Tab Audio Mirroring: WebContentsAudioInputStream is a new implementation which represents the lifet…

media/audio/virtual_audio_input_stream.cc

Index: media/audio/virtual_audio_input_stream.cc
diff --git a/media/audio/virtual_audio_input_stream.cc b/media/audio/virtual_audio_input_stream.cc
index 4f2f9bd04fb84a4619766597777c75665b5641c6..e3145093be792a79cfed91f3fa715301159a44de 100644
--- a/media/audio/virtual_audio_input_stream.cc
+++ b/media/audio/virtual_audio_input_stream.cc
@@ -9,6 +9,7 @@
 
 #include "base/bind.h"
 #include "base/message_loop.h"
+#include "base/message_loop_proxy.h"
 #include "media/audio/virtual_audio_output_stream.h"
 
 namespace media {
@@ -47,26 +48,20 @@ class LoopbackAudioConverter : public AudioConverter::InputCallback {
   DISALLOW_COPY_AND_ASSIGN(LoopbackAudioConverter);
 };
 
-VirtualAudioInputStream* VirtualAudioInputStream::MakeStream(
-    AudioManagerBase* manager, const AudioParameters& params,
-    base::MessageLoopProxy* message_loop) {
-  return new VirtualAudioInputStream(manager, params, message_loop);
-}
-
 VirtualAudioInputStream::VirtualAudioInputStream(
-    AudioManagerBase* manager, const AudioParameters& params,
-    base::MessageLoopProxy* message_loop)
-    : audio_manager_(manager),
-      message_loop_(message_loop),
+    const AudioParameters& params, base::MessageLoopProxy* message_loop)
+    : message_loop_(message_loop),
       callback_(NULL),
-      buffer_duration_ms_(base::TimeDelta::FromMilliseconds(
-          params.frames_per_buffer() * base::Time::kMillisecondsPerSecond /
-          static_cast<float>(params.sample_rate()))),
+      buffer_duration_(base::TimeDelta::FromMicroseconds(
+          params.frames_per_buffer() * base::Time::kMicrosecondsPerSecond /
+          params.sample_rate())),
       buffer_(new uint8[params.GetBytesPerBuffer()]),
       params_(params),
       audio_bus_(AudioBus::Create(params_)),
       mixer_(params_, params_, false),
-      num_attached_outputs_streams_(0) {
+      num_attached_output_streams_(0) {
+  DCHECK(params_.IsValid());
+  DCHECK(message_loop_);
 }
 
 VirtualAudioInputStream::~VirtualAudioInputStream() {
@@ -74,10 +69,16 @@ VirtualAudioInputStream::~VirtualAudioInputStream() {
        it != converters_.end(); ++it)
     delete it->second;
 
-  DCHECK_EQ(0, num_attached_outputs_streams_);
+  DCHECK_EQ(0, num_attached_output_streams_);
+}
+
+void VirtualAudioInputStream::RunOnceClosed(const base::Closure& cb) {
+  DCHECK(on_close_cb_.is_null());
+  on_close_cb_ = cb;
 }
 
 bool VirtualAudioInputStream::Open() {
+  DCHECK(message_loop_->BelongsToCurrentThread());
   memset(buffer_.get(), 0, params_.GetBytesPerBuffer());
   return true;
 }
@@ -87,8 +88,7 @@ void VirtualAudioInputStream::Start(AudioInputCallback* callback) {
   callback_ = callback;
   on_more_data_cb_.Reset(base::Bind(&VirtualAudioInputStream::ReadAudio,
                                     base::Unretained(this)));
-  audio_manager_->GetMessageLoop()->PostTask(FROM_HERE,
-                                             on_more_data_cb_.callback());
+  message_loop_->PostTask(FROM_HERE, on_more_data_cb_.callback());
 }
 
 void VirtualAudioInputStream::Stop() {
@@ -111,7 +111,7 @@ void VirtualAudioInputStream::AddOutputStream(
     mixer_.AddInput(converter->second);
   }
   converter->second->AddInput(stream);
-  ++num_attached_outputs_streams_;
+  ++num_attached_output_streams_;
 }
 
 void VirtualAudioInputStream::RemoveOutputStream(
@@ -121,7 +121,8 @@ void VirtualAudioInputStream::RemoveOutputStream(
   DCHECK(converters_.find(output_params) != converters_.end());
   converters_[output_params]->RemoveInput(stream);
 
-  --num_attached_outputs_streams_;
+  --num_attached_output_streams_;
+  DCHECK_LE(0, num_attached_output_streams_);
 }
 
 void VirtualAudioInputStream::ReadAudio() {
@@ -139,9 +140,14 @@ void VirtualAudioInputStream::ReadAudio() {
                     params_.GetBytesPerBuffer(),
                     1.0);
 
+  // TODO(miu): We want ReadAudio() to be called every buffer_duration_ amount
+  // of time.  However, we're not accounting for the time spent doing the
+  // conversion and data delivery above, which is significant.  Therefore, audio
+  // data is always being delivered at a slower rate than required, and this
+  // will cause cut-outs!

[DaleCurtis, 2013/01/15 22:02:18]

Justin has a fix out for this?

[miu, 2013/01/16 03:22:18]

Yep (https://codereview.chromium.org/11889041/).  The TODO comment was written
before that work was started.  I'll take it out and rebase once his change is
committed.

   message_loop_->PostDelayedTask(FROM_HERE,
                                  on_more_data_cb_.callback(),
-                                 buffer_duration_ms_);
+                                 buffer_duration_);
 }
 
 void VirtualAudioInputStream::Close() {
@@ -151,7 +157,11 @@ void VirtualAudioInputStream::Close() {
     callback_->OnClose(this);
     callback_ = NULL;
   }
-  audio_manager_->ReleaseInputStream(this);
+  if (!on_close_cb_.is_null()) {
+    const base::Closure cb = on_close_cb_;
+    on_close_cb_.Reset();
+    cb.Run();
+  }
 }
 
 double VirtualAudioInputStream::GetMaxVolume() {