Adding audio unit tests for Android

media/audio/android/opensles_output.cc

 // 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/android/opensles_output.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "media/audio/android/audio_manager_android.h"
 
 #define LOG_ON_FAILURE_AND_RETURN(op, ...) \
   do { \
     SLresult err = (op);            \
     if (err != SL_RESULT_SUCCESS) { \
       DLOG(ERROR) << #op << " failed: " << err; \
       return __VA_ARGS__; \
     } \
   } while (0)
 
 namespace media {
 
 OpenSLESOutputStream::OpenSLESOutputStream(AudioManagerAndroid* manager,
                                            const AudioParameters& params)
     : audio_manager_(manager),
       callback_(NULL),
       player_(NULL),
       simple_buffer_queue_(NULL),
-      active_queue_(0),
+      active_buffer_index_(0),
       buffer_size_bytes_(0),
       started_(false),
       volume_(1.0) {
+  DVLOG(2) << "OpenSLESOutputStream::OpenSLESOutputStream()";
   format_.formatType = SL_DATAFORMAT_PCM;
   format_.numChannels = static_cast<SLuint32>(params.channels());
   // Provides sampling rate in milliHertz to OpenSLES.
   format_.samplesPerSec = static_cast<SLuint32>(params.sample_rate() * 1000);
   format_.bitsPerSample = params.bits_per_sample();
   format_.containerSize = params.bits_per_sample();
   format_.endianness = SL_BYTEORDER_LITTLEENDIAN;
   if (format_.numChannels == 1)
     format_.channelMask = SL_SPEAKER_FRONT_CENTER;
   else if (format_.numChannels == 2)
     format_.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
   else
     NOTREACHED() << "Unsupported number of channels: " << format_.numChannels;
 
   buffer_size_bytes_ = params.GetBytesPerBuffer();
   audio_bus_ = AudioBus::Create(params);
 
   memset(&audio_data_, 0, sizeof(audio_data_));
 }
 
 OpenSLESOutputStream::~OpenSLESOutputStream() {
+  DVLOG(2) << "OpenSLESOutputStream::~OpenSLESOutputStream()";
+  DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(!engine_object_.Get());
   DCHECK(!player_object_.Get());
   DCHECK(!output_mixer_.Get());
   DCHECK(!player_);
   DCHECK(!simple_buffer_queue_);
   DCHECK(!audio_data_[0]);
 }
 
 bool OpenSLESOutputStream::Open() {
+  DVLOG(2) << "OpenSLESOutputStream::Open()";
+  DCHECK(thread_checker_.CalledOnValidThread());
   if (engine_object_.Get())
     return false;
 
   if (!CreatePlayer())
     return false;
 
   SetupAudioBuffer();
+  active_buffer_index_ = 0;
 
   return true;
 }
 
 void OpenSLESOutputStream::Start(AudioSourceCallback* callback) {
+  DVLOG(2) << "OpenSLESOutputStream::Start()";
+  DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(callback);
   DCHECK(player_);
   DCHECK(simple_buffer_queue_);
   if (started_)
     return;
 
-  // Enable the flags before streaming.
+  base::AutoLock lock(lock_);
+  DCHECK(callback_ == NULL || callback_ == callback);
   callback_ = callback;
-  active_queue_ = 0;
-  started_ = true;
 
   // Avoid start-up glitches by filling up one buffer queue before starting
   // the stream.
-  FillBufferQueue();
+  FillBufferQueueNoLock();
 
-  // Start streaming data by setting the play state to |SL_PLAYSTATE_PLAYING|.
+  // Start streaming data by setting the play state to SL_PLAYSTATE_PLAYING.
+  // For a player object, when the object is in the SL_PLAYSTATE_PLAYING
+  // state, adding buffers will implicitly start playback.
   LOG_ON_FAILURE_AND_RETURN(
       (*player_)->SetPlayState(player_, SL_PLAYSTATE_PLAYING));
+
+  started_ = true;
 }
 
 void OpenSLESOutputStream::Stop() {
+  DVLOG(2) << "OpenSLESOutputStream::Stop()";
+  DCHECK(thread_checker_.CalledOnValidThread());
   if (!started_)
     return;
 
-  started_ = false;
-  // Stop playing by setting the play state to |SL_PLAYSTATE_STOPPED|.
+  base::AutoLock lock(lock_);
+
+  // Stop playing by setting the play state to SL_PLAYSTATE_STOPPED.
   LOG_ON_FAILURE_AND_RETURN(
       (*player_)->SetPlayState(player_, SL_PLAYSTATE_STOPPED));
 
   // Clear the buffer queue so that the old data won't be played when
   // resuming playing.
   LOG_ON_FAILURE_AND_RETURN(
       (*simple_buffer_queue_)->Clear(simple_buffer_queue_));
+
+#ifndef NDEBUG
+  // Verify that the buffer queue is in fact cleared as it should.
+  SLAndroidSimpleBufferQueueState buffer_queue_state;
+  LOG_ON_FAILURE_AND_RETURN(
+      (*simple_buffer_queue_)->GetState(simple_buffer_queue_,
+                                        &buffer_queue_state));
+  DCHECK_EQ(0u, buffer_queue_state.count);
+  DCHECK_EQ(0u, buffer_queue_state.index);
+#endif
+
+  started_ = false;
 }
 
 void OpenSLESOutputStream::Close() {
+  DVLOG(2) << "OpenSLESOutputStream::Close()";
+  DCHECK(thread_checker_.CalledOnValidThread());
+
   // Stop the stream if it is still playing.
   Stop();
+  {
+    // Destroy the buffer queue player object and invalidate all associated
+    // interfaces.
+    player_object_.Reset();
+    simple_buffer_queue_ = NULL;
+    player_ = NULL;
 
-  // Explicitly free the player objects and invalidate their associated
-  // interfaces. They have to be done in the correct order.
-  player_object_.Reset();
-  output_mixer_.Reset();
-  engine_object_.Reset();
-  simple_buffer_queue_ = NULL;
-  player_ = NULL;
+    // Destroy the mixer object. We don't store any associated interface for
+    // this object.
+    output_mixer_.Reset();
 
-  ReleaseAudioBuffer();
+    // Destroy the engine object. We don't store any associated interface for
+    // this object.
+    engine_object_.Reset();
+    ReleaseAudioBuffer();
+  }
 
   audio_manager_->ReleaseOutputStream(this);
 }
 
 void OpenSLESOutputStream::SetVolume(double volume) {
+  DVLOG(2) << "OpenSLESOutputStream::SetVolume(" << volume << ")";
+  DCHECK(thread_checker_.CalledOnValidThread());
   float volume_float = static_cast<float>(volume);
   if (volume_float < 0.0f || volume_float > 1.0f) {
     return;
   }
   volume_ = volume_float;
 }
 
 void OpenSLESOutputStream::GetVolume(double* volume) {
+  DCHECK(thread_checker_.CalledOnValidThread());
   *volume = static_cast<double>(volume_);
 }
 
 bool OpenSLESOutputStream::CreatePlayer() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(!engine_object_.Get());
+  DCHECK(!player_object_.Get());
+  DCHECK(!output_mixer_.Get());
+  DCHECK(!player_);
+  DCHECK(!simple_buffer_queue_);
+
   // Initializes the engine object with specific option. After working with the
   // object, we need to free the object and its resources.
   SLEngineOption option[] = {
     { SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE) }
   };
   LOG_ON_FAILURE_AND_RETURN(
       slCreateEngine(engine_object_.Receive(), 1, option, 0, NULL, NULL),
       false);
 
   // Realize the SL engine object in synchronous mode.
@@ skipping 19 matching lines @@
       false);
 
   // Realizing the output mix object in synchronous mode.
   LOG_ON_FAILURE_AND_RETURN(
       output_mixer_->Realize(output_mixer_.Get(), SL_BOOLEAN_FALSE),
       false);
 
   // Audio source configuration.
   SLDataLocator_AndroidSimpleBufferQueue simple_buffer_queue = {
     SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
-    static_cast<SLuint32>(kNumOfQueuesInBuffer)
+    static_cast<SLuint32>(kMaxNumOfBuffersInQueue)
   };
   SLDataSource audio_source = { &simple_buffer_queue, &format_ };
 
   // Audio sink configuration.
   SLDataLocator_OutputMix locator_output_mix = {
     SL_DATALOCATOR_OUTPUTMIX, output_mixer_.Get()
   };
   SLDataSink audio_sink = { &locator_output_mix, NULL };
 
   // Create an audio player.
@@ skipping 61 matching lines @@
 }
 
 void OpenSLESOutputStream::SimpleBufferQueueCallback(
     SLAndroidSimpleBufferQueueItf buffer_queue, void* instance) {
   OpenSLESOutputStream* stream =
       reinterpret_cast<OpenSLESOutputStream*>(instance);
   stream->FillBufferQueue();
 }
 
 void OpenSLESOutputStream::FillBufferQueue() {
-  if (!started_)
+  base::AutoLock lock(lock_);
+  if (!player_)
     return;
 
   TRACE_EVENT0("audio", "OpenSLESOutputStream::FillBufferQueue");
+
+  // Verify that we are in a playing state.
+  SLuint32 state;
+  SLresult err = (*player_)->GetPlayState(player_, &state);
+  if (SL_RESULT_SUCCESS != err) {
+    HandleError(err);
+    return;
+  }
+  if (state != SL_PLAYSTATE_PLAYING) {
+    DLOG(WARNING) << "Received callback in non-playing state";
+    return;
+  }
+
+  // Fill up one buffer in the queue by asking the registered source for
+  // data using the OnMoreData() callback.
+  FillBufferQueueNoLock();
+}
+
+void OpenSLESOutputStream::FillBufferQueueNoLock() {
+  // Ensure that the calling thread has acquired the lock since it is not
+  // done in this method.
+  lock_.AssertAcquired();
+
   // Read data from the registered client source.
-  // TODO(xians): Get an accurate delay estimation.
-  uint32 hardware_delay = buffer_size_bytes_;
+  // TODO(henrika): Investigate if it is possible to get a more accurate
+  // delay estimation.
+  const uint32 hardware_delay = buffer_size_bytes_;
   int frames_filled = callback_->OnMoreData(
       audio_bus_.get(), AudioBuffersState(0, hardware_delay));
-  if (frames_filled <= 0)
-    return;  // Audio source is shutting down, or halted on error.
-  int num_filled_bytes =
+  if (frames_filled <= 0) {
+    // Audio source is shutting down, or halted on error.
+    return;
+  }
+
+  // Note: If the internal representation ever changes from 16-bit PCM to
+  // raw float, the data must be clipped and sanitized since it may come
+  // from an untrusted source such as NaCl.
+  audio_bus_->Scale(volume_);
+  audio_bus_->ToInterleaved(
+      frames_filled, format_.bitsPerSample / 8,
+      audio_data_[active_buffer_index_]);
+
+  const int num_filled_bytes =
       frames_filled * audio_bus_->channels() * format_.bitsPerSample / 8;
   DCHECK_LE(static_cast<size_t>(num_filled_bytes), buffer_size_bytes_);
-  // Note: If this ever changes to output raw float the data must be clipped and
-  // sanitized since it may come from an untrusted source such as NaCl.
-  audio_bus_->Scale(volume_);
-  audio_bus_->ToInterleaved(
-      frames_filled, format_.bitsPerSample / 8, audio_data_[active_queue_]);
 
   // Enqueue the buffer for playback.
   SLresult err = (*simple_buffer_queue_)->Enqueue(
       simple_buffer_queue_,
-      audio_data_[active_queue_],
+      audio_data_[active_buffer_index_],
       num_filled_bytes);
   if (SL_RESULT_SUCCESS != err)
     HandleError(err);
 
-  active_queue_ = (active_queue_  + 1) % kNumOfQueuesInBuffer;
+  active_buffer_index_ = (active_buffer_index_  + 1) % kMaxNumOfBuffersInQueue;
 }
 
 void OpenSLESOutputStream::SetupAudioBuffer() {
+  DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(!audio_data_[0]);
-  for (int i = 0; i < kNumOfQueuesInBuffer; ++i) {
+  for (int i = 0; i < kMaxNumOfBuffersInQueue; ++i) {
     audio_data_[i] = new uint8[buffer_size_bytes_];
   }
 }
 
 void OpenSLESOutputStream::ReleaseAudioBuffer() {
+  DCHECK(thread_checker_.CalledOnValidThread());
   if (audio_data_[0]) {
-    for (int i = 0; i < kNumOfQueuesInBuffer; ++i) {
+    for (int i = 0; i < kMaxNumOfBuffersInQueue; ++i) {
       delete [] audio_data_[i];
       audio_data_[i] = NULL;
     }
   }
 }
 
 void OpenSLESOutputStream::HandleError(SLresult error) {
   DLOG(ERROR) << "OpenSLES Output error " << error;
   if (callback_)
     callback_->OnError(this);
 }
 
 }  // namespace media