Use ChannelMixer where applicable. Remove FoldChannels.

- Adds ChannelMixer to AudioOutputResampler.
- Switches all users of FoldChannels over to ChannelMixer.
- Removes channel mixing from WASAPIAudioOutputStream.

BUG=138762
TEST=unit tests, manual playback on all platforms.


Review URL: https://chromiumcodereview.appspot.com/11188019

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@163163 0039d316-1c4b-4281-b951-d872f2087c98

3 files changed, 36 insertions, 32 deletions

diff --git a/media/audio/linux/alsa_output.cc b/media/audio/linux/alsa_output.cc
index c51a27c..96be896 100644
--- a/media/audio/linux/alsa_output.cc
+++ b/media/audio/linux/alsa_output.cc
@@ -46,6 +46,7 @@
 #include "media/audio/linux/alsa_util.h"
 #include "media/audio/linux/alsa_wrapper.h"
 #include "media/audio/linux/audio_manager_linux.h"
+#include "media/base/channel_mixer.h"
 #include "media/base/data_buffer.h"
 #include "media/base/seekable_buffer.h"
 
@@ -155,10 +156,10 @@ AlsaPcmOutputStream::AlsaPcmOutputStream(const std::string& device_name,
     : requested_device_name_(device_name),
       pcm_format_(alsa_util::BitsToFormat(params.bits_per_sample())),
       channels_(params.channels()),
+      channel_layout_(params.channel_layout()),
       sample_rate_(params.sample_rate()),
       bytes_per_sample_(params.bits_per_sample() / 8),
       bytes_per_frame_(channels_ * params.bits_per_sample() / 8),
-      should_downmix_(false),
       packet_size_(params.GetBytesPerBuffer()),
       micros_per_packet_(FramesToMicros(
           params.frames_per_buffer(), sample_rate_)),
@@ -242,8 +243,8 @@ bool AlsaPcmOutputStream::Open() {
     TransitionTo(kInError);
     return false;
   } else {
-    bytes_per_output_frame_ = should_downmix_ ? 2 * bytes_per_sample_ :
-        bytes_per_frame_;
+    bytes_per_output_frame_ = channel_mixer_ ?
+        mixed_audio_bus_->channels() * bytes_per_sample_ : bytes_per_frame_;
     uint32 output_packet_size = frames_per_packet_ * bytes_per_output_frame_;
     buffer_.reset(new media::SeekableBuffer(0, output_packet_size));
 
@@ -382,35 +383,31 @@ void AlsaPcmOutputStream::BufferPacket(bool* source_exhausted) {
         audio_bus_.get(), AudioBuffersState(buffer_delay, hardware_delay));
     size_t packet_size = frames_filled * bytes_per_frame_;
     DCHECK_LE(packet_size, packet_size_);
-    // 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_->ToInterleaved(
-        frames_filled, bytes_per_sample_, packet->GetWritableData());
 
     // Reset the |last_fill_time| to avoid back to back RunDataCallback().
     last_fill_time_ = base::Time::Now();
 
     // TODO(dalecurtis): Channel downmixing, upmixing, should be done in mixer;
     // volume adjust should use SSE optimized vector_fmul() prior to interleave.
-    if (should_downmix_) {
-      if (media::FoldChannels(packet->GetWritableData(),
-                              packet_size,
-                              channels_,
-                              bytes_per_sample_,
-                              volume_)) {
-        // Adjust packet size for downmix.
-        packet_size = packet_size / bytes_per_frame_ * bytes_per_output_frame_;
-      } else {
-        LOG(ERROR) << "Folding failed";
-      }
-    } else {
-      media::AdjustVolume(packet->GetWritableData(),
-                          packet_size,
-                          channels_,
-                          bytes_per_sample_,
-                          volume_);
+    AudioBus* output_bus = audio_bus_.get();
+    if (channel_mixer_) {
+      output_bus = mixed_audio_bus_.get();
+      channel_mixer_->Transform(audio_bus_.get(), output_bus);
+      // Adjust packet size for downmix.
+      packet_size = packet_size / bytes_per_frame_ * bytes_per_output_frame_;
     }
 
+    // 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.
+    output_bus->ToInterleaved(
+        frames_filled, bytes_per_sample_, packet->GetWritableData());
+
+    media::AdjustVolume(packet->GetWritableData(),
+                        packet_size,
+                        output_bus->channels(),
+                        bytes_per_sample_,
+                        volume_);
+
     if (packet_size > 0) {
       packet->SetDataSize(packet_size);
       // Add the packet to the buffer.
@@ -708,12 +705,13 @@ snd_pcm_t* AlsaPcmOutputStream::AutoSelectDevice(unsigned int latency) {
   // output to have the correct ordering according to Lennart.  For the channel
   // formats that we know how to downmix from (3 channel to 8 channel), setup
   // downmixing.
-  //
-  // TODO(ajwong): We need a SupportsFolding() function.
   uint32 default_channels = channels_;
-  if (default_channels > 2 && default_channels <= 8) {
-    should_downmix_ = true;
+  if (default_channels > 2) {
+    channel_mixer_.reset(new ChannelMixer(
+        channel_layout_, CHANNEL_LAYOUT_STEREO));
     default_channels = 2;
+    mixed_audio_bus_ = AudioBus::Create(
+        default_channels, audio_bus_->frames());
   }
 
   // Step 3.
diff --git a/media/audio/linux/alsa_output.h b/media/audio/linux/alsa_output.h
index 736015e..ffb29f4 100644
--- a/media/audio/linux/alsa_output.h
+++ b/media/audio/linux/alsa_output.h
@@ -39,6 +39,7 @@ namespace media {
 
 class AlsaWrapper;
 class AudioManagerLinux;
+class ChannelMixer;
 class SeekableBuffer;
 
 class MEDIA_EXPORT AlsaPcmOutputStream : public AudioOutputStream {
@@ -131,7 +132,7 @@ class MEDIA_EXPORT AlsaPcmOutputStream : public AudioOutputStream {
   snd_pcm_sframes_t GetCurrentDelay();
 
   // Attempts to find the best matching linux audio device for the given number
-  // of channels.  This function will set |device_name_| and |should_downmix_|.
+  // of channels.  This function will set |device_name_| and |channel_mixer_|.
   snd_pcm_t* AutoSelectDevice(uint32 latency);
 
   // Functions to safeguard state transitions.  All changes to the object state
@@ -164,13 +165,13 @@ class MEDIA_EXPORT AlsaPcmOutputStream : public AudioOutputStream {
   const std::string requested_device_name_;
   const snd_pcm_format_t pcm_format_;
   const uint32 channels_;
+  const ChannelLayout channel_layout_;
   const uint32 sample_rate_;
   const uint32 bytes_per_sample_;
   const uint32 bytes_per_frame_;
 
   // Device configuration data. Populated after OpenTask() completes.
   std::string device_name_;
-  bool should_downmix_;
   uint32 packet_size_;
   uint32 micros_per_packet_;
   uint32 latency_micros_;
@@ -214,6 +215,10 @@ class MEDIA_EXPORT AlsaPcmOutputStream : public AudioOutputStream {
   // Container for retrieving data from AudioSourceCallback::OnMoreData().
   scoped_ptr<AudioBus> audio_bus_;
 
+  // Channel mixer and temporary bus for the final mixed channel data.
+  scoped_ptr<ChannelMixer> channel_mixer_;
+  scoped_ptr<AudioBus> mixed_audio_bus_;
+
   DISALLOW_COPY_AND_ASSIGN(AlsaPcmOutputStream);
 };
 
diff --git a/media/audio/linux/alsa_output_unittest.cc b/media/audio/linux/alsa_output_unittest.cc
index 69bcf4f..9d6185c 100644
--- a/media/audio/linux/alsa_output_unittest.cc
+++ b/media/audio/linux/alsa_output_unittest.cc
@@ -734,7 +734,8 @@ TEST_F(AlsaPcmOutputStreamTest, AutoSelectDevice_DeviceSelect) {
 
     AlsaPcmOutputStream* test_stream = CreateStream(kExpectedLayouts[i]);
     EXPECT_TRUE(test_stream->AutoSelectDevice(i));
-    EXPECT_EQ(kExpectedDownmix[i], test_stream->should_downmix_);
+    EXPECT_EQ(kExpectedDownmix[i],
+              static_cast<bool>(test_stream->channel_mixer_));
 
     Mock::VerifyAndClearExpectations(&mock_alsa_wrapper_);
     Mock::VerifyAndClearExpectations(mock_manager_.get());
@@ -804,7 +805,7 @@ TEST_F(AlsaPcmOutputStreamTest, AutoSelectDevice_HintFail) {
 
   AlsaPcmOutputStream* test_stream = CreateStream(CHANNEL_LAYOUT_5_0);
   EXPECT_TRUE(test_stream->AutoSelectDevice(5));
-  EXPECT_TRUE(test_stream->should_downmix_);
+  EXPECT_TRUE(test_stream->channel_mixer_);
   test_stream->Close();
 }