Detect errors in audio output and report them upstream.

Stop feeding audio packets to AUDIO DemuxerStreams once audio has been disabled.

BUG=111409
TEST=uninstall pulseaudio, make /dev/snd inaccessible, and observe <video> plays correctly (muted) instead of hanging.


Review URL: http://codereview.chromium.org/9234066

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@119488 0039d316-1c4b-4281-b951-d872f2087c98

12 files changed, 46 insertions, 16 deletions

diff --git a/content/renderer/media/audio_device.cc b/content/renderer/media/audio_device.cc
index 99b01f2..b6ed566 100644
--- a/content/renderer/media/audio_device.cc
+++ b/content/renderer/media/audio_device.cc
@@ -211,6 +211,7 @@ void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) {
 void AudioDevice::OnStateChanged(AudioStreamState state) {
   if (state == kAudioStreamError) {
     DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)";
+    callback_->OnError();
   }
 }
 
diff --git a/content/renderer/media/audio_renderer_impl.cc b/content/renderer/media/audio_renderer_impl.cc
index eb54186..fda28ef 100644
--- a/content/renderer/media/audio_renderer_impl.cc
+++ b/content/renderer/media/audio_renderer_impl.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -14,6 +14,7 @@
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_buffers_state.h"
 #include "media/audio/audio_util.h"
+#include "media/base/filter_host.h"
 
 // We define GetBufferSizeForSampleRate() instead of using
 // GetAudioHardwareBufferSize() in audio_util because we're using
@@ -247,3 +248,7 @@ size_t AudioRendererImpl::Render(const std::vector<float*>& audio_data,
   }
   return filled_frames;
 }
+
+void AudioRendererImpl::OnError() {
+  host()->DisableAudioRenderer();
+}
diff --git a/content/renderer/media/audio_renderer_impl.h b/content/renderer/media/audio_renderer_impl.h
index ed09528..9e900d8 100644
--- a/content/renderer/media/audio_renderer_impl.h
+++ b/content/renderer/media/audio_renderer_impl.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// 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.
 //
@@ -80,6 +80,7 @@ class CONTENT_EXPORT AudioRendererImpl
   virtual size_t Render(const std::vector<float*>& audio_data,
                         size_t number_of_frames,
                         size_t audio_delay_milliseconds) OVERRIDE;
+  virtual void OnError() OVERRIDE;
 
   // Accessors used by tests.
   base::Time earliest_end_time() const {
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index 954b8c9..6cb3bc7 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -118,6 +118,10 @@ size_t WebRtcAudioDeviceImpl::Render(
   return number_of_frames;
 }
 
+void WebRtcAudioDeviceImpl::OnError() {
+  // TODO(henrika): Implement error handling.
+}
+
 void WebRtcAudioDeviceImpl::Capture(
     const std::vector<float*>& audio_data,
     size_t number_of_frames,
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index 28f4ae4..210b9ac 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -115,6 +115,7 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   virtual size_t Render(const std::vector<float*>& audio_data,
                         size_t number_of_frames,
                         size_t audio_delay_milliseconds) OVERRIDE;
+  virtual void OnError() OVERRIDE;
 
   // AudioInputDevice::CaptureCallback implementation.
   virtual void Capture(const std::vector<float*>& audio_data,
diff --git a/content/renderer/renderer_webaudiodevice_impl.cc b/content/renderer/renderer_webaudiodevice_impl.cc
index 77c6ba37..623fb5a 100644
--- a/content/renderer/renderer_webaudiodevice_impl.cc
+++ b/content/renderer/renderer_webaudiodevice_impl.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -51,3 +51,7 @@ size_t RendererWebAudioDeviceImpl::Render(const std::vector<float*>& audio_data,
   }
   return number_of_frames;
 }
+
+void RendererWebAudioDeviceImpl::OnError() {
+  // TODO(crogers): implement error handling.
+}
diff --git a/content/renderer/renderer_webaudiodevice_impl.h b/content/renderer/renderer_webaudiodevice_impl.h
index e4a5bf1..4a6d462 100644
--- a/content/renderer/renderer_webaudiodevice_impl.h
+++ b/content/renderer/renderer_webaudiodevice_impl.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -30,6 +30,7 @@ class RendererWebAudioDeviceImpl : public WebKit::WebAudioDevice,
   virtual size_t Render(const std::vector<float*>& audio_data,
                         size_t number_of_frames,
                         size_t audio_delay_milliseconds) OVERRIDE;
+  virtual void OnError() OVERRIDE;
 
  private:
   scoped_refptr<AudioDevice> audio_device_;
diff --git a/media/audio/linux/alsa_output.cc b/media/audio/linux/alsa_output.cc
index b3f931d..75de924 100644
--- a/media/audio/linux/alsa_output.cc
+++ b/media/audio/linux/alsa_output.cc
@@ -268,6 +268,8 @@ bool AlsaPcmOutputStream::Open() {
   // Finish initializing the stream if the device was opened successfully.
   if (playback_handle_ == NULL) {
     stop_stream_ = true;
+    TransitionTo(kInError);
+    return false;
   } else {
     bytes_per_output_frame_ = should_downmix_ ? 2 * bytes_per_sample_ :
         bytes_per_frame_;
diff --git a/media/audio/linux/alsa_output_unittest.cc b/media/audio/linux/alsa_output_unittest.cc
index f1870d2..a8359f6 100644
--- a/media/audio/linux/alsa_output_unittest.cc
+++ b/media/audio/linux/alsa_output_unittest.cc
@@ -355,11 +355,10 @@ TEST_F(AlsaPcmOutputStreamTest, PcmOpenFailed) {
   EXPECT_CALL(mock_alsa_wrapper_, StrError(kTestFailedErrno))
       .WillOnce(Return(kDummyMessage));
 
-  ASSERT_TRUE(test_stream_->Open());
-  ASSERT_EQ(AlsaPcmOutputStream::kIsOpened, test_stream_->state());
+  ASSERT_FALSE(test_stream_->Open());
+  ASSERT_EQ(AlsaPcmOutputStream::kInError, test_stream_->state());
 
   // Ensure internal state is set for a no-op stream if PcmOpen() failes.
-  EXPECT_EQ(AlsaPcmOutputStream::kIsOpened, test_stream_->state());
   EXPECT_TRUE(test_stream_->stop_stream_);
   EXPECT_TRUE(test_stream_->playback_handle_ == NULL);
   EXPECT_FALSE(test_stream_->buffer_.get());
@@ -384,11 +383,10 @@ TEST_F(AlsaPcmOutputStreamTest, PcmSetParamsFailed) {
 
   // If open fails, the stream stays in kCreated because it has effectively had
   // no changes.
-  ASSERT_TRUE(test_stream_->Open());
-  EXPECT_EQ(AlsaPcmOutputStream::kIsOpened, test_stream_->state());
+  ASSERT_FALSE(test_stream_->Open());
+  EXPECT_EQ(AlsaPcmOutputStream::kInError, test_stream_->state());
 
   // Ensure internal state is set for a no-op stream if PcmSetParams() failes.
-  EXPECT_EQ(AlsaPcmOutputStream::kIsOpened, test_stream_->state());
   EXPECT_TRUE(test_stream_->stop_stream_);
   EXPECT_TRUE(test_stream_->playback_handle_ == NULL);
   EXPECT_FALSE(test_stream_->buffer_.get());
diff --git a/media/base/audio_renderer_sink.h b/media/base/audio_renderer_sink.h
index 5888c6b..350210c 100644
--- a/media/base/audio_renderer_sink.h
+++ b/media/base/audio_renderer_sink.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -32,6 +32,9 @@ class AudioRendererSink
                           size_t number_of_frames,
                           size_t audio_delay_milliseconds) = 0;
 
+    // Signals an error has occurred.
+    virtual void OnError() = 0;
+
    protected:
     virtual ~RenderCallback() {}
   };
diff --git a/media/filters/ffmpeg_demuxer.cc b/media/filters/ffmpeg_demuxer.cc
index 6e3627e..0cb2ea4 100644
--- a/media/filters/ffmpeg_demuxer.cc
+++ b/media/filters/ffmpeg_demuxer.cc
@@ -289,7 +289,8 @@ FFmpegDemuxer::FFmpegDemuxer(MessageLoop* message_loop, bool local_source)
       max_duration_(base::TimeDelta::FromMicroseconds(-1)),
       deferred_status_(PIPELINE_OK),
       first_seek_hack_(true),
-      start_time_(kNoTimestamp()) {
+      start_time_(kNoTimestamp()),
+      audio_disabled_(false) {
   DCHECK(message_loop_);
 }
 
@@ -662,7 +663,9 @@ void FFmpegDemuxer::DemuxTask() {
   // Defend against ffmpeg giving us a bad stream index.
   if (packet->stream_index >= 0 &&
       packet->stream_index < static_cast<int>(streams_.size()) &&
-      streams_[packet->stream_index]) {
+      streams_[packet->stream_index] &&
+      (!audio_disabled_ ||
+       streams_[packet->stream_index]->type() != DemuxerStream::AUDIO)) {
     FFmpegDemuxerStream* demuxer_stream = streams_[packet->stream_index];
 
     // If a packet is returned by FFmpeg's av_parser_parse2()
@@ -699,6 +702,7 @@ void FFmpegDemuxer::StopTask(const base::Closure& callback) {
 
 void FFmpegDemuxer::DisableAudioStreamTask() {
   DCHECK_EQ(MessageLoop::current(), message_loop_);
+  audio_disabled_ = true;
   StreamVector::iterator iter;
   for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
     if (*iter && (*iter)->type() == DemuxerStream::AUDIO) {
@@ -722,8 +726,10 @@ void FFmpegDemuxer::StreamHasEnded() {
   DCHECK_EQ(MessageLoop::current(), message_loop_);
   StreamVector::iterator iter;
   for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
-    if (!*iter)
+    if (!*iter ||
+        (audio_disabled_ && (*iter)->type() == DemuxerStream::AUDIO)) {
       continue;
+    }
     scoped_ptr_malloc<AVPacket, ScopedPtrAVFreePacket> packet(new AVPacket());
     memset(packet.get(), 0, sizeof(*packet.get()));
     (*iter)->EnqueuePacket(packet.Pass());
diff --git a/media/filters/ffmpeg_demuxer.h b/media/filters/ffmpeg_demuxer.h
index 66c6ac5..a180113 100644
--- a/media/filters/ffmpeg_demuxer.h
+++ b/media/filters/ffmpeg_demuxer.h
@@ -261,6 +261,10 @@ class MEDIA_EXPORT FFmpegDemuxer : public Demuxer, public FFmpegURLProtocol {
   // is 0.
   base::TimeDelta start_time_;
 
+  // Whether audio has been disabled for this demuxer (in which case this class
+  // drops packets destined for AUDIO demuxer streams on the floor).
+  bool audio_disabled_;
+
   DISALLOW_COPY_AND_ASSIGN(FFmpegDemuxer);
 };