediately after it.
 
diff --git a/media/audio/audio_util.h b/media/audio/audio_util.h
index 2fccc1e..df5683f 100644
--- a/media/audio/audio_util.h
+++ b/media/audio/audio_util.h
@@ -107,6 +107,10 @@ MEDIA_EXPORT size_t GetAudioHardwareBufferSize();
 MEDIA_EXPORT ChannelLayout GetAudioInputHardwareChannelLayout(
     const std::string& device_id);
 
+// Computes a buffer size based on the given |sample_rate|. Must be used in
+// conjunction with AUDIO_PCM_LINEAR.
+MEDIA_EXPORT size_t GetHighLatencyOutputBufferSize(int sample_rate);
+
 // Functions that handle data buffer passed between processes in the shared
 // memory. Called on both IPC sides.
 
diff --git a/media/audio/null_audio_sink.cc b/media/audio/null_audio_sink.cc
new file mode 100644
index 0000000..e8cd4eb
--- /dev/null
+++ b/media/audio/null_audio_sink.cc
@@ -0,0 +1,110 @@
+// 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.
+
+#include "media/audio/null_audio_sink.h"
+
+#include "base/bind.h"
+#include "base/threading/platform_thread.h"
+
+namespace media {
+
+NullAudioSink::NullAudioSink()
+    : initialized_(false),
+      playback_rate_(0.0),
+      playing_(false),
+      callback_(NULL),
+      thread_("NullAudioThread") {
+}
+
+NullAudioSink::~NullAudioSink() {
+  DCHECK(!thread_.IsRunning());
+  for (size_t i = 0; i < audio_data_.size(); ++i)
+    delete [] audio_data_[i];
+}
+
+
+void NullAudioSink::Start() {
+  if (!thread_.Start())
+    return;
+
+  thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
+      &NullAudioSink::FillBufferTask, this));
+}
+
+void NullAudioSink::Stop() {
+  SetPlaying(false);
+  thread_.Stop();
+}
+
+void NullAudioSink::Play() {
+  SetPlaying(true);
+}
+
+void NullAudioSink::Pause(bool /* flush */) {
+  SetPlaying(false);
+}
+
+void NullAudioSink::SetPlaybackRate(float rate) {
+  base::AutoLock auto_lock(lock_);
+  playback_rate_ = rate;
+}
+
+bool NullAudioSink::SetVolume(double volume) {
+  // Audio is always muted.
+  return volume == 0.0;
+}
+
+void NullAudioSink::GetVolume(double* volume) {
+  // Audio is always muted.
+  *volume = 0.0;
+}
+
+void NullAudioSink::SetPlaying(bool is_playing) {
+  base::AutoLock auto_lock(lock_);
+  playing_ = is_playing;
+}
+
+void NullAudioSink::Initialize(const AudioParameters& params,
+                               RenderCallback* callback) {
+  DCHECK(!initialized_);
+  params_ = params;
+
+  audio_data_.reserve(params.channels());
+  for (int i = 0; i < params.channels(); ++i) {
+    float* channel_data = new float[params.frames_per_buffer()];
+    audio_data_.push_back(channel_data);
+  }
+
+  callback_ = callback;
+  initialized_ = true;
+}
+
+void NullAudioSink::FillBufferTask() {
+  base::AutoLock auto_lock(lock_);
+
+  base::TimeDelta delay;
+  // Only consume buffers when actually playing.
+  if (playing_)  {
+    DCHECK_GT(playback_rate_, 0.0f);
+    int requested_frames = params_.frames_per_buffer();
+    int frames_received = callback_->Render(audio_data_, requested_frames, 0);
+    int frames_per_millisecond =
+        params_.sample_rate() / base::Time::kMillisecondsPerSecond;
+
+    // Calculate our sleep duration, taking playback rate into consideration.
+    delay = base::TimeDelta::FromMilliseconds(
+        frames_received / (frames_per_millisecond * playback_rate_));
+  } else {
+    // If paused, sleep for 10 milliseconds before polling again.
+    delay = base::TimeDelta::FromMilliseconds(10);
+  }
+
+  // Sleep for at least one millisecond so we don't spin the CPU.
+  MessageLoop::current()->PostDelayedTask(
+      FROM_HERE,
+      base::Bind(&NullAudioSink::FillBufferTask, this),
+      std::max(delay, base::TimeDelta::FromMilliseconds(1)));
+}
+
+}  // namespace media
diff --git a/media/audio/null_audio_sink.h b/media/audio/null_audio_sink.h
new file mode 100644
index 0000000..32245eb
--- /dev/null
+++ b/media/audio/null_audio_sink.h
@@ -0,0 +1,65 @@
+// 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_FILTERS_NULL_AUDIO_RENDERER_H_
+#define MEDIA_FILTERS_NULL_AUDIO_RENDERER_H_
+
+// NullAudioSink effectively uses an extra thread to "throw away" the
+// audio data at a rate resembling normal playback speed.  It's just like
+// decoding to /dev/null!
+//
+// NullAudioSink can also be used in situations where the client has no
+// audio device or we haven't written an audio implementation for a particular
+// platform yet.
+
+#include <vector>
+
+#include "base/threading/thread.h"
+#include "media/base/audio_renderer_sink.h"
+
+namespace media {
+
+class MEDIA_EXPORT NullAudioSink
+    : NON_EXPORTED_BASE(public AudioRendererSink) {
+ public:
+  NullAudioSink();
+  virtual ~NullAudioSink();
+
+  // AudioRendererSink implementation.
+  virtual void Initialize(const AudioParameters& params,
+                          RenderCallback* callback) OVERRIDE;
+  virtual void Start() OVERRIDE;
+  virtual void Stop() OVERRIDE;
+  virtual void Pause(bool flush) OVERRIDE;
+  virtual void Play() OVERRIDE;
+  virtual void SetPlaybackRate(float rate) OVERRIDE;
+  virtual bool SetVolume(double volume) OVERRIDE;
+  virtual void GetVolume(double* volume) OVERRIDE;
+
+ private:
+  // Audio thread task that periodically calls FillBuffer() to consume
+  // audio data.
+  void FillBufferTask();
+
+  void SetPlaying(bool is_playing);
+
+  // A buffer passed to FillBuffer to advance playback.
+  std::vector<float*> audio_data_;
+
+  AudioParameters params_;
+  bool initialized_;
+  float playback_rate_;
+  bool playing_;
+  RenderCallback* callback_;
+
+  // Separate thread used to throw away data.
+  base::Thread thread_;
+  base::Lock lock_;
+
+  DISALLOW_COPY_AND_ASSIGN(NullAudioSink);
+};
+
+}  // namespace media
+
+#endif  // MEDIA_FILTERS_NULL_AUDIO_RENDERER_H_